Bump cmake version

The division opcode typically also generates the remainder, making the result
of x%y free with x/y.

.gitignore

@@ -1,7 +1,9 @@
-build
+build*/
 winbuild
 win64build
-include/SLES
-include/sndio.h
-include/sys
-openal-soft.kdev4
+
+## kdevelop
+*.kdev4
+
+## qt-creator
+CMakeLists.txt.user*

.travis.yml

@@ -1,70 +0,0 @@
-language: c
-matrix:
-  include:
-    - os: linux
-      dist: trusty
-    - os: linux
-      dist: trusty
-      env:
-        - BUILD_ANDROID=true
-    - os: osx
-sudo: required
-cache:
-  directories:
-    - $HOME/android-ndk-r14
-install:
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then
-      # Install pulseaudio, portaudio, ALSA, JACK dependencies for
-      # corresponding backends.
-      # Install Qt5 dependency for alsoft-config.
-      sudo apt-get install -qq \
-        libpulse-dev \
-        portaudio19-dev \
-        libasound2-dev \
-        libjack-dev \
-        qtbase5-dev
-    fi
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
-      if [[ ! -d ~/android-ndk-r14 || -z "$(ls -A ~/android-ndk-r14)" ]]; then
-        curl -o ~/android-ndk.zip https://dl.google.com/android/repository/android-ndk-r14-linux-x86_64.zip
-        unzip -q ~/android-ndk.zip -d ~ \
-          'android-ndk-r14/build/cmake/*' \
-          'android-ndk-r14/platforms/android-9/arch-arm/*' \
-          'android-ndk-r14/source.properties' \
-          'android-ndk-r14/sources/cxx-stl/gnu-libstdc++/4.9/libs/armeabi-v7a/*' \
-          'android-ndk-r14/sysroot/*' \
-          'android-ndk-r14/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/*' \
-          'android-ndk-r14/toolchains/llvm/prebuilt/linux-x86_64/*'
-        sed -i -e 's/VERSION 3.6.0/VERSION 3.2/' ~/android-ndk-r14/build/cmake/android.toolchain.cmake
-      fi
-    fi
-script:
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "linux" && -z "${BUILD_ANDROID}" ]]; then
-      cmake \
-        -DALSOFT_REQUIRE_ALSA=ON \
-        -DALSOFT_REQUIRE_OSS=ON \
-        -DALSOFT_REQUIRE_PORTAUDIO=ON \
-        -DALSOFT_REQUIRE_PULSEAUDIO=ON \
-        -DALSOFT_REQUIRE_JACK=ON \
-        -DALSOFT_EMBED_HRTF_DATA=YES \
-        .
-    fi
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
-      cmake \
-        -DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r14/build/cmake/android.toolchain.cmake \
-        -DALSOFT_REQUIRE_OPENSL=ON \
-        -DALSOFT_EMBED_HRTF_DATA=YES \
-        .
-    fi
-  - >
-    if [[ "${TRAVIS_OS_NAME}" == "osx" ]]; then
-      cmake \
-        -DALSOFT_REQUIRE_COREAUDIO=ON \
-        -DALSOFT_EMBED_HRTF_DATA=YES \
-        .
-    fi
-  - make -j2

Alc/alcConfig.c

@@ -1,652 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#ifdef _WIN32
-#ifdef __MINGW32__
-#define _WIN32_IE 0x501
-#else
-#define _WIN32_IE 0x400
-#endif
-#endif
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <ctype.h>
-#include <string.h>
-#ifdef _WIN32_IE
-#include <windows.h>
-#include <shlobj.h>
-#endif
-
-#include "alMain.h"
-#include "compat.h"
-#include "bool.h"
-
-
-typedef struct ConfigEntry {
-    char *key;
-    char *value;
-} ConfigEntry;
-
-typedef struct ConfigBlock {
-    ConfigEntry *entries;
-    unsigned int entryCount;
-} ConfigBlock;
-static ConfigBlock cfgBlock;
-
-
-static char *lstrip(char *line)
-{
-    while(isspace(line[0]))
-        line++;
-    return line;
-}
-
-static char *rstrip(char *line)
-{
-    size_t len = strlen(line);
-    while(len > 0 && isspace(line[len-1]))
-        len--;
-    line[len] = 0;
-    return line;
-}
-
-static int readline(FILE *f, char **output, size_t *maxlen)
-{
-    size_t len = 0;
-    int c;
-
-    while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n'))
-        ;
-    if(c == EOF)
-        return 0;
-
-    do {
-        if(len+1 >= *maxlen)
-        {
-            void *temp = NULL;
-            size_t newmax;
-
-            newmax = (*maxlen ? (*maxlen)<<1 : 32);
-            if(newmax > *maxlen)
-                temp = realloc(*output, newmax);
-            if(!temp)
-            {
-                ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen);
-                return 0;
-            }
-
-            *output = temp;
-            *maxlen = newmax;
-        }
-        (*output)[len++] = c;
-        (*output)[len] = '\0';
-    } while((c=fgetc(f)) != EOF && c != '\r' && c != '\n');
-
-    return 1;
-}
-
-
-static char *expdup(const char *str)
-{
-    char *output = NULL;
-    size_t maxlen = 0;
-    size_t len = 0;
-
-    while(*str != '\0')
-    {
-        const char *addstr;
-        size_t addstrlen;
-        size_t i;
-
-        if(str[0] != '$')
-        {
-            const char *next = strchr(str, '$');
-            addstr = str;
-            addstrlen = next ? (size_t)(next-str) : strlen(str);
-
-            str += addstrlen;
-        }
-        else
-        {
-            str++;
-            if(*str == '$')
-            {
-                const char *next = strchr(str+1, '$');
-                addstr = str;
-                addstrlen = next ? (size_t)(next-str) : strlen(str);
-
-                str += addstrlen;
-            }
-            else
-            {
-                bool hasbraces;
-                char envname[1024];
-                size_t k = 0;
-
-                hasbraces = (*str == '{');
-                if(hasbraces) str++;
-
-                while((isalnum(*str) || *str == '_') && k < sizeof(envname)-1)
-                    envname[k++] = *(str++);
-                envname[k++] = '\0';
-
-                if(hasbraces && *str != '}')
-                    continue;
-
-                if(hasbraces) str++;
-                if((addstr=getenv(envname)) == NULL)
-                    continue;
-                addstrlen = strlen(addstr);
-            }
-        }
-        if(addstrlen == 0)
-            continue;
-
-        if(addstrlen >= maxlen-len)
-        {
-            void *temp = NULL;
-            size_t newmax;
-
-            newmax = len+addstrlen+1;
-            if(newmax > maxlen)
-                temp = realloc(output, newmax);
-            if(!temp)
-            {
-                ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, maxlen);
-                return output;
-            }
-
-            output = temp;
-            maxlen = newmax;
-        }
-
-        for(i = 0;i < addstrlen;i++)
-            output[len++] = addstr[i];
-        output[len] = '\0';
-    }
-
-    return output ? output : calloc(1, 1);
-}
-
-
-static void LoadConfigFromFile(FILE *f)
-{
-    char curSection[128] = "";
-    char *buffer = NULL;
-    size_t maxlen = 0;
-    ConfigEntry *ent;
-
-    while(readline(f, &buffer, &maxlen))
-    {
-        char *line, *comment;
-        char key[256] = "";
-        char value[256] = "";
-
-        line = rstrip(lstrip(buffer));
-        if(!line[0]) continue;
-
-        if(line[0] == '[')
-        {
-            char *section = line+1;
-            char *endsection;
-
-            endsection = strchr(section, ']');
-            if(!endsection || section == endsection)
-            {
-                ERR("config parse error: bad line \"%s\"\n", line);
-                continue;
-            }
-            if(endsection[1] != 0)
-            {
-                char *end = endsection+1;
-                while(isspace(*end))
-                    ++end;
-                if(*end != 0 && *end != '#')
-                {
-                    ERR("config parse error: bad line \"%s\"\n", line);
-                    continue;
-                }
-            }
-            *endsection = 0;
-
-            if(strcasecmp(section, "general") == 0)
-                curSection[0] = 0;
-            else
-            {
-                size_t len, p = 0;
-                do {
-                    char *nextp = strchr(section, '%');
-                    if(!nextp)
-                    {
-                        strncpy(curSection+p, section, sizeof(curSection)-1-p);
-                        break;
-                    }
-
-                    len = nextp - section;
-                    if(len > sizeof(curSection)-1-p)
-                        len = sizeof(curSection)-1-p;
-                    strncpy(curSection+p, section, len);
-                    p += len;
-                    section = nextp;
-
-                    if(((section[1] >= '0' && section[1] <= '9') ||
-                        (section[1] >= 'a' && section[1] <= 'f') ||
-                        (section[1] >= 'A' && section[1] <= 'F')) &&
-                       ((section[2] >= '0' && section[2] <= '9') ||
-                        (section[2] >= 'a' && section[2] <= 'f') ||
-                        (section[2] >= 'A' && section[2] <= 'F')))
-                    {
-                        unsigned char b = 0;
-                        if(section[1] >= '0' && section[1] <= '9')
-                            b = (section[1]-'0') << 4;
-                        else if(section[1] >= 'a' && section[1] <= 'f')
-                            b = (section[1]-'a'+0xa) << 4;
-                        else if(section[1] >= 'A' && section[1] <= 'F')
-                            b = (section[1]-'A'+0x0a) << 4;
-                        if(section[2] >= '0' && section[2] <= '9')
-                            b |= (section[2]-'0');
-                        else if(section[2] >= 'a' && section[2] <= 'f')
-                            b |= (section[2]-'a'+0xa);
-                        else if(section[2] >= 'A' && section[2] <= 'F')
-                            b |= (section[2]-'A'+0x0a);
-                        if(p < sizeof(curSection)-1)
-                            curSection[p++] = b;
-                        section += 3;
-                    }
-                    else if(section[1] == '%')
-                    {
-                        if(p < sizeof(curSection)-1)
-                            curSection[p++] = '%';
-                        section += 2;
-                    }
-                    else
-                    {
-                        if(p < sizeof(curSection)-1)
-                            curSection[p++] = '%';
-                        section += 1;
-                    }
-                    if(p < sizeof(curSection)-1)
-                        curSection[p] = 0;
-                } while(p < sizeof(curSection)-1 && *section != 0);
-                curSection[sizeof(curSection)-1] = 0;
-            }
-
-            continue;
-        }
-
-        comment = strchr(line, '#');
-        if(comment) *(comment++) = 0;
-        if(!line[0]) continue;
-
-        if(sscanf(line, "%255[^=] = \"%255[^\"]\"", key, value) == 2 ||
-           sscanf(line, "%255[^=] = '%255[^\']'", key, value) == 2 ||
-           sscanf(line, "%255[^=] = %255[^\n]", key, value) == 2)
-        {
-            /* sscanf doesn't handle '' or "" as empty values, so clip it
-             * manually. */
-            if(strcmp(value, "\"\"") == 0 || strcmp(value, "''") == 0)
-                value[0] = 0;
-        }
-        else if(sscanf(line, "%255[^=] %255[=]", key, value) == 2)
-        {
-            /* Special case for 'key =' */
-            value[0] = 0;
-        }
-        else
-        {
-            ERR("config parse error: malformed option line: \"%s\"\n\n", line);
-            continue;
-        }
-        rstrip(key);
-
-        if(curSection[0] != 0)
-        {
-            size_t len = strlen(curSection);
-            memmove(&key[len+1], key, sizeof(key)-1-len);
-            key[len] = '/';
-            memcpy(key, curSection, len);
-        }
-
-        /* Check if we already have this option set */
-        ent = cfgBlock.entries;
-        while((unsigned int)(ent-cfgBlock.entries) < cfgBlock.entryCount)
-        {
-            if(strcasecmp(ent->key, key) == 0)
-                break;
-            ent++;
-        }
-
-        if((unsigned int)(ent-cfgBlock.entries) >= cfgBlock.entryCount)
-        {
-            /* Allocate a new option entry */
-            ent = realloc(cfgBlock.entries, (cfgBlock.entryCount+1)*sizeof(ConfigEntry));
-            if(!ent)
-            {
-                 ERR("config parse error: error reallocating config entries\n");
-                 continue;
-            }
-            cfgBlock.entries = ent;
-            ent = cfgBlock.entries + cfgBlock.entryCount;
-            cfgBlock.entryCount++;
-
-            ent->key = strdup(key);
-            ent->value = NULL;
-        }
-
-        free(ent->value);
-        ent->value = expdup(value);
-
-        TRACE("found '%s' = '%s'\n", ent->key, ent->value);
-    }
-
-    free(buffer);
-}
-
-#ifdef _WIN32
-void ReadALConfig(void)
-{
-    WCHAR buffer[PATH_MAX];
-    const WCHAR *str;
-    al_string ppath;
-    FILE *f;
-
-    if(SHGetSpecialFolderPathW(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE)
-    {
-        al_string filepath = AL_STRING_INIT_STATIC();
-        alstr_copy_wcstr(&filepath, buffer);
-        alstr_append_cstr(&filepath, "\\alsoft.ini");
-
-        TRACE("Loading config %s...\n", alstr_get_cstr(filepath));
-        f = al_fopen(alstr_get_cstr(filepath), "rt");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-        alstr_reset(&filepath);
-    }
-
-    ppath = GetProcPath();
-    if(!alstr_empty(ppath))
-    {
-        alstr_append_cstr(&ppath, "\\alsoft.ini");
-        TRACE("Loading config %s...\n", alstr_get_cstr(ppath));
-        f = al_fopen(alstr_get_cstr(ppath), "r");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-
-    if((str=_wgetenv(L"ALSOFT_CONF")) != NULL && *str)
-    {
-        al_string filepath = AL_STRING_INIT_STATIC();
-        alstr_copy_wcstr(&filepath, str);
-
-        TRACE("Loading config %s...\n", alstr_get_cstr(filepath));
-        f = al_fopen(alstr_get_cstr(filepath), "rt");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-        alstr_reset(&filepath);
-    }
-
-    alstr_reset(&ppath);
-}
-#else
-void ReadALConfig(void)
-{
-    char buffer[PATH_MAX];
-    const char *str;
-    al_string ppath;
-    FILE *f;
-
-    str = "/etc/openal/alsoft.conf";
-
-    TRACE("Loading config %s...\n", str);
-    f = al_fopen(str, "r");
-    if(f)
-    {
-        LoadConfigFromFile(f);
-        fclose(f);
-    }
-
-    if(!(str=getenv("XDG_CONFIG_DIRS")) || str[0] == 0)
-        str = "/etc/xdg";
-    strncpy(buffer, str, sizeof(buffer)-1);
-    buffer[sizeof(buffer)-1] = 0;
-    /* Go through the list in reverse, since "the order of base directories
-     * denotes their importance; the first directory listed is the most
-     * important". Ergo, we need to load the settings from the later dirs
-     * first so that the settings in the earlier dirs override them.
-     */
-    while(1)
-    {
-        char *next = strrchr(buffer, ':');
-        if(next) *(next++) = 0;
-        else next = buffer;
-
-        if(next[0] != '/')
-            WARN("Ignoring XDG config dir: %s\n", next);
-        else
-        {
-            size_t len = strlen(next);
-            strncpy(next+len, "/alsoft.conf", buffer+sizeof(buffer)-next-len);
-            buffer[sizeof(buffer)-1] = 0;
-
-            TRACE("Loading config %s...\n", next);
-            f = al_fopen(next, "r");
-            if(f)
-            {
-                LoadConfigFromFile(f);
-                fclose(f);
-            }
-        }
-        if(next == buffer)
-            break;
-    }
-
-    if((str=getenv("HOME")) != NULL && *str)
-    {
-        snprintf(buffer, sizeof(buffer), "%s/.alsoftrc", str);
-
-        TRACE("Loading config %s...\n", buffer);
-        f = al_fopen(buffer, "r");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-
-    if((str=getenv("XDG_CONFIG_HOME")) != NULL && str[0] != 0)
-        snprintf(buffer, sizeof(buffer), "%s/%s", str, "alsoft.conf");
-    else
-    {
-        buffer[0] = 0;
-        if((str=getenv("HOME")) != NULL && str[0] != 0)
-            snprintf(buffer, sizeof(buffer), "%s/.config/%s", str, "alsoft.conf");
-    }
-    if(buffer[0] != 0)
-    {
-        TRACE("Loading config %s...\n", buffer);
-        f = al_fopen(buffer, "r");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-
-    ppath = GetProcPath();
-    if(!alstr_empty(ppath))
-    {
-        alstr_append_cstr(&ppath, "/alsoft.conf");
-        TRACE("Loading config %s...\n", alstr_get_cstr(ppath));
-        f = al_fopen(alstr_get_cstr(ppath), "r");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-
-    if((str=getenv("ALSOFT_CONF")) != NULL && *str)
-    {
-        TRACE("Loading config %s...\n", str);
-        f = al_fopen(str, "r");
-        if(f)
-        {
-            LoadConfigFromFile(f);
-            fclose(f);
-        }
-    }
-
-    alstr_reset(&ppath);
-}
-#endif
-
-void FreeALConfig(void)
-{
-    unsigned int i;
-
-    for(i = 0;i < cfgBlock.entryCount;i++)
-    {
-        free(cfgBlock.entries[i].key);
-        free(cfgBlock.entries[i].value);
-    }
-    free(cfgBlock.entries);
-}
-
-const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def)
-{
-    unsigned int i;
-    char key[256];
-
-    if(!keyName)
-        return def;
-
-    if(blockName && strcasecmp(blockName, "general") != 0)
-    {
-        if(devName)
-            snprintf(key, sizeof(key), "%s/%s/%s", blockName, devName, keyName);
-        else
-            snprintf(key, sizeof(key), "%s/%s", blockName, keyName);
-    }
-    else
-    {
-        if(devName)
-            snprintf(key, sizeof(key), "%s/%s", devName, keyName);
-        else
-        {
-            strncpy(key, keyName, sizeof(key)-1);
-            key[sizeof(key)-1] = 0;
-        }
-    }
-
-    for(i = 0;i < cfgBlock.entryCount;i++)
-    {
-        if(strcmp(cfgBlock.entries[i].key, key) == 0)
-        {
-            TRACE("Found %s = \"%s\"\n", key, cfgBlock.entries[i].value);
-            if(cfgBlock.entries[i].value[0])
-                return cfgBlock.entries[i].value;
-            return def;
-        }
-    }
-
-    if(!devName)
-    {
-        TRACE("Key %s not found\n", key);
-        return def;
-    }
-    return GetConfigValue(NULL, blockName, keyName, def);
-}
-
-int ConfigValueExists(const char *devName, const char *blockName, const char *keyName)
-{
-    const char *val = GetConfigValue(devName, blockName, keyName, "");
-    return !!val[0];
-}
-
-int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret)
-{
-    const char *val = GetConfigValue(devName, blockName, keyName, "");
-    if(!val[0]) return 0;
-
-    *ret = val;
-    return 1;
-}
-
-int ConfigValueInt(const char *devName, const char *blockName, const char *keyName, int *ret)
-{
-    const char *val = GetConfigValue(devName, blockName, keyName, "");
-    if(!val[0]) return 0;
-
-    *ret = strtol(val, NULL, 0);
-    return 1;
-}
-
-int ConfigValueUInt(const char *devName, const char *blockName, const char *keyName, unsigned int *ret)
-{
-    const char *val = GetConfigValue(devName, blockName, keyName, "");
-    if(!val[0]) return 0;
-
-    *ret = strtoul(val, NULL, 0);
-    return 1;
-}
-
-int ConfigValueFloat(const char *devName, const char *blockName, const char *keyName, float *ret)
-{
-    const char *val = GetConfigValue(devName, blockName, keyName, "");
-    if(!val[0]) return 0;
-
-#ifdef HAVE_STRTOF
-    *ret = strtof(val, NULL);
-#else
-    *ret = (float)strtod(val, NULL);
-#endif
-    return 1;
-}
-
-int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret)
-{
-    const char *val = GetConfigValue(devName, blockName, keyName, "");
-    if(!val[0]) return 0;
-
-    *ret = (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||
-            strcasecmp(val, "on") == 0 || atoi(val) != 0);
-    return 1;
-}
-
-int GetConfigValueBool(const char *devName, const char *blockName, const char *keyName, int def)
-{
-    const char *val = GetConfigValue(devName, blockName, keyName, "");
-
-    if(!val[0]) return !!def;
-    return (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||
-            strcasecmp(val, "on") == 0 || atoi(val) != 0);
-}

Alc/alcRing.c

@@ -1,317 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <string.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "threads.h"
-#include "almalloc.h"
-#include "compat.h"
-
-
-/* NOTE: This lockless ringbuffer implementation is copied from JACK, extended
- * to include an element size. Consequently, parameters and return values for a
- * size or count is in 'elements', not bytes. Additionally, it only supports
- * single-consumer/single-provider operation. */
-struct ll_ringbuffer {
-    ATOMIC(size_t) write_ptr;
-    ATOMIC(size_t) read_ptr;
-    size_t size;
-    size_t size_mask;
-    size_t elem_size;
-    int mlocked;
-
-    alignas(16) char buf[];
-};
-
-/* Create a new ringbuffer to hold at least `sz' elements of `elem_sz' bytes.
- * The number of elements is rounded up to the next power of two. */
-ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz)
-{
-    ll_ringbuffer_t *rb;
-    ALuint power_of_two;
-
-    power_of_two = NextPowerOf2(sz);
-    if(power_of_two < sz)
-        return NULL;
-
-    rb = al_malloc(16, sizeof(*rb) + power_of_two*elem_sz);
-    if(!rb) return NULL;
-
-    ATOMIC_INIT(&rb->write_ptr, 0);
-    ATOMIC_INIT(&rb->read_ptr, 0);
-    rb->size = power_of_two;
-    rb->size_mask = rb->size - 1;
-    rb->elem_size = elem_sz;
-    rb->mlocked = 0;
-    return rb;
-}
-
-/* Free all data associated with the ringbuffer `rb'. */
-void ll_ringbuffer_free(ll_ringbuffer_t *rb)
-{
-    if(rb)
-    {
-#ifdef USE_MLOCK
-        if(rb->mlocked)
-            munlock(rb, sizeof(*rb) + rb->size*rb->elem_size);
-#endif /* USE_MLOCK */
-        al_free(rb);
-    }
-}
-
-/* Lock the data block of `rb' using the system call 'mlock'. */
-int ll_ringbuffer_mlock(ll_ringbuffer_t *rb)
-{
-#ifdef USE_MLOCK
-    if(!rb->mlocked && mlock(rb, sizeof(*rb) + rb->size*rb->elem_size))
-        return -1;
-#endif /* USE_MLOCK */
-    rb->mlocked = 1;
-    return 0;
-}
-
-/* Reset the read and write pointers to zero. This is not thread safe. */
-void ll_ringbuffer_reset(ll_ringbuffer_t *rb)
-{
-    ATOMIC_STORE(&rb->write_ptr, 0, almemory_order_release);
-    ATOMIC_STORE(&rb->read_ptr, 0, almemory_order_release);
-    memset(rb->buf, 0, rb->size*rb->elem_size);
-}
-
-/* Return the number of elements available for reading. This is the number of
- * elements in front of the read pointer and behind the write pointer. */
-size_t ll_ringbuffer_read_space(const ll_ringbuffer_t *rb)
-{
-    size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
-    size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
-    return (w-r) & rb->size_mask;
-}
-/* Return the number of elements available for writing. This is the number of
- * elements in front of the write pointer and behind the read pointer. */
-size_t ll_ringbuffer_write_space(const ll_ringbuffer_t *rb)
-{
-    size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
-    size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
-    return (r-w-1) & rb->size_mask;
-}
-
-/* The copying data reader. Copy at most `cnt' elements from `rb' to `dest'.
- * Returns the actual number of elements copied. */
-size_t ll_ringbuffer_read(ll_ringbuffer_t *rb, char *dest, size_t cnt)
-{
-    size_t read_ptr;
-    size_t free_cnt;
-    size_t cnt2;
-    size_t to_read;
-    size_t n1, n2;
-
-    free_cnt = ll_ringbuffer_read_space(rb);
-    if(free_cnt == 0) return 0;
-
-    to_read = (cnt > free_cnt) ? free_cnt : cnt;
-    read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask;
-
-    cnt2 = read_ptr + to_read;
-    if(cnt2 > rb->size)
-    {
-        n1 = rb->size - read_ptr;
-        n2 = cnt2 & rb->size_mask;
-    }
-    else
-    {
-        n1 = to_read;
-        n2 = 0;
-    }
-
-    memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size);
-    read_ptr += n1;
-    if(n2)
-    {
-        memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size],
-               n2*rb->elem_size);
-        read_ptr += n2;
-    }
-    ATOMIC_STORE(&rb->read_ptr, read_ptr, almemory_order_release);
-    return to_read;
-}
-
-/* The copying data reader w/o read pointer advance. Copy at most `cnt'
- * elements from `rb' to `dest'. Returns the actual number of elements copied.
- */
-size_t ll_ringbuffer_peek(ll_ringbuffer_t *rb, char *dest, size_t cnt)
-{
-    size_t free_cnt;
-    size_t cnt2;
-    size_t to_read;
-    size_t n1, n2;
-    size_t read_ptr;
-
-    free_cnt = ll_ringbuffer_read_space(rb);
-    if(free_cnt == 0) return 0;
-
-    to_read = (cnt > free_cnt) ? free_cnt : cnt;
-    read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask;
-
-    cnt2 = read_ptr + to_read;
-    if(cnt2 > rb->size)
-    {
-        n1 = rb->size - read_ptr;
-        n2 = cnt2 & rb->size_mask;
-    }
-    else
-    {
-        n1 = to_read;
-        n2 = 0;
-    }
-
-    memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size);
-    if(n2)
-    {
-        read_ptr += n1;
-        memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size],
-               n2*rb->elem_size);
-    }
-    return to_read;
-}
-
-/* The copying data writer. Copy at most `cnt' elements to `rb' from `src'.
- * Returns the actual number of elements copied. */
-size_t ll_ringbuffer_write(ll_ringbuffer_t *rb, const char *src, size_t cnt)
-{
-    size_t write_ptr;
-    size_t free_cnt;
-    size_t cnt2;
-    size_t to_write;
-    size_t n1, n2;
-
-    free_cnt = ll_ringbuffer_write_space(rb);
-    if(free_cnt == 0) return 0;
-
-    to_write = (cnt > free_cnt) ? free_cnt : cnt;
-    write_ptr = ATOMIC_LOAD(&rb->write_ptr, almemory_order_relaxed) & rb->size_mask;
-
-    cnt2 = write_ptr + to_write;
-    if(cnt2 > rb->size)
-    {
-        n1 = rb->size - write_ptr;
-        n2 = cnt2 & rb->size_mask;
-    }
-    else
-    {
-        n1 = to_write;
-        n2 = 0;
-    }
-
-    memcpy(&rb->buf[write_ptr*rb->elem_size], src, n1*rb->elem_size);
-    write_ptr += n1;
-    if(n2)
-    {
-        memcpy(&rb->buf[(write_ptr&rb->size_mask)*rb->elem_size], src + n1*rb->elem_size,
-               n2*rb->elem_size);
-        write_ptr += n2;
-    }
-    ATOMIC_STORE(&rb->write_ptr, write_ptr, almemory_order_release);
-    return to_write;
-}
-
-/* Advance the read pointer `cnt' places. */
-void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt)
-{
-    ATOMIC_ADD(&rb->read_ptr, cnt, almemory_order_acq_rel);
-}
-
-/* Advance the write pointer `cnt' places. */
-void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt)
-{
-    ATOMIC_ADD(&rb->write_ptr, cnt, almemory_order_acq_rel);
-}
-
-/* The non-copying data reader. `vec' is an array of two places. Set the values
- * at `vec' to hold the current readable data at `rb'. If the readable data is
- * in one segment the second segment has zero length. */
-void ll_ringbuffer_get_read_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t * vec)
-{
-    size_t free_cnt;
-    size_t cnt2;
-    size_t w, r;
-
-    w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
-    r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
-    w &= rb->size_mask;
-    r &= rb->size_mask;
-    free_cnt = (w-r) & rb->size_mask;
-
-    cnt2 = r + free_cnt;
-    if(cnt2 > rb->size)
-    {
-        /* Two part vector: the rest of the buffer after the current write ptr,
-         * plus some from the start of the buffer. */
-        vec[0].buf = (char*)&rb->buf[r*rb->elem_size];
-        vec[0].len = rb->size - r;
-        vec[1].buf = (char*)rb->buf;
-        vec[1].len = cnt2 & rb->size_mask;
-    }
-    else
-    {
-        /* Single part vector: just the rest of the buffer */
-        vec[0].buf = (char*)&rb->buf[r*rb->elem_size];
-        vec[0].len = free_cnt;
-        vec[1].buf = NULL;
-        vec[1].len = 0;
-    }
-}
-
-/* The non-copying data writer. `vec' is an array of two places. Set the values
- * at `vec' to hold the current writeable data at `rb'. If the writeable data
- * is in one segment the second segment has zero length. */
-void ll_ringbuffer_get_write_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t *vec)
-{
-    size_t free_cnt;
-    size_t cnt2;
-    size_t w, r;
-
-    w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
-    r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
-    w &= rb->size_mask;
-    r &= rb->size_mask;
-    free_cnt = (r-w-1) & rb->size_mask;
-
-    cnt2 = w + free_cnt;
-    if(cnt2 > rb->size)
-    {
-        /* Two part vector: the rest of the buffer after the current write ptr,
-         * plus some from the start of the buffer. */
-        vec[0].buf = (char*)&rb->buf[w*rb->elem_size];
-        vec[0].len = rb->size - w;
-        vec[1].buf = (char*)rb->buf;
-        vec[1].len = cnt2 & rb->size_mask;
-    }
-    else
-    {
-        vec[0].buf = (char*)&rb->buf[w*rb->elem_size];
-        vec[0].len = free_cnt;
-        vec[1].buf = NULL;
-        vec[1].len = 0;
-    }
-}

Alc/alstring.h

@@ -1,49 +0,0 @@
-#ifndef ALSTRING_H
-#define ALSTRING_H
-
-#include <string.h>
-
-#include "vector.h"
-
-
-typedef char al_string_char_type;
-TYPEDEF_VECTOR(al_string_char_type, al_string)
-TYPEDEF_VECTOR(al_string, vector_al_string)
-
-inline void alstr_reset(al_string *str)
-{ VECTOR_DEINIT(*str); }
-#define AL_STRING_INIT(_x)       do { (_x) = (al_string)NULL; } while(0)
-#define AL_STRING_INIT_STATIC()  ((al_string)NULL)
-#define AL_STRING_DEINIT(_x)     alstr_reset(&(_x))
-
-inline size_t alstr_length(const_al_string str)
-{ return VECTOR_SIZE(str); }
-
-inline ALboolean alstr_empty(const_al_string str)
-{ return alstr_length(str) == 0; }
-
-inline const al_string_char_type *alstr_get_cstr(const_al_string str)
-{ return str ? &VECTOR_FRONT(str) : ""; }
-
-void alstr_clear(al_string *str);
-
-int alstr_cmp(const_al_string str1, const_al_string str2);
-int alstr_cmp_cstr(const_al_string str1, const al_string_char_type *str2);
-
-void alstr_copy(al_string *str, const_al_string from);
-void alstr_copy_cstr(al_string *str, const al_string_char_type *from);
-void alstr_copy_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to);
-
-void alstr_append_char(al_string *str, const al_string_char_type c);
-void alstr_append_cstr(al_string *str, const al_string_char_type *from);
-void alstr_append_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to);
-
-#ifdef _WIN32
-#include <wchar.h>
-/* Windows-only methods to deal with WideChar strings. */
-void alstr_copy_wcstr(al_string *str, const wchar_t *from);
-void alstr_append_wcstr(al_string *str, const wchar_t *from);
-void alstr_append_wrange(al_string *str, const wchar_t *from, const wchar_t *to);
-#endif
-
-#endif /* ALSTRING_H */

Alc/ambdec.c

@@ -1,566 +0,0 @@
-
-#include "config.h"
-
-#include "ambdec.h"
-
-#include <stdio.h>
-#include <string.h>
-#include <ctype.h>
-
-#include "compat.h"
-
-
-static char *lstrip(char *line)
-{
-    while(isspace(line[0]))
-        line++;
-    return line;
-}
-
-static char *rstrip(char *line)
-{
-    size_t len = strlen(line);
-    while(len > 0 && isspace(line[len-1]))
-        len--;
-    line[len] = 0;
-    return line;
-}
-
-static int readline(FILE *f, char **output, size_t *maxlen)
-{
-    size_t len = 0;
-    int c;
-
-    while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n'))
-        ;
-    if(c == EOF)
-        return 0;
-
-    do {
-        if(len+1 >= *maxlen)
-        {
-            void *temp = NULL;
-            size_t newmax;
-
-            newmax = (*maxlen ? (*maxlen)<<1 : 32);
-            if(newmax > *maxlen)
-                temp = realloc(*output, newmax);
-            if(!temp)
-            {
-                ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen);
-                return 0;
-            }
-
-            *output = temp;
-            *maxlen = newmax;
-        }
-        (*output)[len++] = c;
-        (*output)[len] = '\0';
-    } while((c=fgetc(f)) != EOF && c != '\r' && c != '\n');
-
-    return 1;
-}
-
-
-/* Custom strtok_r, since we can't rely on it existing. */
-static char *my_strtok_r(char *str, const char *delim, char **saveptr)
-{
-    /* Sanity check and update internal pointer. */
-    if(!saveptr || !delim) return NULL;
-    if(str) *saveptr = str;
-    str = *saveptr;
-
-    /* Nothing more to do with this string. */
-    if(!str) return NULL;
-
-    /* Find the first non-delimiter character. */
-    while(*str != '\0' && strchr(delim, *str) != NULL)
-        str++;
-    if(*str == '\0')
-    {
-        /* End of string. */
-        *saveptr = NULL;
-        return NULL;
-    }
-
-    /* Find the next delimiter character. */
-    *saveptr = strpbrk(str, delim);
-    if(*saveptr) *((*saveptr)++) = '\0';
-
-    return str;
-}
-
-static char *read_int(ALint *num, const char *line, int base)
-{
-    char *end;
-    *num = strtol(line, &end, base);
-    if(end && *end != '\0')
-        end = lstrip(end);
-    return end;
-}
-
-static char *read_uint(ALuint *num, const char *line, int base)
-{
-    char *end;
-    *num = strtoul(line, &end, base);
-    if(end && *end != '\0')
-        end = lstrip(end);
-    return end;
-}
-
-static char *read_float(ALfloat *num, const char *line)
-{
-    char *end;
-#ifdef HAVE_STRTOF
-    *num = strtof(line, &end);
-#else
-    *num = (ALfloat)strtod(line, &end);
-#endif
-    if(end && *end != '\0')
-        end = lstrip(end);
-    return end;
-}
-
-
-char *read_clipped_line(FILE *f, char **buffer, size_t *maxlen)
-{
-    while(readline(f, buffer, maxlen))
-    {
-        char *line, *comment;
-
-        line = lstrip(*buffer);
-        comment = strchr(line, '#');
-        if(comment) *(comment++) = 0;
-
-        line = rstrip(line);
-        if(line[0]) return line;
-    }
-    return NULL;
-}
-
-static int load_ambdec_speakers(AmbDecConf *conf, FILE *f, char **buffer, size_t *maxlen, char **saveptr)
-{
-    ALsizei cur = 0;
-    while(cur < conf->NumSpeakers)
-    {
-        const char *cmd = my_strtok_r(NULL, " \t", saveptr);
-        if(!cmd)
-        {
-            char *line = read_clipped_line(f, buffer, maxlen);
-            if(!line)
-            {
-                ERR("Unexpected end of file\n");
-                return 0;
-            }
-            cmd = my_strtok_r(line, " \t", saveptr);
-        }
-
-        if(strcmp(cmd, "add_spkr") == 0)
-        {
-            const char *name = my_strtok_r(NULL, " \t", saveptr);
-            const char *dist = my_strtok_r(NULL, " \t", saveptr);
-            const char *az = my_strtok_r(NULL, " \t", saveptr);
-            const char *elev = my_strtok_r(NULL, " \t", saveptr);
-            const char *conn = my_strtok_r(NULL, " \t", saveptr);
-
-            if(!name) WARN("Name not specified for speaker %u\n", cur+1);
-            else alstr_copy_cstr(&conf->Speakers[cur].Name, name);
-            if(!dist) WARN("Distance not specified for speaker %u\n", cur+1);
-            else read_float(&conf->Speakers[cur].Distance, dist);
-            if(!az) WARN("Azimuth not specified for speaker %u\n", cur+1);
-            else read_float(&conf->Speakers[cur].Azimuth, az);
-            if(!elev) WARN("Elevation not specified for speaker %u\n", cur+1);
-            else read_float(&conf->Speakers[cur].Elevation, elev);
-            if(!conn) TRACE("Connection not specified for speaker %u\n", cur+1);
-            else alstr_copy_cstr(&conf->Speakers[cur].Connection, conn);
-
-            cur++;
-        }
-        else
-        {
-            ERR("Unexpected speakers command: %s\n", cmd);
-            return 0;
-        }
-
-        cmd = my_strtok_r(NULL, " \t", saveptr);
-        if(cmd)
-        {
-            ERR("Unexpected junk on line: %s\n", cmd);
-            return 0;
-        }
-    }
-
-    return 1;
-}
-
-static int load_ambdec_matrix(ALfloat *gains, ALfloat (*matrix)[MAX_AMBI_COEFFS], ALsizei maxrow, FILE *f, char **buffer, size_t *maxlen, char **saveptr)
-{
-    int gotgains = 0;
-    ALsizei cur = 0;
-    while(cur < maxrow)
-    {
-        const char *cmd = my_strtok_r(NULL, " \t", saveptr);
-        if(!cmd)
-        {
-            char *line = read_clipped_line(f, buffer, maxlen);
-            if(!line)
-            {
-                ERR("Unexpected end of file\n");
-                return 0;
-            }
-            cmd = my_strtok_r(line, " \t", saveptr);
-        }
-
-        if(strcmp(cmd, "order_gain") == 0)
-        {
-            ALuint curgain = 0;
-            char *line;
-            while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL)
-            {
-                ALfloat value;
-                line = read_float(&value, line);
-                if(line && *line != '\0')
-                {
-                    ERR("Extra junk on gain %u: %s\n", curgain+1, line);
-                    return 0;
-                }
-                if(curgain < MAX_AMBI_ORDER+1)
-                    gains[curgain] = value;
-                curgain++;
-            }
-            while(curgain < MAX_AMBI_ORDER+1)
-                gains[curgain++] = 0.0f;
-            gotgains = 1;
-        }
-        else if(strcmp(cmd, "add_row") == 0)
-        {
-            ALuint curidx = 0;
-            char *line;
-            while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL)
-            {
-                ALfloat value;
-                line = read_float(&value, line);
-                if(line && *line != '\0')
-                {
-                    ERR("Extra junk on matrix element %ux%u: %s\n", cur, curidx, line);
-                    return 0;
-                }
-                if(curidx < MAX_AMBI_COEFFS)
-                    matrix[cur][curidx] = value;
-                curidx++;
-            }
-            while(curidx < MAX_AMBI_COEFFS)
-                matrix[cur][curidx++] = 0.0f;
-            cur++;
-        }
-        else
-        {
-            ERR("Unexpected speakers command: %s\n", cmd);
-            return 0;
-        }
-
-        cmd = my_strtok_r(NULL, " \t", saveptr);
-        if(cmd)
-        {
-            ERR("Unexpected junk on line: %s\n", cmd);
-            return 0;
-        }
-    }
-
-    if(!gotgains)
-    {
-        ERR("Matrix order_gain not specified\n");
-        return 0;
-    }
-
-    return 1;
-}
-
-void ambdec_init(AmbDecConf *conf)
-{
-    ALsizei i;
-
-    memset(conf, 0, sizeof(*conf));
-    AL_STRING_INIT(conf->Description);
-    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-    {
-        AL_STRING_INIT(conf->Speakers[i].Name);
-        AL_STRING_INIT(conf->Speakers[i].Connection);
-    }
-}
-
-void ambdec_deinit(AmbDecConf *conf)
-{
-    ALsizei i;
-
-    alstr_reset(&conf->Description);
-    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-    {
-        alstr_reset(&conf->Speakers[i].Name);
-        alstr_reset(&conf->Speakers[i].Connection);
-    }
-    memset(conf, 0, sizeof(*conf));
-}
-
-int ambdec_load(AmbDecConf *conf, const char *fname)
-{
-    char *buffer = NULL;
-    size_t maxlen = 0;
-    char *line;
-    FILE *f;
-
-    f = al_fopen(fname, "r");
-    if(!f)
-    {
-        ERR("Failed to open: %s\n", fname);
-        return 0;
-    }
-
-    while((line=read_clipped_line(f, &buffer, &maxlen)) != NULL)
-    {
-        char *saveptr;
-        char *command;
-
-        command = my_strtok_r(line, "/ \t", &saveptr);
-        if(!command)
-        {
-            ERR("Malformed line: %s\n", line);
-            goto fail;
-        }
-
-        if(strcmp(command, "description") == 0)
-        {
-            char *value = my_strtok_r(NULL, "", &saveptr);
-            alstr_copy_cstr(&conf->Description, lstrip(value));
-        }
-        else if(strcmp(command, "version") == 0)
-        {
-            line = my_strtok_r(NULL, "", &saveptr);
-            line = read_uint(&conf->Version, line, 10);
-            if(line && *line != '\0')
-            {
-                ERR("Extra junk after version: %s\n", line);
-                goto fail;
-            }
-            if(conf->Version != 3)
-            {
-                ERR("Unsupported version: %u\n", conf->Version);
-                goto fail;
-            }
-        }
-        else if(strcmp(command, "dec") == 0)
-        {
-            const char *dec = my_strtok_r(NULL, "/ \t", &saveptr);
-            if(strcmp(dec, "chan_mask") == 0)
-            {
-                line = my_strtok_r(NULL, "", &saveptr);
-                line = read_uint(&conf->ChanMask, line, 16);
-                if(line && *line != '\0')
-                {
-                    ERR("Extra junk after mask: %s\n", line);
-                    goto fail;
-                }
-            }
-            else if(strcmp(dec, "freq_bands") == 0)
-            {
-                line = my_strtok_r(NULL, "", &saveptr);
-                line = read_uint(&conf->FreqBands, line, 10);
-                if(line && *line != '\0')
-                {
-                    ERR("Extra junk after freq_bands: %s\n", line);
-                    goto fail;
-                }
-                if(conf->FreqBands != 1 && conf->FreqBands != 2)
-                {
-                    ERR("Invalid freq_bands value: %u\n", conf->FreqBands);
-                    goto fail;
-                }
-            }
-            else if(strcmp(dec, "speakers") == 0)
-            {
-                line = my_strtok_r(NULL, "", &saveptr);
-                line = read_int(&conf->NumSpeakers, line, 10);
-                if(line && *line != '\0')
-                {
-                    ERR("Extra junk after speakers: %s\n", line);
-                    goto fail;
-                }
-                if(conf->NumSpeakers > MAX_OUTPUT_CHANNELS)
-                {
-                    ERR("Unsupported speaker count: %u\n", conf->NumSpeakers);
-                    goto fail;
-                }
-            }
-            else if(strcmp(dec, "coeff_scale") == 0)
-            {
-                line = my_strtok_r(NULL, " \t", &saveptr);
-                if(strcmp(line, "n3d") == 0)
-                    conf->CoeffScale = ADS_N3D;
-                else if(strcmp(line, "sn3d") == 0)
-                    conf->CoeffScale = ADS_SN3D;
-                else if(strcmp(line, "fuma") == 0)
-                    conf->CoeffScale = ADS_FuMa;
-                else
-                {
-                    ERR("Unsupported coeff scale: %s\n", line);
-                    goto fail;
-                }
-            }
-            else
-            {
-                ERR("Unexpected /dec option: %s\n", dec);
-                goto fail;
-            }
-        }
-        else if(strcmp(command, "opt") == 0)
-        {
-            const char *opt = my_strtok_r(NULL, "/ \t", &saveptr);
-            if(strcmp(opt, "xover_freq") == 0)
-            {
-                line = my_strtok_r(NULL, "", &saveptr);
-                line = read_float(&conf->XOverFreq, line);
-                if(line && *line != '\0')
-                {
-                    ERR("Extra junk after xover_freq: %s\n", line);
-                    goto fail;
-                }
-            }
-            else if(strcmp(opt, "xover_ratio") == 0)
-            {
-                line = my_strtok_r(NULL, "", &saveptr);
-                line = read_float(&conf->XOverRatio, line);
-                if(line && *line != '\0')
-                {
-                    ERR("Extra junk after xover_ratio: %s\n", line);
-                    goto fail;
-                }
-            }
-            else if(strcmp(opt, "input_scale") == 0 || strcmp(opt, "nfeff_comp") == 0 ||
-                    strcmp(opt, "delay_comp") == 0 || strcmp(opt, "level_comp") == 0)
-            {
-                /* Unused */
-                my_strtok_r(NULL, " \t", &saveptr);
-            }
-            else
-            {
-                ERR("Unexpected /opt option: %s\n", opt);
-                goto fail;
-            }
-        }
-        else if(strcmp(command, "speakers") == 0)
-        {
-            const char *value = my_strtok_r(NULL, "/ \t", &saveptr);
-            if(strcmp(value, "{") != 0)
-            {
-                ERR("Expected { after %s command, got %s\n", command, value);
-                goto fail;
-            }
-            if(!load_ambdec_speakers(conf, f, &buffer, &maxlen, &saveptr))
-                goto fail;
-            value = my_strtok_r(NULL, "/ \t", &saveptr);
-            if(!value)
-            {
-                line = read_clipped_line(f, &buffer, &maxlen);
-                if(!line)
-                {
-                    ERR("Unexpected end of file\n");
-                    goto fail;
-                }
-                value = my_strtok_r(line, "/ \t", &saveptr);
-            }
-            if(strcmp(value, "}") != 0)
-            {
-                ERR("Expected } after speaker definitions, got %s\n", value);
-                goto fail;
-            }
-        }
-        else if(strcmp(command, "lfmatrix") == 0 || strcmp(command, "hfmatrix") == 0 ||
-                strcmp(command, "matrix") == 0)
-        {
-            const char *value = my_strtok_r(NULL, "/ \t", &saveptr);
-            if(strcmp(value, "{") != 0)
-            {
-                ERR("Expected { after %s command, got %s\n", command, value);
-                goto fail;
-            }
-            if(conf->FreqBands == 1)
-            {
-                if(strcmp(command, "matrix") != 0)
-                {
-                    ERR("Unexpected \"%s\" type for a single-band decoder\n", command);
-                    goto fail;
-                }
-                if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers,
-                                       f, &buffer, &maxlen, &saveptr))
-                    goto fail;
-            }
-            else
-            {
-                if(strcmp(command, "lfmatrix") == 0)
-                {
-                    if(!load_ambdec_matrix(conf->LFOrderGain, conf->LFMatrix, conf->NumSpeakers,
-                                           f, &buffer, &maxlen, &saveptr))
-                        goto fail;
-                }
-                else if(strcmp(command, "hfmatrix") == 0)
-                {
-                    if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers,
-                                           f, &buffer, &maxlen, &saveptr))
-                        goto fail;
-                }
-                else
-                {
-                    ERR("Unexpected \"%s\" type for a dual-band decoder\n", command);
-                    goto fail;
-                }
-            }
-            value = my_strtok_r(NULL, "/ \t", &saveptr);
-            if(!value)
-            {
-                line = read_clipped_line(f, &buffer, &maxlen);
-                if(!line)
-                {
-                    ERR("Unexpected end of file\n");
-                    goto fail;
-                }
-                value = my_strtok_r(line, "/ \t", &saveptr);
-            }
-            if(strcmp(value, "}") != 0)
-            {
-                ERR("Expected } after matrix definitions, got %s\n", value);
-                goto fail;
-            }
-        }
-        else if(strcmp(command, "end") == 0)
-        {
-            line = my_strtok_r(NULL, "/ \t", &saveptr);
-            if(line)
-            {
-                ERR("Unexpected junk on end: %s\n", line);
-                goto fail;
-            }
-
-            fclose(f);
-            free(buffer);
-            return 1;
-        }
-        else
-        {
-            ERR("Unexpected command: %s\n", command);
-            goto fail;
-        }
-
-        line = my_strtok_r(NULL, "/ \t", &saveptr);
-        if(line)
-        {
-            ERR("Unexpected junk on line: %s\n", line);
-            goto fail;
-        }
-    }
-    ERR("Unexpected end of file\n");
-
-fail:
-    fclose(f);
-    free(buffer);
-    return 0;
-}

Alc/ambdec.h

@@ -1,46 +0,0 @@
-#ifndef AMBDEC_H
-#define AMBDEC_H
-
-#include "alstring.h"
-#include "alMain.h"
-
-/* Helpers to read .ambdec configuration files. */
-
-enum AmbDecScaleType {
-    ADS_N3D,
-    ADS_SN3D,
-    ADS_FuMa,
-};
-typedef struct AmbDecConf {
-    al_string Description;
-    ALuint Version; /* Must be 3 */
-
-    ALuint ChanMask;
-    ALuint FreqBands; /* Must be 1 or 2 */
-    ALsizei NumSpeakers;
-    enum AmbDecScaleType CoeffScale;
-
-    ALfloat XOverFreq;
-    ALfloat XOverRatio;
-
-    struct {
-        al_string Name;
-        ALfloat Distance;
-        ALfloat Azimuth;
-        ALfloat Elevation;
-        al_string Connection;
-    } Speakers[MAX_OUTPUT_CHANNELS];
-
-    /* Unused when FreqBands == 1 */
-    ALfloat LFOrderGain[MAX_AMBI_ORDER+1];
-    ALfloat LFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
-
-    ALfloat HFOrderGain[MAX_AMBI_ORDER+1];
-    ALfloat HFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
-} AmbDecConf;
-
-void ambdec_init(AmbDecConf *conf);
-void ambdec_deinit(AmbDecConf *conf);
-int ambdec_load(AmbDecConf *conf, const char *fname);
-
-#endif /* AMBDEC_H */

Alc/backends/base.c

@@ -1,81 +0,0 @@
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alu.h"
-
-#include "backends/base.h"
-
-
-extern inline ALuint64 GetDeviceClockTime(ALCdevice *device);
-
-/* Base ALCbackend method implementations. */
-void ALCbackend_Construct(ALCbackend *self, ALCdevice *device)
-{
-    int ret = almtx_init(&self->mMutex, almtx_recursive);
-    assert(ret == althrd_success);
-    self->mDevice = device;
-}
-
-void ALCbackend_Destruct(ALCbackend *self)
-{
-    almtx_destroy(&self->mMutex);
-}
-
-ALCboolean ALCbackend_reset(ALCbackend* UNUSED(self))
-{
-    return ALC_FALSE;
-}
-
-ALCenum ALCbackend_captureSamples(ALCbackend* UNUSED(self), void* UNUSED(buffer), ALCuint UNUSED(samples))
-{
-    return ALC_INVALID_DEVICE;
-}
-
-ALCuint ALCbackend_availableSamples(ALCbackend* UNUSED(self))
-{
-    return 0;
-}
-
-ClockLatency ALCbackend_getClockLatency(ALCbackend *self)
-{
-    ALCdevice *device = self->mDevice;
-    ALuint refcount;
-    ClockLatency ret;
-
-    do {
-        while(((refcount=ATOMIC_LOAD(&device->MixCount, almemory_order_acquire))&1))
-            althrd_yield();
-        ret.ClockTime = GetDeviceClockTime(device);
-        ATOMIC_THREAD_FENCE(almemory_order_acquire);
-    } while(refcount != ATOMIC_LOAD(&device->MixCount, almemory_order_relaxed));
-
-    /* NOTE: The device will generally have about all but one periods filled at
-     * any given time during playback. Without a more accurate measurement from
-     * the output, this is an okay approximation.
-     */
-    ret.Latency = device->UpdateSize * DEVICE_CLOCK_RES / device->Frequency *
-                  maxu(device->NumUpdates-1, 1);
-
-    return ret;
-}
-
-void ALCbackend_lock(ALCbackend *self)
-{
-    int ret = almtx_lock(&self->mMutex);
-    assert(ret == althrd_success);
-}
-
-void ALCbackend_unlock(ALCbackend *self)
-{
-    int ret = almtx_unlock(&self->mMutex);
-    assert(ret == althrd_success);
-}
-
-
-/* Base ALCbackendFactory method implementations. */
-void ALCbackendFactory_deinit(ALCbackendFactory* UNUSED(self))
-{
-}

Alc/backends/base.h

@@ -1,155 +0,0 @@
-#ifndef AL_BACKENDS_BASE_H
-#define AL_BACKENDS_BASE_H
-
-#include "alMain.h"
-#include "threads.h"
-
-
-typedef struct ClockLatency {
-    ALint64 ClockTime;
-    ALint64 Latency;
-} ClockLatency;
-
-/* Helper to get the current clock time from the device's ClockBase, and
- * SamplesDone converted from the sample rate.
- */
-inline ALuint64 GetDeviceClockTime(ALCdevice *device)
-{
-    return device->ClockBase + (device->SamplesDone * DEVICE_CLOCK_RES /
-                                device->Frequency);
-}
-
-
-struct ALCbackendVtable;
-
-typedef struct ALCbackend {
-    const struct ALCbackendVtable *vtbl;
-
-    ALCdevice *mDevice;
-
-    almtx_t mMutex;
-} ALCbackend;
-
-void ALCbackend_Construct(ALCbackend *self, ALCdevice *device);
-void ALCbackend_Destruct(ALCbackend *self);
-ALCboolean ALCbackend_reset(ALCbackend *self);
-ALCenum ALCbackend_captureSamples(ALCbackend *self, void *buffer, ALCuint samples);
-ALCuint ALCbackend_availableSamples(ALCbackend *self);
-ClockLatency ALCbackend_getClockLatency(ALCbackend *self);
-void ALCbackend_lock(ALCbackend *self);
-void ALCbackend_unlock(ALCbackend *self);
-
-struct ALCbackendVtable {
-    void (*const Destruct)(ALCbackend*);
-
-    ALCenum (*const open)(ALCbackend*, const ALCchar*);
-    void (*const close)(ALCbackend*);
-
-    ALCboolean (*const reset)(ALCbackend*);
-    ALCboolean (*const start)(ALCbackend*);
-    void (*const stop)(ALCbackend*);
-
-    ALCenum (*const captureSamples)(ALCbackend*, void*, ALCuint);
-    ALCuint (*const availableSamples)(ALCbackend*);
-
-    ClockLatency (*const getClockLatency)(ALCbackend*);
-
-    void (*const lock)(ALCbackend*);
-    void (*const unlock)(ALCbackend*);
-
-    void (*const Delete)(void*);
-};
-
-#define DEFINE_ALCBACKEND_VTABLE(T)                                           \
-DECLARE_THUNK(T, ALCbackend, void, Destruct)                                  \
-DECLARE_THUNK1(T, ALCbackend, ALCenum, open, const ALCchar*)                  \
-DECLARE_THUNK(T, ALCbackend, void, close)                                     \
-DECLARE_THUNK(T, ALCbackend, ALCboolean, reset)                               \
-DECLARE_THUNK(T, ALCbackend, ALCboolean, start)                               \
-DECLARE_THUNK(T, ALCbackend, void, stop)                                      \
-DECLARE_THUNK2(T, ALCbackend, ALCenum, captureSamples, void*, ALCuint)        \
-DECLARE_THUNK(T, ALCbackend, ALCuint, availableSamples)                       \
-DECLARE_THUNK(T, ALCbackend, ClockLatency, getClockLatency)                   \
-DECLARE_THUNK(T, ALCbackend, void, lock)                                      \
-DECLARE_THUNK(T, ALCbackend, void, unlock)                                    \
-static void T##_ALCbackend_Delete(void *ptr)                                  \
-{ T##_Delete(STATIC_UPCAST(T, ALCbackend, (ALCbackend*)ptr)); }               \
-                                                                              \
-static const struct ALCbackendVtable T##_ALCbackend_vtable = {                \
-    T##_ALCbackend_Destruct,                                                  \
-                                                                              \
-    T##_ALCbackend_open,                                                      \
-    T##_ALCbackend_close,                                                     \
-    T##_ALCbackend_reset,                                                     \
-    T##_ALCbackend_start,                                                     \
-    T##_ALCbackend_stop,                                                      \
-    T##_ALCbackend_captureSamples,                                            \
-    T##_ALCbackend_availableSamples,                                          \
-    T##_ALCbackend_getClockLatency,                                           \
-    T##_ALCbackend_lock,                                                      \
-    T##_ALCbackend_unlock,                                                    \
-                                                                              \
-    T##_ALCbackend_Delete,                                                    \
-}
-
-
-typedef enum ALCbackend_Type {
-    ALCbackend_Playback,
-    ALCbackend_Capture,
-    ALCbackend_Loopback
-} ALCbackend_Type;
-
-
-struct ALCbackendFactoryVtable;
-
-typedef struct ALCbackendFactory {
-    const struct ALCbackendFactoryVtable *vtbl;
-} ALCbackendFactory;
-
-void ALCbackendFactory_deinit(ALCbackendFactory *self);
-
-struct ALCbackendFactoryVtable {
-    ALCboolean (*const init)(ALCbackendFactory *self);
-    void (*const deinit)(ALCbackendFactory *self);
-
-    ALCboolean (*const querySupport)(ALCbackendFactory *self, ALCbackend_Type type);
-
-    void (*const probe)(ALCbackendFactory *self, enum DevProbe type);
-
-    ALCbackend* (*const createBackend)(ALCbackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-};
-
-#define DEFINE_ALCBACKENDFACTORY_VTABLE(T)                                    \
-DECLARE_THUNK(T, ALCbackendFactory, ALCboolean, init)                         \
-DECLARE_THUNK(T, ALCbackendFactory, void, deinit)                             \
-DECLARE_THUNK1(T, ALCbackendFactory, ALCboolean, querySupport, ALCbackend_Type) \
-DECLARE_THUNK1(T, ALCbackendFactory, void, probe, enum DevProbe)              \
-DECLARE_THUNK2(T, ALCbackendFactory, ALCbackend*, createBackend, ALCdevice*, ALCbackend_Type) \
-                                                                              \
-static const struct ALCbackendFactoryVtable T##_ALCbackendFactory_vtable = {  \
-    T##_ALCbackendFactory_init,                                               \
-    T##_ALCbackendFactory_deinit,                                             \
-    T##_ALCbackendFactory_querySupport,                                       \
-    T##_ALCbackendFactory_probe,                                              \
-    T##_ALCbackendFactory_createBackend,                                      \
-}
-
-
-ALCbackendFactory *ALCpulseBackendFactory_getFactory(void);
-ALCbackendFactory *ALCalsaBackendFactory_getFactory(void);
-ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
-ALCbackendFactory *ALCossBackendFactory_getFactory(void);
-ALCbackendFactory *ALCjackBackendFactory_getFactory(void);
-ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void);
-ALCbackendFactory *ALCsndioBackendFactory_getFactory(void);
-ALCbackendFactory *ALCqsaBackendFactory_getFactory(void);
-ALCbackendFactory *ALCmmdevBackendFactory_getFactory(void);
-ALCbackendFactory *ALCdsoundBackendFactory_getFactory(void);
-ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void);
-ALCbackendFactory *ALCportBackendFactory_getFactory(void);
-ALCbackendFactory *ALCopenslBackendFactory_getFactory(void);
-ALCbackendFactory *ALCnullBackendFactory_getFactory(void);
-ALCbackendFactory *ALCwaveBackendFactory_getFactory(void);
-ALCbackendFactory *ALCloopbackFactory_getFactory(void);
-
-#endif /* AL_BACKENDS_BASE_H */

Alc/backends/coreaudio.c

@@ -1,828 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <alloca.h>
-
-#include "alMain.h"
-#include "alu.h"
-
-#include <CoreServices/CoreServices.h>
-#include <unistd.h>
-#include <AudioUnit/AudioUnit.h>
-#include <AudioToolbox/AudioToolbox.h>
-
-#include "backends/base.h"
-
-
-typedef struct {
-    AudioUnit audioUnit;
-
-    ALuint frameSize;
-    ALdouble sampleRateRatio;              // Ratio of hardware sample rate / requested sample rate
-    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
-
-    AudioConverterRef audioConverter;      // Sample rate converter if needed
-    AudioBufferList *bufferList;           // Buffer for data coming from the input device
-    ALCvoid *resampleBuffer;               // Buffer for returned RingBuffer data when resampling
-
-    ll_ringbuffer_t *ring;
-} ca_data;
-
-static const ALCchar ca_device[] = "CoreAudio Default";
-
-
-static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
-{
-    AudioBufferList *list;
-
-    list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
-    if(list)
-    {
-        list->mNumberBuffers = 1;
-
-        list->mBuffers[0].mNumberChannels = channelCount;
-        list->mBuffers[0].mDataByteSize = byteSize;
-        list->mBuffers[0].mData = malloc(byteSize);
-        if(list->mBuffers[0].mData == NULL)
-        {
-            free(list);
-            list = NULL;
-        }
-    }
-    return list;
-}
-
-static void destroy_buffer_list(AudioBufferList* list)
-{
-    if(list)
-    {
-        UInt32 i;
-        for(i = 0;i < list->mNumberBuffers;i++)
-            free(list->mBuffers[i].mData);
-        free(list);
-    }
-}
-
-
-typedef struct ALCcoreAudioPlayback {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    AudioUnit audioUnit;
-
-    ALuint frameSize;
-    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
-} ALCcoreAudioPlayback;
-
-static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device);
-static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self);
-static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name);
-static void ALCcoreAudioPlayback_close(ALCcoreAudioPlayback *self);
-static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self);
-static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self);
-static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self);
-static DECLARE_FORWARD2(ALCcoreAudioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
-static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioPlayback)
-
-DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioPlayback);
-
-
-static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self);
-
-    self->frameSize = 0;
-    memset(&self->format, 0, sizeof(self->format));
-}
-
-static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
-{
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon,
-  AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp),
-  UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData)
-{
-    ALCcoreAudioPlayback *self = inRefCon;
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-
-    ALCdevice_Lock(device);
-    aluMixData(device, ioData->mBuffers[0].mData,
-               ioData->mBuffers[0].mDataByteSize / self->frameSize);
-    ALCdevice_Unlock(device);
-
-    return noErr;
-}
-
-
-static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-    AudioComponentDescription desc;
-    AudioComponent comp;
-    OSStatus err;
-
-    if(!name)
-        name = ca_device;
-    else if(strcmp(name, ca_device) != 0)
-        return ALC_INVALID_VALUE;
-
-    /* open the default output unit */
-    desc.componentType = kAudioUnitType_Output;
-    desc.componentSubType = kAudioUnitSubType_DefaultOutput;
-    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-    desc.componentFlags = 0;
-    desc.componentFlagsMask = 0;
-
-    comp = AudioComponentFindNext(NULL, &desc);
-    if(comp == NULL)
-    {
-        ERR("AudioComponentFindNext failed\n");
-        return ALC_INVALID_VALUE;
-    }
-
-    err = AudioComponentInstanceNew(comp, &self->audioUnit);
-    if(err != noErr)
-    {
-        ERR("AudioComponentInstanceNew failed\n");
-        return ALC_INVALID_VALUE;
-    }
-
-    /* init and start the default audio unit... */
-    err = AudioUnitInitialize(self->audioUnit);
-    if(err != noErr)
-    {
-        ERR("AudioUnitInitialize failed\n");
-        AudioComponentInstanceDispose(self->audioUnit);
-        return ALC_INVALID_VALUE;
-    }
-
-    alstr_copy_cstr(&device->DeviceName, name);
-    return ALC_NO_ERROR;
-}
-
-static void ALCcoreAudioPlayback_close(ALCcoreAudioPlayback *self)
-{
-    AudioUnitUninitialize(self->audioUnit);
-    AudioComponentInstanceDispose(self->audioUnit);
-}
-
-static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-    AudioStreamBasicDescription streamFormat;
-    AURenderCallbackStruct input;
-    OSStatus err;
-    UInt32 size;
-
-    err = AudioUnitUninitialize(self->audioUnit);
-    if(err != noErr)
-        ERR("-- AudioUnitUninitialize failed.\n");
-
-    /* retrieve default output unit's properties (output side) */
-    size = sizeof(AudioStreamBasicDescription);
-    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
-    if(err != noErr || size != sizeof(AudioStreamBasicDescription))
-    {
-        ERR("AudioUnitGetProperty failed\n");
-        return ALC_FALSE;
-    }
-
-#if 0
-    TRACE("Output streamFormat of default output unit -\n");
-    TRACE("  streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket);
-    TRACE("  streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame);
-    TRACE("  streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel);
-    TRACE("  streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket);
-    TRACE("  streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame);
-    TRACE("  streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate);
-#endif
-
-    /* set default output unit's input side to match output side */
-    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        return ALC_FALSE;
-    }
-
-    if(device->Frequency != streamFormat.mSampleRate)
-    {
-        device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates *
-                                      streamFormat.mSampleRate /
-                                      device->Frequency);
-        device->Frequency = streamFormat.mSampleRate;
-    }
-
-    /* FIXME: How to tell what channels are what in the output device, and how
-     * to specify what we're giving?  eg, 6.0 vs 5.1 */
-    switch(streamFormat.mChannelsPerFrame)
-    {
-        case 1:
-            device->FmtChans = DevFmtMono;
-            break;
-        case 2:
-            device->FmtChans = DevFmtStereo;
-            break;
-        case 4:
-            device->FmtChans = DevFmtQuad;
-            break;
-        case 6:
-            device->FmtChans = DevFmtX51;
-            break;
-        case 7:
-            device->FmtChans = DevFmtX61;
-            break;
-        case 8:
-            device->FmtChans = DevFmtX71;
-            break;
-        default:
-            ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
-            device->FmtChans = DevFmtStereo;
-            streamFormat.mChannelsPerFrame = 2;
-            break;
-    }
-    SetDefaultWFXChannelOrder(device);
-
-    /* use channel count and sample rate from the default output unit's current
-     * parameters, but reset everything else */
-    streamFormat.mFramesPerPacket = 1;
-    streamFormat.mFormatFlags = 0;
-    switch(device->FmtType)
-    {
-        case DevFmtUByte:
-            device->FmtType = DevFmtByte;
-            /* fall-through */
-        case DevFmtByte:
-            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
-            streamFormat.mBitsPerChannel = 8;
-            break;
-        case DevFmtUShort:
-            device->FmtType = DevFmtShort;
-            /* fall-through */
-        case DevFmtShort:
-            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
-            streamFormat.mBitsPerChannel = 16;
-            break;
-        case DevFmtUInt:
-            device->FmtType = DevFmtInt;
-            /* fall-through */
-        case DevFmtInt:
-            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
-            streamFormat.mBitsPerChannel = 32;
-            break;
-        case DevFmtFloat:
-            streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat;
-            streamFormat.mBitsPerChannel = 32;
-            break;
-    }
-    streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame *
-                                  streamFormat.mBitsPerChannel / 8;
-    streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame;
-    streamFormat.mFormatID = kAudioFormatLinearPCM;
-    streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
-                                 kLinearPCMFormatFlagIsPacked;
-
-    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        return ALC_FALSE;
-    }
-
-    /* setup callback */
-    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-    input.inputProc = ALCcoreAudioPlayback_MixerProc;
-    input.inputProcRefCon = self;
-
-    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        return ALC_FALSE;
-    }
-
-    /* init the default audio unit... */
-    err = AudioUnitInitialize(self->audioUnit);
-    if(err != noErr)
-    {
-        ERR("AudioUnitInitialize failed\n");
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self)
-{
-    OSStatus err = AudioOutputUnitStart(self->audioUnit);
-    if(err != noErr)
-    {
-        ERR("AudioOutputUnitStart failed\n");
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self)
-{
-    OSStatus err = AudioOutputUnitStop(self->audioUnit);
-    if(err != noErr)
-        ERR("AudioOutputUnitStop failed\n");
-}
-
-
-
-
-typedef struct ALCcoreAudioCapture {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    AudioUnit audioUnit;
-
-    ALuint frameSize;
-    ALdouble sampleRateRatio;              // Ratio of hardware sample rate / requested sample rate
-    AudioStreamBasicDescription format;    // This is the OpenAL format as a CoreAudio ASBD
-
-    AudioConverterRef audioConverter;      // Sample rate converter if needed
-    AudioBufferList *bufferList;           // Buffer for data coming from the input device
-    ALCvoid *resampleBuffer;               // Buffer for returned RingBuffer data when resampling
-
-    ll_ringbuffer_t *ring;
-} ALCcoreAudioCapture;
-
-static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device);
-static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self);
-static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name);
-static void ALCcoreAudioCapture_close(ALCcoreAudioCapture *self);
-static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ALCboolean, reset)
-static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self);
-static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self);
-static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples);
-static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self);
-static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioCapture)
-
-DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioCapture);
-
-
-static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self);
-
-}
-
-static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self)
-{
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon,
-  AudioUnitRenderActionFlags* UNUSED(ioActionFlags),
-  const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber),
-  UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData))
-{
-    ALCcoreAudioCapture *self = inRefCon;
-    AudioUnitRenderActionFlags flags = 0;
-    OSStatus err;
-
-    // fill the bufferList with data from the input device
-    err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList);
-    if(err != noErr)
-    {
-        ERR("AudioUnitRender error: %d\n", err);
-        return err;
-    }
-
-    ll_ringbuffer_write(self->ring, self->bufferList->mBuffers[0].mData, inNumberFrames);
-
-    return noErr;
-}
-
-static OSStatus ALCcoreAudioCapture_ConvertCallback(AudioConverterRef UNUSED(inAudioConverter),
-  UInt32 *ioNumberDataPackets, AudioBufferList *ioData,
-  AudioStreamPacketDescription** UNUSED(outDataPacketDescription),
-  void *inUserData)
-{
-    ALCcoreAudioCapture *self = inUserData;
-
-    // Read from the ring buffer and store temporarily in a large buffer
-    ll_ringbuffer_read(self->ring, self->resampleBuffer, *ioNumberDataPackets);
-
-    // Set the input data
-    ioData->mNumberBuffers = 1;
-    ioData->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
-    ioData->mBuffers[0].mData = self->resampleBuffer;
-    ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * self->format.mBytesPerFrame;
-
-    return noErr;
-}
-
-
-static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-    AudioStreamBasicDescription requestedFormat;  // The application requested format
-    AudioStreamBasicDescription hardwareFormat;   // The hardware format
-    AudioStreamBasicDescription outputFormat;     // The AudioUnit output format
-    AURenderCallbackStruct input;
-    AudioComponentDescription desc;
-    AudioDeviceID inputDevice;
-    UInt32 outputFrameCount;
-    UInt32 propertySize;
-    AudioObjectPropertyAddress propertyAddress;
-    UInt32 enableIO;
-    AudioComponent comp;
-    OSStatus err;
-
-    if(!name)
-        name = ca_device;
-    else if(strcmp(name, ca_device) != 0)
-        return ALC_INVALID_VALUE;
-
-    desc.componentType = kAudioUnitType_Output;
-    desc.componentSubType = kAudioUnitSubType_HALOutput;
-    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-    desc.componentFlags = 0;
-    desc.componentFlagsMask = 0;
-
-    // Search for component with given description
-    comp = AudioComponentFindNext(NULL, &desc);
-    if(comp == NULL)
-    {
-        ERR("AudioComponentFindNext failed\n");
-        return ALC_INVALID_VALUE;
-    }
-
-    // Open the component
-    err = AudioComponentInstanceNew(comp, &self->audioUnit);
-    if(err != noErr)
-    {
-        ERR("AudioComponentInstanceNew failed\n");
-        goto error;
-    }
-
-    // Turn off AudioUnit output
-    enableIO = 0;
-    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        goto error;
-    }
-
-    // Turn on AudioUnit input
-    enableIO = 1;
-    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        goto error;
-    }
-
-    // Get the default input device
-
-    propertySize = sizeof(AudioDeviceID);
-    propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
-    propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
-    propertyAddress.mElement = kAudioObjectPropertyElementMaster;
-
-    err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice);
-    if(err != noErr)
-    {
-        ERR("AudioObjectGetPropertyData failed\n");
-        goto error;
-    }
-
-    if(inputDevice == kAudioDeviceUnknown)
-    {
-        ERR("No input device found\n");
-        goto error;
-    }
-
-    // Track the input device
-    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        goto error;
-    }
-
-    // set capture callback
-    input.inputProc = ALCcoreAudioCapture_RecordProc;
-    input.inputProcRefCon = self;
-
-    err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        goto error;
-    }
-
-    // Initialize the device
-    err = AudioUnitInitialize(self->audioUnit);
-    if(err != noErr)
-    {
-        ERR("AudioUnitInitialize failed\n");
-        goto error;
-    }
-
-    // Get the hardware format
-    propertySize = sizeof(AudioStreamBasicDescription);
-    err = AudioUnitGetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
-    if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
-    {
-        ERR("AudioUnitGetProperty failed\n");
-        goto error;
-    }
-
-    // Set up the requested format description
-    switch(device->FmtType)
-    {
-        case DevFmtUByte:
-            requestedFormat.mBitsPerChannel = 8;
-            requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
-            break;
-        case DevFmtShort:
-            requestedFormat.mBitsPerChannel = 16;
-            requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
-            break;
-        case DevFmtInt:
-            requestedFormat.mBitsPerChannel = 32;
-            requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
-            break;
-        case DevFmtFloat:
-            requestedFormat.mBitsPerChannel = 32;
-            requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
-            break;
-        case DevFmtByte:
-        case DevFmtUShort:
-        case DevFmtUInt:
-            ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
-            goto error;
-    }
-
-    switch(device->FmtChans)
-    {
-        case DevFmtMono:
-            requestedFormat.mChannelsPerFrame = 1;
-            break;
-        case DevFmtStereo:
-            requestedFormat.mChannelsPerFrame = 2;
-            break;
-
-        case DevFmtQuad:
-        case DevFmtX51:
-        case DevFmtX51Rear:
-        case DevFmtX61:
-        case DevFmtX71:
-        case DevFmtAmbi3D:
-            ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
-            goto error;
-    }
-
-    requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
-    requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
-    requestedFormat.mSampleRate = device->Frequency;
-    requestedFormat.mFormatID = kAudioFormatLinearPCM;
-    requestedFormat.mReserved = 0;
-    requestedFormat.mFramesPerPacket = 1;
-
-    // save requested format description for later use
-    self->format = requestedFormat;
-    self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    // Use intermediate format for sample rate conversion (outputFormat)
-    // Set sample rate to the same as hardware for resampling later
-    outputFormat = requestedFormat;
-    outputFormat.mSampleRate = hardwareFormat.mSampleRate;
-
-    // Determine sample rate ratio for resampling
-    self->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
-
-    // The output format should be the requested format, but using the hardware sample rate
-    // This is because the AudioUnit will automatically scale other properties, except for sample rate
-    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed\n");
-        goto error;
-    }
-
-    // Set the AudioUnit output format frame count
-    outputFrameCount = device->UpdateSize * self->sampleRateRatio;
-    err = AudioUnitSetProperty(self->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
-    if(err != noErr)
-    {
-        ERR("AudioUnitSetProperty failed: %d\n", err);
-        goto error;
-    }
-
-    // Set up sample converter
-    err = AudioConverterNew(&outputFormat, &requestedFormat, &self->audioConverter);
-    if(err != noErr)
-    {
-        ERR("AudioConverterNew failed: %d\n", err);
-        goto error;
-    }
-
-    // Create a buffer for use in the resample callback
-    self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio);
-
-    // Allocate buffer for the AudioUnit output
-    self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio);
-    if(self->bufferList == NULL)
-        goto error;
-
-    self->ring = ll_ringbuffer_create(
-        device->UpdateSize*self->sampleRateRatio*device->NumUpdates + 1,
-        self->frameSize
-    );
-    if(!self->ring) goto error;
-
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-
-error:
-    ll_ringbuffer_free(self->ring);
-    self->ring = NULL;
-    free(self->resampleBuffer);
-    destroy_buffer_list(self->bufferList);
-
-    if(self->audioConverter)
-        AudioConverterDispose(self->audioConverter);
-    if(self->audioUnit)
-        AudioComponentInstanceDispose(self->audioUnit);
-
-    return ALC_INVALID_VALUE;
-}
-
-
-static void ALCcoreAudioCapture_close(ALCcoreAudioCapture *self)
-{
-    ll_ringbuffer_free(self->ring);
-    self->ring = NULL;
-
-    free(self->resampleBuffer);
-
-    destroy_buffer_list(self->bufferList);
-
-    AudioConverterDispose(self->audioConverter);
-    AudioComponentInstanceDispose(self->audioUnit);
-}
-
-static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
-{
-    OSStatus err = AudioOutputUnitStart(self->audioUnit);
-    if(err != noErr)
-    {
-        ERR("AudioOutputUnitStart failed\n");
-        return ALC_FALSE;
-    }
-    return ALC_TRUE;
-}
-
-static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
-{
-    OSStatus err = AudioOutputUnitStop(self->audioUnit);
-    if(err != noErr)
-        ERR("AudioOutputUnitStop failed\n");
-}
-
-static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
-{
-    AudioBufferList *list;
-    UInt32 frameCount;
-    OSStatus err;
-
-    // If no samples are requested, just return
-    if(samples == 0)
-        return ALC_NO_ERROR;
-
-    // Allocate a temporary AudioBufferList to use as the return resamples data
-    list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
-
-    // Point the resampling buffer to the capture buffer
-    list->mNumberBuffers = 1;
-    list->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
-    list->mBuffers[0].mDataByteSize = samples * self->frameSize;
-    list->mBuffers[0].mData = buffer;
-
-    // Resample into another AudioBufferList
-    frameCount = samples;
-    err = AudioConverterFillComplexBuffer(self->audioConverter,
-        ALCcoreAudioCapture_ConvertCallback, self, &frameCount, list, NULL
-    );
-    if(err != noErr)
-    {
-        ERR("AudioConverterFillComplexBuffer error: %d\n", err);
-        return ALC_INVALID_VALUE;
-    }
-    return ALC_NO_ERROR;
-}
-
-static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self)
-{
-    return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio;
-}
-
-
-typedef struct ALCcoreAudioBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCcoreAudioBackendFactory;
-#define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } }
-
-ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
-
-static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory *self);
-static DECLARE_FORWARD(ALCcoreAudioBackendFactory, ALCbackendFactory, void, deinit)
-static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory *self, ALCbackend_Type type);
-static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCcoreAudioBackendFactory);
-
-
-ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void)
-{
-    static ALCcoreAudioBackendFactory factory = ALCCOREAUDIOBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}
-
-
-static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory* UNUSED(self))
-{
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback || ALCbackend_Capture)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(ca_device);
-            break;
-        case CAPTURE_DEVICE_PROBE:
-            AppendCaptureDeviceList(ca_device);
-            break;
-    }
-}
-
-static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCcoreAudioPlayback *backend;
-        NEW_OBJ(backend, ALCcoreAudioPlayback)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-    if(type == ALCbackend_Capture)
-    {
-        ALCcoreAudioCapture *backend;
-        NEW_OBJ(backend, ALCcoreAudioCapture)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}

Alc/backends/jack.c

@@ -1,632 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "threads.h"
-#include "compat.h"
-
-#include "backends/base.h"
-
-#include <jack/jack.h>
-#include <jack/ringbuffer.h>
-
-
-static const ALCchar jackDevice[] = "JACK Default";
-
-
-#ifdef HAVE_DYNLOAD
-#define JACK_FUNCS(MAGIC)          \
-    MAGIC(jack_client_open);       \
-    MAGIC(jack_client_close);      \
-    MAGIC(jack_client_name_size);  \
-    MAGIC(jack_get_client_name);   \
-    MAGIC(jack_connect);           \
-    MAGIC(jack_activate);          \
-    MAGIC(jack_deactivate);        \
-    MAGIC(jack_port_register);     \
-    MAGIC(jack_port_unregister);   \
-    MAGIC(jack_port_get_buffer);   \
-    MAGIC(jack_port_name);         \
-    MAGIC(jack_get_ports);         \
-    MAGIC(jack_free);              \
-    MAGIC(jack_get_sample_rate);   \
-    MAGIC(jack_set_error_function); \
-    MAGIC(jack_set_process_callback); \
-    MAGIC(jack_set_buffer_size_callback); \
-    MAGIC(jack_set_buffer_size);   \
-    MAGIC(jack_get_buffer_size);
-
-static void *jack_handle;
-#define MAKE_FUNC(f) static __typeof(f) * p##f
-JACK_FUNCS(MAKE_FUNC);
-#undef MAKE_FUNC
-
-#define jack_client_open pjack_client_open
-#define jack_client_close pjack_client_close
-#define jack_client_name_size pjack_client_name_size
-#define jack_get_client_name pjack_get_client_name
-#define jack_connect pjack_connect
-#define jack_activate pjack_activate
-#define jack_deactivate pjack_deactivate
-#define jack_port_register pjack_port_register
-#define jack_port_unregister pjack_port_unregister
-#define jack_port_get_buffer pjack_port_get_buffer
-#define jack_port_name pjack_port_name
-#define jack_get_ports pjack_get_ports
-#define jack_free pjack_free
-#define jack_get_sample_rate pjack_get_sample_rate
-#define jack_set_error_function pjack_set_error_function
-#define jack_set_process_callback pjack_set_process_callback
-#define jack_set_buffer_size_callback pjack_set_buffer_size_callback
-#define jack_set_buffer_size pjack_set_buffer_size
-#define jack_get_buffer_size pjack_get_buffer_size
-#endif
-
-
-static jack_options_t ClientOptions = JackNullOption;
-
-static ALCboolean jack_load(void)
-{
-    ALCboolean error = ALC_FALSE;
-
-#ifdef HAVE_DYNLOAD
-    if(!jack_handle)
-    {
-        al_string missing_funcs = AL_STRING_INIT_STATIC();
-
-#ifdef _WIN32
-#define JACKLIB "libjack.dll"
-#else
-#define JACKLIB "libjack.so.0"
-#endif
-        jack_handle = LoadLib(JACKLIB);
-        if(!jack_handle)
-        {
-            WARN("Failed to load %s\n", JACKLIB);
-            return ALC_FALSE;
-        }
-
-        error = ALC_FALSE;
-#define LOAD_FUNC(f) do {                                                     \
-    p##f = GetSymbol(jack_handle, #f);                                        \
-    if(p##f == NULL) {                                                        \
-        error = ALC_TRUE;                                                     \
-        alstr_append_cstr(&missing_funcs, "\n" #f);                           \
-    }                                                                         \
-} while(0)
-        JACK_FUNCS(LOAD_FUNC);
-#undef LOAD_FUNC
-
-        if(error)
-        {
-            WARN("Missing expected functions:%s\n", alstr_get_cstr(missing_funcs));
-            CloseLib(jack_handle);
-            jack_handle = NULL;
-        }
-        alstr_reset(&missing_funcs);
-    }
-#endif
-
-    return !error;
-}
-
-
-typedef struct ALCjackPlayback {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    jack_client_t *Client;
-    jack_port_t *Port[MAX_OUTPUT_CHANNELS];
-
-    ll_ringbuffer_t *Ring;
-    alcnd_t Cond;
-
-    volatile int killNow;
-    althrd_t thread;
-} ALCjackPlayback;
-
-static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg);
-
-static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg);
-static int ALCjackPlayback_mixerProc(void *arg);
-
-static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device);
-static void ALCjackPlayback_Destruct(ALCjackPlayback *self);
-static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name);
-static void ALCjackPlayback_close(ALCjackPlayback *self);
-static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self);
-static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self);
-static void ALCjackPlayback_stop(ALCjackPlayback *self);
-static DECLARE_FORWARD2(ALCjackPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
-static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, ALCuint, availableSamples)
-static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self);
-static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCjackPlayback)
-
-DEFINE_ALCBACKEND_VTABLE(ALCjackPlayback);
-
-
-static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device)
-{
-    ALuint i;
-
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCjackPlayback, ALCbackend, self);
-
-    alcnd_init(&self->Cond);
-
-    self->Client = NULL;
-    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-        self->Port[i] = NULL;
-    self->Ring = NULL;
-
-    self->killNow = 1;
-}
-
-static void ALCjackPlayback_Destruct(ALCjackPlayback *self)
-{
-    ALuint i;
-
-    if(self->Client)
-    {
-        for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-        {
-            if(self->Port[i])
-                jack_port_unregister(self->Client, self->Port[i]);
-            self->Port[i] = NULL;
-        }
-        jack_client_close(self->Client);
-        self->Client = NULL;
-    }
-
-    alcnd_destroy(&self->Cond);
-
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg)
-{
-    ALCjackPlayback *self = arg;
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-    ALuint bufsize;
-
-    ALCjackPlayback_lock(self);
-    device->UpdateSize = numframes;
-    device->NumUpdates = 2;
-
-    bufsize = device->UpdateSize;
-    if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize))
-        bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize);
-    bufsize += device->UpdateSize;
-    device->NumUpdates = bufsize / device->UpdateSize;
-
-    TRACE("%u update size x%u\n", device->UpdateSize, device->NumUpdates);
-
-    ll_ringbuffer_free(self->Ring);
-    self->Ring = ll_ringbuffer_create(bufsize,
-        FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder)
-    );
-    if(!self->Ring)
-    {
-        ERR("Failed to reallocate ringbuffer\n");
-        aluHandleDisconnect(device);
-    }
-    ALCjackPlayback_unlock(self);
-    return 0;
-}
-
-
-static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg)
-{
-    ALCjackPlayback *self = arg;
-    jack_default_audio_sample_t *out[MAX_OUTPUT_CHANNELS];
-    ll_ringbuffer_data_t data[2];
-    jack_nframes_t total = 0;
-    jack_nframes_t todo;
-    ALsizei i, c, numchans;
-
-    ll_ringbuffer_get_read_vector(self->Ring, data);
-
-    for(c = 0;c < MAX_OUTPUT_CHANNELS && self->Port[c];c++)
-        out[c] = jack_port_get_buffer(self->Port[c], numframes);
-    numchans = c;
-
-    todo = minu(numframes, data[0].len);
-    for(c = 0;c < numchans;c++)
-    {
-        const ALfloat *restrict in = ((ALfloat*)data[0].buf) + c;
-        for(i = 0;(jack_nframes_t)i < todo;i++)
-            out[c][i] = in[i*numchans];
-        out[c] += todo;
-    }
-    total += todo;
-
-    todo = minu(numframes-total, data[1].len);
-    if(todo > 0)
-    {
-        for(c = 0;c < numchans;c++)
-        {
-            const ALfloat *restrict in = ((ALfloat*)data[1].buf) + c;
-            for(i = 0;(jack_nframes_t)i < todo;i++)
-                out[c][i] = in[i*numchans];
-            out[c] += todo;
-        }
-        total += todo;
-    }
-
-    ll_ringbuffer_read_advance(self->Ring, total);
-    alcnd_signal(&self->Cond);
-
-    if(numframes > total)
-    {
-        todo = numframes-total;
-        for(c = 0;c < numchans;c++)
-        {
-            for(i = 0;(jack_nframes_t)i < todo;i++)
-                out[c][i] = 0.0f;
-        }
-    }
-
-    return 0;
-}
-
-static int ALCjackPlayback_mixerProc(void *arg)
-{
-    ALCjackPlayback *self = arg;
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-    ll_ringbuffer_data_t data[2];
-
-    SetRTPriority();
-    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
-
-    ALCjackPlayback_lock(self);
-    while(!self->killNow && device->Connected)
-    {
-        ALuint todo, len1, len2;
-
-        /* NOTE: Unfortunately, there is an unavoidable race condition here.
-         * It's possible for the process() method to run, updating the read
-         * pointer and signaling the condition variable, in between the mixer
-         * loop checking the write size and waiting for the condition variable.
-         * This will cause the mixer loop to wait until the *next* process()
-         * invocation, most likely writing silence for it.
-         *
-         * However, this should only happen if the mixer is running behind
-         * anyway (as ideally we'll be asleep in alcnd_wait by the time the
-         * process() method is invoked), so this behavior is not unwarranted.
-         * It's unfortunate since it'll be wasting time sleeping that could be
-         * used to catch up, but there's no way around it without blocking in
-         * the process() method.
-         */
-        if(ll_ringbuffer_write_space(self->Ring) < device->UpdateSize)
-        {
-            alcnd_wait(&self->Cond, &STATIC_CAST(ALCbackend,self)->mMutex);
-            continue;
-        }
-
-        ll_ringbuffer_get_write_vector(self->Ring, data);
-        todo  = data[0].len + data[1].len;
-        todo -= todo%device->UpdateSize;
-
-        len1 = minu(data[0].len, todo);
-        len2 = minu(data[1].len, todo-len1);
-
-        aluMixData(device, data[0].buf, len1);
-        if(len2 > 0)
-            aluMixData(device, data[1].buf, len2);
-        ll_ringbuffer_write_advance(self->Ring, todo);
-    }
-    ALCjackPlayback_unlock(self);
-
-    return 0;
-}
-
-
-static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    const char *client_name = "alsoft";
-    jack_status_t status;
-
-    if(!name)
-        name = jackDevice;
-    else if(strcmp(name, jackDevice) != 0)
-        return ALC_INVALID_VALUE;
-
-    self->Client = jack_client_open(client_name, ClientOptions, &status, NULL);
-    if(self->Client == NULL)
-    {
-        ERR("jack_client_open() failed, status = 0x%02x\n", status);
-        return ALC_INVALID_VALUE;
-    }
-    if((status&JackServerStarted))
-        TRACE("JACK server started\n");
-    if((status&JackNameNotUnique))
-    {
-        client_name = jack_get_client_name(self->Client);
-        TRACE("Client name not unique, got `%s' instead\n", client_name);
-    }
-
-    jack_set_process_callback(self->Client, ALCjackPlayback_process, self);
-    jack_set_buffer_size_callback(self->Client, ALCjackPlayback_bufferSizeNotify, self);
-
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCjackPlayback_close(ALCjackPlayback *self)
-{
-    ALuint i;
-
-    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-    {
-        if(self->Port[i])
-            jack_port_unregister(self->Client, self->Port[i]);
-        self->Port[i] = NULL;
-    }
-    jack_client_close(self->Client);
-    self->Client = NULL;
-}
-
-static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    ALsizei numchans, i;
-    ALuint bufsize;
-
-    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-    {
-        if(self->Port[i])
-            jack_port_unregister(self->Client, self->Port[i]);
-        self->Port[i] = NULL;
-    }
-
-    /* Ignore the requested buffer metrics and just keep one JACK-sized buffer
-     * ready for when requested. Note that one period's worth of audio in the
-     * ring buffer will always be left unfilled because one element of the ring
-     * buffer will not be writeable, and we only write in period-sized chunks.
-     */
-    device->Frequency = jack_get_sample_rate(self->Client);
-    device->UpdateSize = jack_get_buffer_size(self->Client);
-    device->NumUpdates = 2;
-
-    bufsize = device->UpdateSize;
-    if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize))
-        bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize);
-    bufsize += device->UpdateSize;
-    device->NumUpdates = bufsize / device->UpdateSize;
-
-    /* Force 32-bit float output. */
-    device->FmtType = DevFmtFloat;
-
-    numchans = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
-    for(i = 0;i < numchans;i++)
-    {
-        char name[64];
-        snprintf(name, sizeof(name), "channel_%d", i+1);
-        self->Port[i] = jack_port_register(self->Client, name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
-        if(self->Port[i] == NULL)
-        {
-            ERR("Not enough JACK ports available for %s output\n", DevFmtChannelsString(device->FmtChans));
-            if(i == 0) return ALC_FALSE;
-            break;
-        }
-    }
-    if(i < numchans)
-    {
-        if(i == 1)
-            device->FmtChans = DevFmtMono;
-        else
-        {
-            for(--i;i >= 2;i--)
-            {
-                jack_port_unregister(self->Client, self->Port[i]);
-                self->Port[i] = NULL;
-            }
-            device->FmtChans = DevFmtStereo;
-        }
-    }
-
-    ll_ringbuffer_free(self->Ring);
-    self->Ring = ll_ringbuffer_create(bufsize,
-        FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder)
-    );
-    if(!self->Ring)
-    {
-        ERR("Failed to allocate ringbuffer\n");
-        return ALC_FALSE;
-    }
-
-    SetDefaultChannelOrder(device);
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self)
-{
-    const char **ports;
-    ALsizei i;
-
-    if(jack_activate(self->Client))
-    {
-        ERR("Failed to activate client\n");
-        return ALC_FALSE;
-    }
-
-    ports = jack_get_ports(self->Client, NULL, NULL, JackPortIsPhysical|JackPortIsInput);
-    if(ports == NULL)
-    {
-        ERR("No physical playback ports found\n");
-        jack_deactivate(self->Client);
-        return ALC_FALSE;
-    }
-    for(i = 0;i < MAX_OUTPUT_CHANNELS && self->Port[i];i++)
-    {
-        if(!ports[i])
-        {
-            ERR("No physical playback port for \"%s\"\n", jack_port_name(self->Port[i]));
-            break;
-        }
-        if(jack_connect(self->Client, jack_port_name(self->Port[i]), ports[i]))
-            ERR("Failed to connect output port \"%s\" to \"%s\"\n", jack_port_name(self->Port[i]), ports[i]);
-    }
-    jack_free(ports);
-
-    self->killNow = 0;
-    if(althrd_create(&self->thread, ALCjackPlayback_mixerProc, self) != althrd_success)
-    {
-        jack_deactivate(self->Client);
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void ALCjackPlayback_stop(ALCjackPlayback *self)
-{
-    int res;
-
-    if(self->killNow)
-        return;
-
-    self->killNow = 1;
-    /* Lock the backend to ensure we don't flag the mixer to die and signal the
-     * mixer to wake up in between it checking the flag and going to sleep and
-     * wait for a wakeup (potentially leading to it never waking back up to see
-     * the flag). */
-    ALCjackPlayback_lock(self);
-    ALCjackPlayback_unlock(self);
-    alcnd_signal(&self->Cond);
-    althrd_join(self->thread, &res);
-
-    jack_deactivate(self->Client);
-}
-
-
-static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    ClockLatency ret;
-
-    ALCjackPlayback_lock(self);
-    ret.ClockTime = GetDeviceClockTime(device);
-    ret.Latency = ll_ringbuffer_read_space(self->Ring) * DEVICE_CLOCK_RES /
-                  device->Frequency;
-    ALCjackPlayback_unlock(self);
-
-    return ret;
-}
-
-
-static void jack_msg_handler(const char *message)
-{
-    WARN("%s\n", message);
-}
-
-typedef struct ALCjackBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCjackBackendFactory;
-#define ALCJACKBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCjackBackendFactory, ALCbackendFactory) } }
-
-static ALCboolean ALCjackBackendFactory_init(ALCjackBackendFactory* UNUSED(self))
-{
-    jack_client_t *client;
-    jack_status_t status;
-
-    if(!jack_load())
-        return ALC_FALSE;
-
-    if(!GetConfigValueBool(NULL, "jack", "spawn-server", 0))
-        ClientOptions |= JackNoStartServer;
-
-    jack_set_error_function(jack_msg_handler);
-    client = jack_client_open("alsoft", ClientOptions, &status, NULL);
-    jack_set_error_function(NULL);
-    if(client == NULL)
-    {
-        WARN("jack_client_open() failed, 0x%02x\n", status);
-        if((status&JackServerFailed) && !(ClientOptions&JackNoStartServer))
-            ERR("Unable to connect to JACK server\n");
-        return ALC_FALSE;
-    }
-
-    jack_client_close(client);
-    return ALC_TRUE;
-}
-
-static void ALCjackBackendFactory_deinit(ALCjackBackendFactory* UNUSED(self))
-{
-#ifdef HAVE_DYNLOAD
-    if(jack_handle)
-        CloseLib(jack_handle);
-    jack_handle = NULL;
-#endif
-}
-
-static ALCboolean ALCjackBackendFactory_querySupport(ALCjackBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCjackBackendFactory_probe(ALCjackBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(jackDevice);
-            break;
-
-        case CAPTURE_DEVICE_PROBE:
-            break;
-    }
-}
-
-static ALCbackend* ALCjackBackendFactory_createBackend(ALCjackBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCjackPlayback *backend;
-        NEW_OBJ(backend, ALCjackPlayback)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}
-
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCjackBackendFactory);
-
-
-ALCbackendFactory *ALCjackBackendFactory_getFactory(void)
-{
-    static ALCjackBackendFactory factory = ALCJACKBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}

Alc/backends/loopback.c

@@ -1,133 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2011 by Chris Robinson
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alu.h"
-
-#include "backends/base.h"
-
-
-typedef struct ALCloopback {
-    DERIVE_FROM_TYPE(ALCbackend);
-} ALCloopback;
-
-static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device);
-static DECLARE_FORWARD(ALCloopback, ALCbackend, void, Destruct)
-static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name);
-static void ALCloopback_close(ALCloopback *self);
-static ALCboolean ALCloopback_reset(ALCloopback *self);
-static ALCboolean ALCloopback_start(ALCloopback *self);
-static void ALCloopback_stop(ALCloopback *self);
-static DECLARE_FORWARD2(ALCloopback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
-static DECLARE_FORWARD(ALCloopback, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCloopback, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCloopback, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCloopback, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCloopback)
-DEFINE_ALCBACKEND_VTABLE(ALCloopback);
-
-
-static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCloopback, ALCbackend, self);
-}
-
-
-static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-
-    alstr_copy_cstr(&device->DeviceName, name);
-    return ALC_NO_ERROR;
-}
-
-static void ALCloopback_close(ALCloopback* UNUSED(self))
-{
-}
-
-static ALCboolean ALCloopback_reset(ALCloopback *self)
-{
-    SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCloopback_start(ALCloopback* UNUSED(self))
-{
-    return ALC_TRUE;
-}
-
-static void ALCloopback_stop(ALCloopback* UNUSED(self))
-{
-}
-
-
-typedef struct ALCloopbackFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCloopbackFactory;
-#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCloopbackFactory, ALCbackendFactory) } }
-
-ALCbackendFactory *ALCloopbackFactory_getFactory(void);
-static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory *self);
-static DECLARE_FORWARD(ALCloopbackFactory, ALCbackendFactory, void, deinit)
-static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory *self, ALCbackend_Type type);
-static void ALCloopbackFactory_probe(ALCloopbackFactory *self, enum DevProbe type);
-static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory *self, ALCdevice *device, ALCbackend_Type type);
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCloopbackFactory);
-
-
-ALCbackendFactory *ALCloopbackFactory_getFactory(void)
-{
-    static ALCloopbackFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}
-
-static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory* UNUSED(self))
-{
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Loopback)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCloopbackFactory_probe(ALCloopbackFactory* UNUSED(self), enum DevProbe UNUSED(type))
-{
-}
-
-static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Loopback)
-    {
-        ALCloopback *backend;
-        NEW_OBJ(backend, ALCloopback)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}

Alc/backends/null.c

@@ -1,225 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2010 by Chris Robinson
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#ifdef HAVE_WINDOWS_H
-#include <windows.h>
-#endif
-
-#include "alMain.h"
-#include "alu.h"
-#include "threads.h"
-#include "compat.h"
-
-#include "backends/base.h"
-
-
-typedef struct ALCnullBackend {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    volatile int killNow;
-    althrd_t thread;
-} ALCnullBackend;
-
-static int ALCnullBackend_mixerProc(void *ptr);
-
-static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device);
-static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, Destruct)
-static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name);
-static void ALCnullBackend_close(ALCnullBackend *self);
-static ALCboolean ALCnullBackend_reset(ALCnullBackend *self);
-static ALCboolean ALCnullBackend_start(ALCnullBackend *self);
-static void ALCnullBackend_stop(ALCnullBackend *self);
-static DECLARE_FORWARD2(ALCnullBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
-static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCnullBackend)
-
-DEFINE_ALCBACKEND_VTABLE(ALCnullBackend);
-
-
-static const ALCchar nullDevice[] = "No Output";
-
-
-static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCnullBackend, ALCbackend, self);
-}
-
-
-static int ALCnullBackend_mixerProc(void *ptr)
-{
-    ALCnullBackend *self = (ALCnullBackend*)ptr;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    struct timespec now, start;
-    ALuint64 avail, done;
-    const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 /
-                                 device->Frequency / 2);
-
-    SetRTPriority();
-    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
-
-    done = 0;
-    if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
-    {
-        ERR("Failed to get starting time\n");
-        return 1;
-    }
-    while(!self->killNow && device->Connected)
-    {
-        if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC)
-        {
-            ERR("Failed to get current time\n");
-            return 1;
-        }
-
-        avail  = (now.tv_sec - start.tv_sec) * device->Frequency;
-        avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
-        if(avail < done)
-        {
-            /* Oops, time skipped backwards. Reset the number of samples done
-             * with one update available since we (likely) just came back from
-             * sleeping. */
-            done = avail - device->UpdateSize;
-        }
-
-        if(avail-done < device->UpdateSize)
-            al_nssleep(restTime);
-        else while(avail-done >= device->UpdateSize)
-        {
-            ALCnullBackend_lock(self);
-            aluMixData(device, NULL, device->UpdateSize);
-            ALCnullBackend_unlock(self);
-            done += device->UpdateSize;
-        }
-    }
-
-    return 0;
-}
-
-
-static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name)
-{
-    ALCdevice *device;
-
-    if(!name)
-        name = nullDevice;
-    else if(strcmp(name, nullDevice) != 0)
-        return ALC_INVALID_VALUE;
-
-    device = STATIC_CAST(ALCbackend, self)->mDevice;
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCnullBackend_close(ALCnullBackend* UNUSED(self))
-{
-}
-
-static ALCboolean ALCnullBackend_reset(ALCnullBackend *self)
-{
-    SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCnullBackend_start(ALCnullBackend *self)
-{
-    self->killNow = 0;
-    if(althrd_create(&self->thread, ALCnullBackend_mixerProc, self) != althrd_success)
-        return ALC_FALSE;
-    return ALC_TRUE;
-}
-
-static void ALCnullBackend_stop(ALCnullBackend *self)
-{
-    int res;
-
-    if(self->killNow)
-        return;
-
-    self->killNow = 1;
-    althrd_join(self->thread, &res);
-}
-
-
-typedef struct ALCnullBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCnullBackendFactory;
-#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCnullBackendFactory, ALCbackendFactory) } }
-
-ALCbackendFactory *ALCnullBackendFactory_getFactory(void);
-
-static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory *self);
-static DECLARE_FORWARD(ALCnullBackendFactory, ALCbackendFactory, void, deinit)
-static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory *self, ALCbackend_Type type);
-static void ALCnullBackendFactory_probe(ALCnullBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCnullBackendFactory);
-
-
-ALCbackendFactory *ALCnullBackendFactory_getFactory(void)
-{
-    static ALCnullBackendFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}
-
-
-static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory* UNUSED(self))
-{
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCnullBackendFactory_probe(ALCnullBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(nullDevice);
-            break;
-        case CAPTURE_DEVICE_PROBE:
-            break;
-    }
-}
-
-static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCnullBackend *backend;
-        NEW_OBJ(backend, ALCnullBackend)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}

Alc/backends/oss.c

@@ -1,868 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <sys/ioctl.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <memory.h>
-#include <unistd.h>
-#include <errno.h>
-#include <math.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "threads.h"
-#include "compat.h"
-
-#include "backends/base.h"
-
-#include <sys/soundcard.h>
-
-/*
- * The OSS documentation talks about SOUND_MIXER_READ, but the header
- * only contains MIXER_READ. Play safe. Same for WRITE.
- */
-#ifndef SOUND_MIXER_READ
-#define SOUND_MIXER_READ MIXER_READ
-#endif
-#ifndef SOUND_MIXER_WRITE
-#define SOUND_MIXER_WRITE MIXER_WRITE
-#endif
-
-#if defined(SOUND_VERSION) && (SOUND_VERSION < 0x040000)
-#define ALC_OSS_COMPAT
-#endif
-#ifndef SNDCTL_AUDIOINFO
-#define ALC_OSS_COMPAT
-#endif
-
-/*
- * FreeBSD strongly discourages the use of specific devices,
- * such as those returned in oss_audioinfo.devnode
- */
-#ifdef __FreeBSD__
-#define ALC_OSS_DEVNODE_TRUC
-#endif
-
-struct oss_device {
-    const ALCchar *handle;
-    const char *path;
-    struct oss_device *next;
-};
-
-static struct oss_device oss_playback = {
-    "OSS Default",
-    "/dev/dsp",
-    NULL
-};
-
-static struct oss_device oss_capture = {
-    "OSS Default",
-    "/dev/dsp",
-    NULL
-};
-
-#ifdef ALC_OSS_COMPAT
-
-#define DSP_CAP_OUTPUT 0x00020000
-#define DSP_CAP_INPUT 0x00010000
-static void ALCossListPopulate(struct oss_device *UNUSED(devlist), int UNUSED(type_flag))
-{
-}
-
-#else
-
-#ifndef HAVE_STRNLEN
-static size_t strnlen(const char *str, size_t maxlen)
-{
-    const char *end = memchr(str, 0, maxlen);
-    if(!end) return maxlen;
-    return end - str;
-}
-#endif
-
-static void ALCossListAppend(struct oss_device *list, const char *handle, size_t hlen, const char *path, size_t plen)
-{
-    struct oss_device *next;
-    struct oss_device *last;
-    size_t i;
-
-    /* skip the first item "OSS Default" */
-    last = list;
-    next = list->next;
-#ifdef ALC_OSS_DEVNODE_TRUC
-    for(i = 0;i < plen;i++)
-    {
-        if(path[i] == '.')
-        {
-            if(strncmp(path + i, handle + hlen + i - plen, plen - i) == 0)
-                hlen = hlen + i - plen;
-            plen = i;
-        }
-    }
-#else
-    (void)i;
-#endif
-    if(handle[0] == '\0')
-    {
-        handle = path;
-        hlen = plen;
-    }
-
-    while(next != NULL)
-    {
-        if(strncmp(next->path, path, plen) == 0)
-            return;
-        last = next;
-        next = next->next;
-    }
-
-    next = (struct oss_device*)malloc(sizeof(struct oss_device) + hlen + plen + 2);
-    next->handle = (char*)(next + 1);
-    next->path = next->handle + hlen + 1;
-    next->next = NULL;
-    last->next = next;
-
-    strncpy((char*)next->handle, handle, hlen);
-    ((char*)next->handle)[hlen] = '\0';
-    strncpy((char*)next->path, path, plen);
-    ((char*)next->path)[plen] = '\0';
-
-    TRACE("Got device \"%s\", \"%s\"\n", next->handle, next->path);
-}
-
-static void ALCossListPopulate(struct oss_device *devlist, int type_flag)
-{
-    struct oss_sysinfo si;
-    struct oss_audioinfo ai;
-    int fd, i;
-
-    if((fd=open("/dev/mixer", O_RDONLY)) < 0)
-    {
-        TRACE("Could not open /dev/mixer: %s\n", strerror(errno));
-        return;
-    }
-    if(ioctl(fd, SNDCTL_SYSINFO, &si) == -1)
-    {
-        TRACE("SNDCTL_SYSINFO failed: %s\n", strerror(errno));
-        goto done;
-    }
-    for(i = 0;i < si.numaudios;i++)
-    {
-        const char *handle;
-        size_t len;
-
-        ai.dev = i;
-        if(ioctl(fd, SNDCTL_AUDIOINFO, &ai) == -1)
-        {
-            ERR("SNDCTL_AUDIOINFO (%d) failed: %s\n", i, strerror(errno));
-            continue;
-        }
-        if(ai.devnode[0] == '\0')
-            continue;
-
-        if(ai.handle[0] != '\0')
-        {
-            len = strnlen(ai.handle, sizeof(ai.handle));
-            handle = ai.handle;
-        }
-        else
-        {
-            len = strnlen(ai.name, sizeof(ai.name));
-            handle = ai.name;
-        }
-        if((ai.caps&type_flag))
-            ALCossListAppend(devlist, handle, len, ai.devnode,
-                             strnlen(ai.devnode, sizeof(ai.devnode)));
-    }
-
-done:
-    close(fd);
-}
-
-#endif
-
-static void ALCossListFree(struct oss_device *list)
-{
-    struct oss_device *cur;
-    if(list == NULL)
-        return;
-
-    /* skip the first item "OSS Default" */
-    cur = list->next;
-    list->next = NULL;
-
-    while(cur != NULL)
-    {
-        struct oss_device *next = cur->next;
-        free(cur);
-        cur = next;
-    }
-}
-
-static int log2i(ALCuint x)
-{
-    int y = 0;
-    while (x > 1)
-    {
-        x >>= 1;
-        y++;
-    }
-    return y;
-}
-
-typedef struct ALCplaybackOSS {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    int fd;
-
-    ALubyte *mix_data;
-    int data_size;
-
-    ATOMIC(ALenum) killNow;
-    althrd_t thread;
-} ALCplaybackOSS;
-
-static int ALCplaybackOSS_mixerProc(void *ptr);
-
-static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device);
-static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, Destruct)
-static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name);
-static void ALCplaybackOSS_close(ALCplaybackOSS *self);
-static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self);
-static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self);
-static void ALCplaybackOSS_stop(ALCplaybackOSS *self);
-static DECLARE_FORWARD2(ALCplaybackOSS, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
-static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCplaybackOSS)
-DEFINE_ALCBACKEND_VTABLE(ALCplaybackOSS);
-
-
-static int ALCplaybackOSS_mixerProc(void *ptr)
-{
-    ALCplaybackOSS *self = (ALCplaybackOSS*)ptr;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    struct timeval timeout;
-    ALubyte *write_ptr;
-    ALint frame_size;
-    ALint to_write;
-    ssize_t wrote;
-    fd_set wfds;
-    int sret;
-
-    SetRTPriority();
-    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
-
-    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    ALCplaybackOSS_lock(self);
-    while(!ATOMIC_LOAD_SEQ(&self->killNow) && device->Connected)
-    {
-        FD_ZERO(&wfds);
-        FD_SET(self->fd, &wfds);
-        timeout.tv_sec = 1;
-        timeout.tv_usec = 0;
-
-        ALCplaybackOSS_unlock(self);
-        sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
-        ALCplaybackOSS_lock(self);
-        if(sret < 0)
-        {
-            if(errno == EINTR)
-                continue;
-            ERR("select failed: %s\n", strerror(errno));
-            aluHandleDisconnect(device);
-            break;
-        }
-        else if(sret == 0)
-        {
-            WARN("select timeout\n");
-            continue;
-        }
-
-        write_ptr = self->mix_data;
-        to_write = self->data_size;
-        aluMixData(device, write_ptr, to_write/frame_size);
-        while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow))
-        {
-            wrote = write(self->fd, write_ptr, to_write);
-            if(wrote < 0)
-            {
-                if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
-                    continue;
-                ERR("write failed: %s\n", strerror(errno));
-                aluHandleDisconnect(device);
-                break;
-            }
-
-            to_write -= wrote;
-            write_ptr += wrote;
-        }
-    }
-    ALCplaybackOSS_unlock(self);
-
-    return 0;
-}
-
-
-static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCplaybackOSS, ALCbackend, self);
-
-    ATOMIC_INIT(&self->killNow, AL_FALSE);
-}
-
-static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name)
-{
-    struct oss_device *dev = &oss_playback;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-
-    if(!name || strcmp(name, dev->handle) == 0)
-        name = dev->handle;
-    else
-    {
-        if(!dev->next)
-        {
-            ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT);
-            dev = &oss_playback;
-        }
-        while(dev != NULL)
-        {
-            if (strcmp(dev->handle, name) == 0)
-                break;
-            dev = dev->next;
-        }
-        if(dev == NULL)
-        {
-            WARN("Could not find \"%s\" in device list\n", name);
-            return ALC_INVALID_VALUE;
-        }
-    }
-
-    self->fd = open(dev->path, O_WRONLY);
-    if(self->fd == -1)
-    {
-        ERR("Could not open %s: %s\n", dev->path, strerror(errno));
-        return ALC_INVALID_VALUE;
-    }
-
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCplaybackOSS_close(ALCplaybackOSS *self)
-{
-    close(self->fd);
-    self->fd = -1;
-}
-
-static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    int numFragmentsLogSize;
-    int log2FragmentSize;
-    unsigned int periods;
-    audio_buf_info info;
-    ALuint frameSize;
-    int numChannels;
-    int ossFormat;
-    int ossSpeed;
-    char *err;
-
-    switch(device->FmtType)
-    {
-        case DevFmtByte:
-            ossFormat = AFMT_S8;
-            break;
-        case DevFmtUByte:
-            ossFormat = AFMT_U8;
-            break;
-        case DevFmtUShort:
-        case DevFmtInt:
-        case DevFmtUInt:
-        case DevFmtFloat:
-            device->FmtType = DevFmtShort;
-            /* fall-through */
-        case DevFmtShort:
-            ossFormat = AFMT_S16_NE;
-            break;
-    }
-
-    periods = device->NumUpdates;
-    numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
-    ossSpeed = device->Frequency;
-    frameSize = numChannels * BytesFromDevFmt(device->FmtType);
-    /* According to the OSS spec, 16 bytes (log2(16)) is the minimum. */
-    log2FragmentSize = maxi(log2i(device->UpdateSize*frameSize), 4);
-    numFragmentsLogSize = (periods << 16) | log2FragmentSize;
-
-#define CHECKERR(func) if((func) < 0) {                                       \
-    err = #func;                                                              \
-    goto err;                                                                 \
-}
-    /* Don't fail if SETFRAGMENT fails. We can handle just about anything
-     * that's reported back via GETOSPACE */
-    ioctl(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize);
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat));
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels));
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed));
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_GETOSPACE, &info));
-    if(0)
-    {
-    err:
-        ERR("%s failed: %s\n", err, strerror(errno));
-        return ALC_FALSE;
-    }
-#undef CHECKERR
-
-    if((int)ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != numChannels)
-    {
-        ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels);
-        return ALC_FALSE;
-    }
-
-    if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) ||
-         (ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) ||
-         (ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort)))
-    {
-        ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat);
-        return ALC_FALSE;
-    }
-
-    device->Frequency = ossSpeed;
-    device->UpdateSize = info.fragsize / frameSize;
-    device->NumUpdates = info.fragments;
-
-    SetDefaultChannelOrder(device);
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-
-    self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
-        device->FmtChans, device->FmtType, device->AmbiOrder
-    );
-    self->mix_data = calloc(1, self->data_size);
-
-    ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
-    if(althrd_create(&self->thread, ALCplaybackOSS_mixerProc, self) != althrd_success)
-    {
-        free(self->mix_data);
-        self->mix_data = NULL;
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void ALCplaybackOSS_stop(ALCplaybackOSS *self)
-{
-    int res;
-
-    if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
-        return;
-    althrd_join(self->thread, &res);
-
-    if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0)
-        ERR("Error resetting device: %s\n", strerror(errno));
-
-    free(self->mix_data);
-    self->mix_data = NULL;
-}
-
-
-typedef struct ALCcaptureOSS {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    int fd;
-
-    ll_ringbuffer_t *ring;
-
-    ATOMIC(ALenum) killNow;
-    althrd_t thread;
-} ALCcaptureOSS;
-
-static int ALCcaptureOSS_recordProc(void *ptr);
-
-static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device);
-static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, Destruct)
-static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name);
-static void ALCcaptureOSS_close(ALCcaptureOSS *self);
-static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ALCboolean, reset)
-static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self);
-static void ALCcaptureOSS_stop(ALCcaptureOSS *self);
-static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples);
-static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self);
-static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCcaptureOSS)
-DEFINE_ALCBACKEND_VTABLE(ALCcaptureOSS);
-
-
-static int ALCcaptureOSS_recordProc(void *ptr)
-{
-    ALCcaptureOSS *self = (ALCcaptureOSS*)ptr;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    struct timeval timeout;
-    int frame_size;
-    fd_set rfds;
-    ssize_t amt;
-    int sret;
-
-    SetRTPriority();
-    althrd_setname(althrd_current(), RECORD_THREAD_NAME);
-
-    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    while(!ATOMIC_LOAD_SEQ(&self->killNow))
-    {
-        ll_ringbuffer_data_t vec[2];
-
-        FD_ZERO(&rfds);
-        FD_SET(self->fd, &rfds);
-        timeout.tv_sec = 1;
-        timeout.tv_usec = 0;
-
-        sret = select(self->fd+1, &rfds, NULL, NULL, &timeout);
-        if(sret < 0)
-        {
-            if(errno == EINTR)
-                continue;
-            ERR("select failed: %s\n", strerror(errno));
-            aluHandleDisconnect(device);
-            break;
-        }
-        else if(sret == 0)
-        {
-            WARN("select timeout\n");
-            continue;
-        }
-
-        ll_ringbuffer_get_write_vector(self->ring, vec);
-        if(vec[0].len > 0)
-        {
-            amt = read(self->fd, vec[0].buf, vec[0].len*frame_size);
-            if(amt < 0)
-            {
-                ERR("read failed: %s\n", strerror(errno));
-                ALCcaptureOSS_lock(self);
-                aluHandleDisconnect(device);
-                ALCcaptureOSS_unlock(self);
-                break;
-            }
-            ll_ringbuffer_write_advance(self->ring, amt/frame_size);
-        }
-    }
-
-    return 0;
-}
-
-
-static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCcaptureOSS, ALCbackend, self);
-
-    ATOMIC_INIT(&self->killNow, AL_FALSE);
-}
-
-static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    struct oss_device *dev = &oss_capture;
-    int numFragmentsLogSize;
-    int log2FragmentSize;
-    unsigned int periods;
-    audio_buf_info info;
-    ALuint frameSize;
-    int numChannels;
-    int ossFormat;
-    int ossSpeed;
-    char *err;
-
-    if(!name || strcmp(name, dev->handle) == 0)
-        name = dev->handle;
-    else
-    {
-        if(!dev->next)
-        {
-            ALCossListPopulate(&oss_capture, DSP_CAP_INPUT);
-            dev = &oss_capture;
-        }
-        while(dev != NULL)
-        {
-            if (strcmp(dev->handle, name) == 0)
-                break;
-            dev = dev->next;
-        }
-        if(dev == NULL)
-        {
-            WARN("Could not find \"%s\" in device list\n", name);
-            return ALC_INVALID_VALUE;
-        }
-    }
-
-    self->fd = open(dev->path, O_RDONLY);
-    if(self->fd == -1)
-    {
-        ERR("Could not open %s: %s\n", dev->path, strerror(errno));
-        return ALC_INVALID_VALUE;
-    }
-
-    switch(device->FmtType)
-    {
-        case DevFmtByte:
-            ossFormat = AFMT_S8;
-            break;
-        case DevFmtUByte:
-            ossFormat = AFMT_U8;
-            break;
-        case DevFmtShort:
-            ossFormat = AFMT_S16_NE;
-            break;
-        case DevFmtUShort:
-        case DevFmtInt:
-        case DevFmtUInt:
-        case DevFmtFloat:
-            ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
-            return ALC_INVALID_VALUE;
-    }
-
-    periods = 4;
-    numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
-    frameSize = numChannels * BytesFromDevFmt(device->FmtType);
-    ossSpeed = device->Frequency;
-    log2FragmentSize = log2i(device->UpdateSize * device->NumUpdates *
-                             frameSize / periods);
-
-    /* according to the OSS spec, 16 bytes are the minimum */
-    if (log2FragmentSize < 4)
-        log2FragmentSize = 4;
-    numFragmentsLogSize = (periods << 16) | log2FragmentSize;
-
-#define CHECKERR(func) if((func) < 0) {                                       \
-    err = #func;                                                              \
-    goto err;                                                                 \
-}
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize));
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat));
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels));
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed));
-    CHECKERR(ioctl(self->fd, SNDCTL_DSP_GETISPACE, &info));
-    if(0)
-    {
-    err:
-        ERR("%s failed: %s\n", err, strerror(errno));
-        close(self->fd);
-        self->fd = -1;
-        return ALC_INVALID_VALUE;
-    }
-#undef CHECKERR
-
-    if((int)ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != numChannels)
-    {
-        ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels);
-        close(self->fd);
-        self->fd = -1;
-        return ALC_INVALID_VALUE;
-    }
-
-    if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) ||
-         (ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) ||
-         (ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort)))
-    {
-        ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat);
-        close(self->fd);
-        self->fd = -1;
-        return ALC_INVALID_VALUE;
-    }
-
-    self->ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates + 1, frameSize);
-    if(!self->ring)
-    {
-        ERR("Ring buffer create failed\n");
-        close(self->fd);
-        self->fd = -1;
-        return ALC_OUT_OF_MEMORY;
-    }
-
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCcaptureOSS_close(ALCcaptureOSS *self)
-{
-    close(self->fd);
-    self->fd = -1;
-
-    ll_ringbuffer_free(self->ring);
-    self->ring = NULL;
-}
-
-static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self)
-{
-    ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
-    if(althrd_create(&self->thread, ALCcaptureOSS_recordProc, self) != althrd_success)
-        return ALC_FALSE;
-    return ALC_TRUE;
-}
-
-static void ALCcaptureOSS_stop(ALCcaptureOSS *self)
-{
-    int res;
-
-    if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
-        return;
-
-    althrd_join(self->thread, &res);
-
-    if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0)
-        ERR("Error resetting device: %s\n", strerror(errno));
-}
-
-static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples)
-{
-    ll_ringbuffer_read(self->ring, buffer, samples);
-    return ALC_NO_ERROR;
-}
-
-static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self)
-{
-    return ll_ringbuffer_read_space(self->ring);
-}
-
-
-typedef struct ALCossBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCossBackendFactory;
-#define ALCOSSBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCossBackendFactory, ALCbackendFactory) } }
-
-ALCbackendFactory *ALCossBackendFactory_getFactory(void);
-
-static ALCboolean ALCossBackendFactory_init(ALCossBackendFactory *self);
-static void ALCossBackendFactory_deinit(ALCossBackendFactory *self);
-static ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory *self, ALCbackend_Type type);
-static void ALCossBackendFactory_probe(ALCossBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCossBackendFactory);
-
-
-ALCbackendFactory *ALCossBackendFactory_getFactory(void)
-{
-    static ALCossBackendFactory factory = ALCOSSBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}
-
-
-ALCboolean ALCossBackendFactory_init(ALCossBackendFactory* UNUSED(self))
-{
-    ConfigValueStr(NULL, "oss", "device", &oss_playback.path);
-    ConfigValueStr(NULL, "oss", "capture", &oss_capture.path);
-
-    return ALC_TRUE;
-}
-
-void  ALCossBackendFactory_deinit(ALCossBackendFactory* UNUSED(self))
-{
-    ALCossListFree(&oss_playback);
-    ALCossListFree(&oss_capture);
-}
-
-
-ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback || type == ALCbackend_Capture)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-void ALCossBackendFactory_probe(ALCossBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    struct oss_device *cur;
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            ALCossListFree(&oss_playback);
-            ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT);
-            cur = &oss_playback;
-            while(cur != NULL)
-            {
-#ifdef HAVE_STAT
-                struct stat buf;
-                if(stat(cur->path, &buf) == 0)
-#endif
-                    AppendAllDevicesList(cur->handle);
-                cur = cur->next;
-            }
-            break;
-
-        case CAPTURE_DEVICE_PROBE:
-            ALCossListFree(&oss_capture);
-            ALCossListPopulate(&oss_capture, DSP_CAP_INPUT);
-            cur = &oss_capture;
-            while(cur != NULL)
-            {
-#ifdef HAVE_STAT
-                struct stat buf;
-                if(stat(cur->path, &buf) == 0)
-#endif
-                    AppendCaptureDeviceList(cur->handle);
-                cur = cur->next;
-            }
-            break;
-    }
-}
-
-ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCplaybackOSS *backend;
-        NEW_OBJ(backend, ALCplaybackOSS)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-    if(type == ALCbackend_Capture)
-    {
-        ALCcaptureOSS *backend;
-        NEW_OBJ(backend, ALCcaptureOSS)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}

Alc/backends/portaudio.c

@@ -1,575 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "compat.h"
-
-#include "backends/base.h"
-
-#include <portaudio.h>
-
-
-static const ALCchar pa_device[] = "PortAudio Default";
-
-
-#ifdef HAVE_DYNLOAD
-static void *pa_handle;
-#define MAKE_FUNC(x) static __typeof(x) * p##x
-MAKE_FUNC(Pa_Initialize);
-MAKE_FUNC(Pa_Terminate);
-MAKE_FUNC(Pa_GetErrorText);
-MAKE_FUNC(Pa_StartStream);
-MAKE_FUNC(Pa_StopStream);
-MAKE_FUNC(Pa_OpenStream);
-MAKE_FUNC(Pa_CloseStream);
-MAKE_FUNC(Pa_GetDefaultOutputDevice);
-MAKE_FUNC(Pa_GetDefaultInputDevice);
-MAKE_FUNC(Pa_GetStreamInfo);
-#undef MAKE_FUNC
-
-#define Pa_Initialize                  pPa_Initialize
-#define Pa_Terminate                   pPa_Terminate
-#define Pa_GetErrorText                pPa_GetErrorText
-#define Pa_StartStream                 pPa_StartStream
-#define Pa_StopStream                  pPa_StopStream
-#define Pa_OpenStream                  pPa_OpenStream
-#define Pa_CloseStream                 pPa_CloseStream
-#define Pa_GetDefaultOutputDevice      pPa_GetDefaultOutputDevice
-#define Pa_GetDefaultInputDevice       pPa_GetDefaultInputDevice
-#define Pa_GetStreamInfo               pPa_GetStreamInfo
-#endif
-
-static ALCboolean pa_load(void)
-{
-    PaError err;
-
-#ifdef HAVE_DYNLOAD
-    if(!pa_handle)
-    {
-#ifdef _WIN32
-# define PALIB "portaudio.dll"
-#elif defined(__APPLE__) && defined(__MACH__)
-# define PALIB "libportaudio.2.dylib"
-#elif defined(__OpenBSD__)
-# define PALIB "libportaudio.so"
-#else
-# define PALIB "libportaudio.so.2"
-#endif
-
-        pa_handle = LoadLib(PALIB);
-        if(!pa_handle)
-            return ALC_FALSE;
-
-#define LOAD_FUNC(f) do {                                                     \
-    p##f = GetSymbol(pa_handle, #f);                                          \
-    if(p##f == NULL)                                                          \
-    {                                                                         \
-        CloseLib(pa_handle);                                                  \
-        pa_handle = NULL;                                                     \
-        return ALC_FALSE;                                                     \
-    }                                                                         \
-} while(0)
-        LOAD_FUNC(Pa_Initialize);
-        LOAD_FUNC(Pa_Terminate);
-        LOAD_FUNC(Pa_GetErrorText);
-        LOAD_FUNC(Pa_StartStream);
-        LOAD_FUNC(Pa_StopStream);
-        LOAD_FUNC(Pa_OpenStream);
-        LOAD_FUNC(Pa_CloseStream);
-        LOAD_FUNC(Pa_GetDefaultOutputDevice);
-        LOAD_FUNC(Pa_GetDefaultInputDevice);
-        LOAD_FUNC(Pa_GetStreamInfo);
-#undef LOAD_FUNC
-
-        if((err=Pa_Initialize()) != paNoError)
-        {
-            ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err));
-            CloseLib(pa_handle);
-            pa_handle = NULL;
-            return ALC_FALSE;
-        }
-    }
-#else
-    if((err=Pa_Initialize()) != paNoError)
-    {
-        ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err));
-        return ALC_FALSE;
-    }
-#endif
-    return ALC_TRUE;
-}
-
-
-typedef struct ALCportPlayback {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    PaStream *stream;
-    PaStreamParameters params;
-    ALuint update_size;
-} ALCportPlayback;
-
-static int ALCportPlayback_WriteCallback(const void *inputBuffer, void *outputBuffer,
-    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
-    const PaStreamCallbackFlags statusFlags, void *userData);
-
-static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device);
-static void ALCportPlayback_Destruct(ALCportPlayback *self);
-static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name);
-static void ALCportPlayback_close(ALCportPlayback *self);
-static ALCboolean ALCportPlayback_reset(ALCportPlayback *self);
-static ALCboolean ALCportPlayback_start(ALCportPlayback *self);
-static void ALCportPlayback_stop(ALCportPlayback *self);
-static DECLARE_FORWARD2(ALCportPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
-static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCportPlayback)
-
-DEFINE_ALCBACKEND_VTABLE(ALCportPlayback);
-
-
-static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCportPlayback, ALCbackend, self);
-
-    self->stream = NULL;
-}
-
-static void ALCportPlayback_Destruct(ALCportPlayback *self)
-{
-    if(self->stream)
-        Pa_CloseStream(self->stream);
-    self->stream = NULL;
-
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-static int ALCportPlayback_WriteCallback(const void *UNUSED(inputBuffer), void *outputBuffer,
-    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo),
-    const PaStreamCallbackFlags UNUSED(statusFlags), void *userData)
-{
-    ALCportPlayback *self = userData;
-
-    ALCportPlayback_lock(self);
-    aluMixData(STATIC_CAST(ALCbackend, self)->mDevice, outputBuffer, framesPerBuffer);
-    ALCportPlayback_unlock(self);
-    return 0;
-}
-
-
-static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    PaError err;
-
-    if(!name)
-        name = pa_device;
-    else if(strcmp(name, pa_device) != 0)
-        return ALC_INVALID_VALUE;
-
-    self->update_size = device->UpdateSize;
-
-    self->params.device = -1;
-    if(!ConfigValueInt(NULL, "port", "device", &self->params.device) ||
-       self->params.device < 0)
-        self->params.device = Pa_GetDefaultOutputDevice();
-    self->params.suggestedLatency = (device->UpdateSize*device->NumUpdates) /
-                                    (float)device->Frequency;
-    self->params.hostApiSpecificStreamInfo = NULL;
-
-    self->params.channelCount = ((device->FmtChans == DevFmtMono) ? 1 : 2);
-
-    switch(device->FmtType)
-    {
-        case DevFmtByte:
-            self->params.sampleFormat = paInt8;
-            break;
-        case DevFmtUByte:
-            self->params.sampleFormat = paUInt8;
-            break;
-        case DevFmtUShort:
-            /* fall-through */
-        case DevFmtShort:
-            self->params.sampleFormat = paInt16;
-            break;
-        case DevFmtUInt:
-            /* fall-through */
-        case DevFmtInt:
-            self->params.sampleFormat = paInt32;
-            break;
-        case DevFmtFloat:
-            self->params.sampleFormat = paFloat32;
-            break;
-    }
-
-retry_open:
-    err = Pa_OpenStream(&self->stream, NULL, &self->params,
-        device->Frequency, device->UpdateSize, paNoFlag,
-        ALCportPlayback_WriteCallback, self
-    );
-    if(err != paNoError)
-    {
-        if(self->params.sampleFormat == paFloat32)
-        {
-            self->params.sampleFormat = paInt16;
-            goto retry_open;
-        }
-        ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
-        return ALC_INVALID_VALUE;
-    }
-
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-
-}
-
-static void ALCportPlayback_close(ALCportPlayback *self)
-{
-    PaError err = Pa_CloseStream(self->stream);
-    if(err != paNoError)
-        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
-    self->stream = NULL;
-}
-
-static ALCboolean ALCportPlayback_reset(ALCportPlayback *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    const PaStreamInfo *streamInfo;
-
-    streamInfo = Pa_GetStreamInfo(self->stream);
-    device->Frequency = streamInfo->sampleRate;
-    device->UpdateSize = self->update_size;
-
-    if(self->params.sampleFormat == paInt8)
-        device->FmtType = DevFmtByte;
-    else if(self->params.sampleFormat == paUInt8)
-        device->FmtType = DevFmtUByte;
-    else if(self->params.sampleFormat == paInt16)
-        device->FmtType = DevFmtShort;
-    else if(self->params.sampleFormat == paInt32)
-        device->FmtType = DevFmtInt;
-    else if(self->params.sampleFormat == paFloat32)
-        device->FmtType = DevFmtFloat;
-    else
-    {
-        ERR("Unexpected sample format: 0x%lx\n", self->params.sampleFormat);
-        return ALC_FALSE;
-    }
-
-    if(self->params.channelCount == 2)
-        device->FmtChans = DevFmtStereo;
-    else if(self->params.channelCount == 1)
-        device->FmtChans = DevFmtMono;
-    else
-    {
-        ERR("Unexpected channel count: %u\n", self->params.channelCount);
-        return ALC_FALSE;
-    }
-    SetDefaultChannelOrder(device);
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCportPlayback_start(ALCportPlayback *self)
-{
-    PaError err;
-
-    err = Pa_StartStream(self->stream);
-    if(err != paNoError)
-    {
-        ERR("Pa_StartStream() returned an error: %s\n", Pa_GetErrorText(err));
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void ALCportPlayback_stop(ALCportPlayback *self)
-{
-    PaError err = Pa_StopStream(self->stream);
-    if(err != paNoError)
-        ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
-}
-
-
-typedef struct ALCportCapture {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    PaStream *stream;
-    PaStreamParameters params;
-
-    ll_ringbuffer_t *ring;
-} ALCportCapture;
-
-static int ALCportCapture_ReadCallback(const void *inputBuffer, void *outputBuffer,
-    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
-    const PaStreamCallbackFlags statusFlags, void *userData);
-
-static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device);
-static void ALCportCapture_Destruct(ALCportCapture *self);
-static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name);
-static void ALCportCapture_close(ALCportCapture *self);
-static DECLARE_FORWARD(ALCportCapture, ALCbackend, ALCboolean, reset)
-static ALCboolean ALCportCapture_start(ALCportCapture *self);
-static void ALCportCapture_stop(ALCportCapture *self);
-static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples);
-static ALCuint ALCportCapture_availableSamples(ALCportCapture *self);
-static DECLARE_FORWARD(ALCportCapture, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCportCapture)
-
-DEFINE_ALCBACKEND_VTABLE(ALCportCapture);
-
-
-static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCportCapture, ALCbackend, self);
-
-    self->stream = NULL;
-}
-
-static void ALCportCapture_Destruct(ALCportCapture *self)
-{
-    if(self->stream)
-        Pa_CloseStream(self->stream);
-    self->stream = NULL;
-
-    if(self->ring)
-        ll_ringbuffer_free(self->ring);
-    self->ring = NULL;
-
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-static int ALCportCapture_ReadCallback(const void *inputBuffer, void *UNUSED(outputBuffer),
-    unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo),
-    const PaStreamCallbackFlags UNUSED(statusFlags), void *userData)
-{
-    ALCportCapture *self = userData;
-    size_t writable = ll_ringbuffer_write_space(self->ring);
-
-    if(framesPerBuffer > writable)
-        framesPerBuffer = writable;
-    ll_ringbuffer_write(self->ring, inputBuffer, framesPerBuffer);
-    return 0;
-}
-
-
-static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    ALuint samples, frame_size;
-    PaError err;
-
-    if(!name)
-        name = pa_device;
-    else if(strcmp(name, pa_device) != 0)
-        return ALC_INVALID_VALUE;
-
-    samples = device->UpdateSize * device->NumUpdates;
-    samples = maxu(samples, 100 * device->Frequency / 1000);
-    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    self->ring = ll_ringbuffer_create(samples, frame_size);
-    if(self->ring == NULL) return ALC_INVALID_VALUE;
-
-    self->params.device = -1;
-    if(!ConfigValueInt(NULL, "port", "capture", &self->params.device) ||
-       self->params.device < 0)
-        self->params.device = Pa_GetDefaultInputDevice();
-    self->params.suggestedLatency = 0.0f;
-    self->params.hostApiSpecificStreamInfo = NULL;
-
-    switch(device->FmtType)
-    {
-        case DevFmtByte:
-            self->params.sampleFormat = paInt8;
-            break;
-        case DevFmtUByte:
-            self->params.sampleFormat = paUInt8;
-            break;
-        case DevFmtShort:
-            self->params.sampleFormat = paInt16;
-            break;
-        case DevFmtInt:
-            self->params.sampleFormat = paInt32;
-            break;
-        case DevFmtFloat:
-            self->params.sampleFormat = paFloat32;
-            break;
-        case DevFmtUInt:
-        case DevFmtUShort:
-            ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
-            return ALC_INVALID_VALUE;
-    }
-    self->params.channelCount = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
-
-    err = Pa_OpenStream(&self->stream, &self->params, NULL,
-        device->Frequency, paFramesPerBufferUnspecified, paNoFlag,
-        ALCportCapture_ReadCallback, self
-    );
-    if(err != paNoError)
-    {
-        ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
-        return ALC_INVALID_VALUE;
-    }
-
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCportCapture_close(ALCportCapture *self)
-{
-    PaError err = Pa_CloseStream(self->stream);
-    if(err != paNoError)
-        ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
-    self->stream = NULL;
-
-    ll_ringbuffer_free(self->ring);
-    self->ring = NULL;
-}
-
-
-static ALCboolean ALCportCapture_start(ALCportCapture *self)
-{
-    PaError err = Pa_StartStream(self->stream);
-    if(err != paNoError)
-    {
-        ERR("Error starting stream: %s\n", Pa_GetErrorText(err));
-        return ALC_FALSE;
-    }
-    return ALC_TRUE;
-}
-
-static void ALCportCapture_stop(ALCportCapture *self)
-{
-    PaError err = Pa_StopStream(self->stream);
-    if(err != paNoError)
-        ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
-}
-
-
-static ALCuint ALCportCapture_availableSamples(ALCportCapture *self)
-{
-    return ll_ringbuffer_read_space(self->ring);
-}
-
-static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples)
-{
-    ll_ringbuffer_read(self->ring, buffer, samples);
-    return ALC_NO_ERROR;
-}
-
-
-typedef struct ALCportBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCportBackendFactory;
-#define ALCPORTBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCportBackendFactory, ALCbackendFactory) } }
-
-static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory *self);
-static void ALCportBackendFactory_deinit(ALCportBackendFactory *self);
-static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory *self, ALCbackend_Type type);
-static void ALCportBackendFactory_probe(ALCportBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCportBackendFactory);
-
-
-static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory* UNUSED(self))
-{
-    if(!pa_load())
-        return ALC_FALSE;
-    return ALC_TRUE;
-}
-
-static void ALCportBackendFactory_deinit(ALCportBackendFactory* UNUSED(self))
-{
-#ifdef HAVE_DYNLOAD
-    if(pa_handle)
-    {
-        Pa_Terminate();
-        CloseLib(pa_handle);
-        pa_handle = NULL;
-    }
-#else
-    Pa_Terminate();
-#endif
-}
-
-static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback || type == ALCbackend_Capture)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCportBackendFactory_probe(ALCportBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(pa_device);
-            break;
-        case CAPTURE_DEVICE_PROBE:
-            AppendCaptureDeviceList(pa_device);
-            break;
-    }
-}
-
-static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCportPlayback *backend;
-        NEW_OBJ(backend, ALCportPlayback)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-    if(type == ALCbackend_Capture)
-    {
-        ALCportCapture *backend;
-        NEW_OBJ(backend, ALCportCapture)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}
-
-ALCbackendFactory *ALCportBackendFactory_getFactory(void)
-{
-    static ALCportBackendFactory factory = ALCPORTBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}

Alc/backends/sndio.c

@@ -1,346 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "threads.h"
-
-#include "backends/base.h"
-
-#include <sndio.h>
-
-
-
-
-typedef struct ALCsndioBackend {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    struct sio_hdl *sndHandle;
-
-    ALvoid *mix_data;
-    ALsizei data_size;
-
-    volatile int killNow;
-    althrd_t thread;
-} ALCsndioBackend;
-
-static int ALCsndioBackend_mixerProc(void *ptr);
-
-static void ALCsndioBackend_Construct(ALCsndioBackend *self, ALCdevice *device);
-static void ALCsndioBackend_Destruct(ALCsndioBackend *self);
-static ALCenum ALCsndioBackend_open(ALCsndioBackend *self, const ALCchar *name);
-static void ALCsndioBackend_close(ALCsndioBackend *self);
-static ALCboolean ALCsndioBackend_reset(ALCsndioBackend *self);
-static ALCboolean ALCsndioBackend_start(ALCsndioBackend *self);
-static void ALCsndioBackend_stop(ALCsndioBackend *self);
-static DECLARE_FORWARD2(ALCsndioBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
-static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCsndioBackend, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCsndioBackend)
-
-DEFINE_ALCBACKEND_VTABLE(ALCsndioBackend);
-
-
-static const ALCchar sndio_device[] = "SndIO Default";
-
-
-static void ALCsndioBackend_Construct(ALCsndioBackend *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCsndioBackend, ALCbackend, self);
-}
-
-static void ALCsndioBackend_Destruct(ALCsndioBackend *self)
-{
-    if(self->sndHandle)
-        sio_close(self->sndHandle);
-    self->sndHandle = NULL;
-
-    al_free(self->mix_data);
-    self->mix_data = NULL;
-
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-static int ALCsndioBackend_mixerProc(void *ptr)
-{
-    ALCsndioBackend *self = (ALCsndioBackend*)ptr;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    ALsizei frameSize;
-    size_t wrote;
-
-    SetRTPriority();
-    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
-
-    frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    while(!self->killNow && device->Connected)
-    {
-        ALsizei len = self->data_size;
-        ALubyte *WritePtr = self->mix_data;
-
-        ALCsndioBackend_lock(self);
-        aluMixData(device, WritePtr, len/frameSize);
-        ALCsndioBackend_unlock(self);
-        while(len > 0 && !self->killNow)
-        {
-            wrote = sio_write(self->sndHandle, WritePtr, len);
-            if(wrote == 0)
-            {
-                ERR("sio_write failed\n");
-                ALCdevice_Lock(device);
-                aluHandleDisconnect(device);
-                ALCdevice_Unlock(device);
-                break;
-            }
-
-            len -= wrote;
-            WritePtr += wrote;
-        }
-    }
-
-    return 0;
-}
-
-
-static ALCenum ALCsndioBackend_open(ALCsndioBackend *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-
-    if(!name)
-        name = sndio_device;
-    else if(strcmp(name, sndio_device) != 0)
-        return ALC_INVALID_VALUE;
-
-    self->sndHandle = sio_open(NULL, SIO_PLAY, 0);
-    if(self->sndHandle == NULL)
-    {
-        ERR("Could not open device\n");
-        return ALC_INVALID_VALUE;
-    }
-
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCsndioBackend_close(ALCsndioBackend *self)
-{
-    sio_close(self->sndHandle);
-    self->sndHandle = NULL;
-}
-
-static ALCboolean ALCsndioBackend_reset(ALCsndioBackend *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-    struct sio_par par;
-
-    sio_initpar(&par);
-
-    par.rate = device->Frequency;
-    par.pchan = ((device->FmtChans != DevFmtMono) ? 2 : 1);
-
-    switch(device->FmtType)
-    {
-        case DevFmtByte:
-            par.bits = 8;
-            par.sig = 1;
-            break;
-        case DevFmtUByte:
-            par.bits = 8;
-            par.sig = 0;
-            break;
-        case DevFmtFloat:
-        case DevFmtShort:
-            par.bits = 16;
-            par.sig = 1;
-            break;
-        case DevFmtUShort:
-            par.bits = 16;
-            par.sig = 0;
-            break;
-        case DevFmtInt:
-            par.bits = 32;
-            par.sig = 1;
-            break;
-        case DevFmtUInt:
-            par.bits = 32;
-            par.sig = 0;
-            break;
-    }
-    par.le = SIO_LE_NATIVE;
-
-    par.round = device->UpdateSize;
-    par.appbufsz = device->UpdateSize * (device->NumUpdates-1);
-    if(!par.appbufsz) par.appbufsz = device->UpdateSize;
-
-    if(!sio_setpar(self->sndHandle, &par) || !sio_getpar(self->sndHandle, &par))
-    {
-        ERR("Failed to set device parameters\n");
-        return ALC_FALSE;
-    }
-
-    if(par.bits != par.bps*8)
-    {
-        ERR("Padded samples not supported (%u of %u bits)\n", par.bits, par.bps*8);
-        return ALC_FALSE;
-    }
-
-    device->Frequency = par.rate;
-    device->FmtChans = ((par.pchan==1) ? DevFmtMono : DevFmtStereo);
-
-    if(par.bits == 8 && par.sig == 1)
-        device->FmtType = DevFmtByte;
-    else if(par.bits == 8 && par.sig == 0)
-        device->FmtType = DevFmtUByte;
-    else if(par.bits == 16 && par.sig == 1)
-        device->FmtType = DevFmtShort;
-    else if(par.bits == 16 && par.sig == 0)
-        device->FmtType = DevFmtUShort;
-    else if(par.bits == 32 && par.sig == 1)
-        device->FmtType = DevFmtInt;
-    else if(par.bits == 32 && par.sig == 0)
-        device->FmtType = DevFmtUInt;
-    else
-    {
-        ERR("Unhandled sample format: %s %u-bit\n", (par.sig?"signed":"unsigned"), par.bits);
-        return ALC_FALSE;
-    }
-
-    device->UpdateSize = par.round;
-    device->NumUpdates = (par.bufsz/par.round) + 1;
-
-    SetDefaultChannelOrder(device);
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCsndioBackend_start(ALCsndioBackend *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-
-    self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
-        device->FmtChans, device->FmtType, device->AmbiOrder
-    );
-    al_free(self->mix_data);
-    self->mix_data = al_calloc(16, self->data_size);
-
-    if(!sio_start(self->sndHandle))
-    {
-        ERR("Error starting playback\n");
-        return ALC_FALSE;
-    }
-
-    self->killNow = 0;
-    if(althrd_create(&self->thread, ALCsndioBackend_mixerProc, self) != althrd_success)
-    {
-        sio_stop(self->sndHandle);
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void ALCsndioBackend_stop(ALCsndioBackend *self)
-{
-    int res;
-
-    if(self->killNow)
-        return;
-
-    self->killNow = 1;
-    althrd_join(self->thread, &res);
-
-    if(!sio_stop(self->sndHandle))
-        ERR("Error stopping device\n");
-
-    al_free(self->mix_data);
-    self->mix_data = NULL;
-}
-
-
-typedef struct ALCsndioBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCsndioBackendFactory;
-#define ALCSNDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsndioBackendFactory, ALCbackendFactory) } }
-
-ALCbackendFactory *ALCsndioBackendFactory_getFactory(void);
-
-static ALCboolean ALCsndioBackendFactory_init(ALCsndioBackendFactory *self);
-static DECLARE_FORWARD(ALCsndioBackendFactory, ALCbackendFactory, void, deinit)
-static ALCboolean ALCsndioBackendFactory_querySupport(ALCsndioBackendFactory *self, ALCbackend_Type type);
-static void ALCsndioBackendFactory_probe(ALCsndioBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCsndioBackendFactory_createBackend(ALCsndioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsndioBackendFactory);
-
-
-ALCbackendFactory *ALCsndioBackendFactory_getFactory(void)
-{
-    static ALCsndioBackendFactory factory = ALCSNDIOBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}
-
-
-static ALCboolean ALCsndioBackendFactory_init(ALCsndioBackendFactory* UNUSED(self))
-{
-    /* No dynamic loading */
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCsndioBackendFactory_querySupport(ALCsndioBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCsndioBackendFactory_probe(ALCsndioBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(sndio_device);
-            break;
-        case CAPTURE_DEVICE_PROBE:
-            break;
-    }
-}
-
-static ALCbackend* ALCsndioBackendFactory_createBackend(ALCsndioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCsndioBackend *backend;
-        NEW_OBJ(backend, ALCsndioBackend)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}

Alc/backends/solaris.c

@@ -1,363 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <sys/ioctl.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-#include <unistd.h>
-#include <errno.h>
-#include <math.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "threads.h"
-#include "compat.h"
-
-#include "backends/base.h"
-
-#include <sys/audioio.h>
-
-
-typedef struct ALCsolarisBackend {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    int fd;
-
-    ALubyte *mix_data;
-    int data_size;
-
-    ATOMIC(ALenum) killNow;
-    althrd_t thread;
-} ALCsolarisBackend;
-
-static int ALCsolarisBackend_mixerProc(void *ptr);
-
-static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device);
-static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self);
-static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name);
-static void ALCsolarisBackend_close(ALCsolarisBackend *self);
-static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self);
-static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self);
-static void ALCsolarisBackend_stop(ALCsolarisBackend *self);
-static DECLARE_FORWARD2(ALCsolarisBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
-static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCsolarisBackend)
-
-DEFINE_ALCBACKEND_VTABLE(ALCsolarisBackend);
-
-
-static const ALCchar solaris_device[] = "Solaris Default";
-
-static const char *solaris_driver = "/dev/audio";
-
-
-static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCsolarisBackend, ALCbackend, self);
-
-    self->fd = -1;
-    ATOMIC_INIT(&self->killNow, AL_FALSE);
-}
-
-static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self)
-{
-    if(self->fd != -1)
-        close(self->fd);
-    self->fd = -1;
-
-    free(self->mix_data);
-    self->mix_data = NULL;
-    self->data_size = 0;
-
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-static int ALCsolarisBackend_mixerProc(void *ptr)
-{
-    ALCsolarisBackend *self = ptr;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    struct timeval timeout;
-    ALubyte *write_ptr;
-    ALint frame_size;
-    ALint to_write;
-    ssize_t wrote;
-    fd_set wfds;
-    int sret;
-
-    SetRTPriority();
-    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
-
-    frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    ALCsolarisBackend_lock(self);
-    while(!ATOMIC_LOAD_SEQ(&self->killNow) && device->Connected)
-    {
-        FD_ZERO(&wfds);
-        FD_SET(self->fd, &wfds);
-        timeout.tv_sec = 1;
-        timeout.tv_usec = 0;
-
-        ALCsolarisBackend_unlock(self);
-        sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
-        ALCsolarisBackend_lock(self);
-        if(sret < 0)
-        {
-            if(errno == EINTR)
-                continue;
-            ERR("select failed: %s\n", strerror(errno));
-            aluHandleDisconnect(device);
-            break;
-        }
-        else if(sret == 0)
-        {
-            WARN("select timeout\n");
-            continue;
-        }
-
-        write_ptr = self->mix_data;
-        to_write = self->data_size;
-        aluMixData(device, write_ptr, to_write/frame_size);
-        while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow))
-        {
-            wrote = write(self->fd, write_ptr, to_write);
-            if(wrote < 0)
-            {
-                if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
-                    continue;
-                ERR("write failed: %s\n", strerror(errno));
-                aluHandleDisconnect(device);
-                break;
-            }
-
-            to_write -= wrote;
-            write_ptr += wrote;
-        }
-    }
-    ALCsolarisBackend_unlock(self);
-
-    return 0;
-}
-
-
-static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name)
-{
-    ALCdevice *device;
-
-    if(!name)
-        name = solaris_device;
-    else if(strcmp(name, solaris_device) != 0)
-        return ALC_INVALID_VALUE;
-
-    self->fd = open(solaris_driver, O_WRONLY);
-    if(self->fd == -1)
-    {
-        ERR("Could not open %s: %s\n", solaris_driver, strerror(errno));
-        return ALC_INVALID_VALUE;
-    }
-
-    device = STATIC_CAST(ALCbackend,self)->mDevice;
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCsolarisBackend_close(ALCsolarisBackend *self)
-{
-    close(self->fd);
-    self->fd = -1;
-}
-
-static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
-    audio_info_t info;
-    ALsizei frameSize;
-    ALsizei numChannels;
-
-    AUDIO_INITINFO(&info);
-
-    info.play.sample_rate = device->Frequency;
-
-    if(device->FmtChans != DevFmtMono)
-        device->FmtChans = DevFmtStereo;
-    numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
-    info.play.channels = numChannels;
-
-    switch(device->FmtType)
-    {
-        case DevFmtByte:
-            info.play.precision = 8;
-            info.play.encoding = AUDIO_ENCODING_LINEAR;
-            break;
-        case DevFmtUByte:
-            info.play.precision = 8;
-            info.play.encoding = AUDIO_ENCODING_LINEAR8;
-            break;
-        case DevFmtUShort:
-        case DevFmtInt:
-        case DevFmtUInt:
-        case DevFmtFloat:
-            device->FmtType = DevFmtShort;
-            /* fall-through */
-        case DevFmtShort:
-            info.play.precision = 16;
-            info.play.encoding = AUDIO_ENCODING_LINEAR;
-            break;
-    }
-
-    frameSize = numChannels * BytesFromDevFmt(device->FmtType);
-    info.play.buffer_size = device->UpdateSize*device->NumUpdates * frameSize;
-
-    if(ioctl(self->fd, AUDIO_SETINFO, &info) < 0)
-    {
-        ERR("ioctl failed: %s\n", strerror(errno));
-        return ALC_FALSE;
-    }
-
-    if(ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)info.play.channels)
-    {
-        ERR("Failed to set %s, got %u channels instead\n", DevFmtChannelsString(device->FmtChans), info.play.channels);
-        return ALC_FALSE;
-    }
-
-    if(!((info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR8 && device->FmtType == DevFmtUByte) ||
-         (info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtByte) ||
-         (info.play.precision == 16 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtShort) ||
-         (info.play.precision == 32 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtInt)))
-    {
-        ERR("Could not set %s samples, got %d (0x%x)\n", DevFmtTypeString(device->FmtType),
-            info.play.precision, info.play.encoding);
-        return ALC_FALSE;
-    }
-
-    device->Frequency = info.play.sample_rate;
-    device->UpdateSize = (info.play.buffer_size/device->NumUpdates) + 1;
-
-    SetDefaultChannelOrder(device);
-
-    free(self->mix_data);
-    self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
-        device->FmtChans, device->FmtType, device->AmbiOrder
-    );
-    self->mix_data = calloc(1, self->data_size);
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self)
-{
-    ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
-    if(althrd_create(&self->thread, ALCsolarisBackend_mixerProc, self) != althrd_success)
-        return ALC_FALSE;
-    return ALC_TRUE;
-}
-
-static void ALCsolarisBackend_stop(ALCsolarisBackend *self)
-{
-    int res;
-
-    if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
-        return;
-
-    althrd_join(self->thread, &res);
-
-    if(ioctl(self->fd, AUDIO_DRAIN) < 0)
-        ERR("Error draining device: %s\n", strerror(errno));
-}
-
-
-typedef struct ALCsolarisBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCsolarisBackendFactory;
-#define ALCSOLARISBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsolarisBackendFactory, ALCbackendFactory) } }
-
-ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void);
-
-static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory *self);
-static DECLARE_FORWARD(ALCsolarisBackendFactory, ALCbackendFactory, void, deinit)
-static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory *self, ALCbackend_Type type);
-static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsolarisBackendFactory);
-
-
-ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void)
-{
-    static ALCsolarisBackendFactory factory = ALCSOLARISBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}
-
-
-static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory* UNUSED(self))
-{
-    ConfigValueStr(NULL, "solaris", "device", &solaris_driver);
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-        {
-#ifdef HAVE_STAT
-            struct stat buf;
-            if(stat(solaris_driver, &buf) == 0)
-#endif
-                AppendAllDevicesList(solaris_device);
-        }
-        break;
-
-        case CAPTURE_DEVICE_PROBE:
-            break;
-    }
-}
-
-ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCsolarisBackend *backend;
-        NEW_OBJ(backend, ALCsolarisBackend)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}

Alc/backends/wave.c

@@ -1,453 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-#include <errno.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "threads.h"
-#include "compat.h"
-
-#include "backends/base.h"
-
-
-static const ALCchar waveDevice[] = "Wave File Writer";
-
-static const ALubyte SUBTYPE_PCM[] = {
-    0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa,
-    0x00, 0x38, 0x9b, 0x71
-};
-static const ALubyte SUBTYPE_FLOAT[] = {
-    0x03, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa,
-    0x00, 0x38, 0x9b, 0x71
-};
-
-static const ALubyte SUBTYPE_BFORMAT_PCM[] = {
-    0x01, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1,
-    0xca, 0x00, 0x00, 0x00
-};
-
-static const ALubyte SUBTYPE_BFORMAT_FLOAT[] = {
-    0x03, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1,
-    0xca, 0x00, 0x00, 0x00
-};
-
-static void fwrite16le(ALushort val, FILE *f)
-{
-    ALubyte data[2] = { val&0xff, (val>>8)&0xff };
-    fwrite(data, 1, 2, f);
-}
-
-static void fwrite32le(ALuint val, FILE *f)
-{
-    ALubyte data[4] = { val&0xff, (val>>8)&0xff, (val>>16)&0xff, (val>>24)&0xff };
-    fwrite(data, 1, 4, f);
-}
-
-
-typedef struct ALCwaveBackend {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    FILE *mFile;
-    long mDataStart;
-
-    ALvoid *mBuffer;
-    ALuint mSize;
-
-    volatile int killNow;
-    althrd_t thread;
-} ALCwaveBackend;
-
-static int ALCwaveBackend_mixerProc(void *ptr);
-
-static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device);
-static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, Destruct)
-static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name);
-static void ALCwaveBackend_close(ALCwaveBackend *self);
-static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self);
-static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self);
-static void ALCwaveBackend_stop(ALCwaveBackend *self);
-static DECLARE_FORWARD2(ALCwaveBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
-static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCwaveBackend)
-
-DEFINE_ALCBACKEND_VTABLE(ALCwaveBackend);
-
-
-static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCwaveBackend, ALCbackend, self);
-
-    self->mFile = NULL;
-    self->mDataStart = -1;
-
-    self->mBuffer = NULL;
-    self->mSize = 0;
-
-    self->killNow = 1;
-}
-
-
-static int ALCwaveBackend_mixerProc(void *ptr)
-{
-    ALCwaveBackend *self = (ALCwaveBackend*)ptr;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    struct timespec now, start;
-    ALint64 avail, done;
-    ALuint frameSize;
-    size_t fs;
-    const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 /
-                                 device->Frequency / 2);
-
-    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
-
-    frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    done = 0;
-    if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
-    {
-        ERR("Failed to get starting time\n");
-        return 1;
-    }
-    while(!self->killNow && device->Connected)
-    {
-        if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC)
-        {
-            ERR("Failed to get current time\n");
-            return 1;
-        }
-
-        avail  = (now.tv_sec - start.tv_sec) * device->Frequency;
-        avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
-        if(avail < done)
-        {
-            /* Oops, time skipped backwards. Reset the number of samples done
-             * with one update available since we (likely) just came back from
-             * sleeping. */
-            done = avail - device->UpdateSize;
-        }
-
-        if(avail-done < device->UpdateSize)
-            al_nssleep(restTime);
-        else while(avail-done >= device->UpdateSize)
-        {
-            ALCwaveBackend_lock(self);
-            aluMixData(device, self->mBuffer, device->UpdateSize);
-            ALCwaveBackend_unlock(self);
-            done += device->UpdateSize;
-
-            if(!IS_LITTLE_ENDIAN)
-            {
-                ALuint bytesize = BytesFromDevFmt(device->FmtType);
-                ALuint i;
-
-                if(bytesize == 2)
-                {
-                    ALushort *samples = self->mBuffer;
-                    ALuint len = self->mSize / 2;
-                    for(i = 0;i < len;i++)
-                    {
-                        ALushort samp = samples[i];
-                        samples[i] = (samp>>8) | (samp<<8);
-                    }
-                }
-                else if(bytesize == 4)
-                {
-                    ALuint *samples = self->mBuffer;
-                    ALuint len = self->mSize / 4;
-                    for(i = 0;i < len;i++)
-                    {
-                        ALuint samp = samples[i];
-                        samples[i] = (samp>>24) | ((samp>>8)&0x0000ff00) |
-                                     ((samp<<8)&0x00ff0000) | (samp<<24);
-                    }
-                }
-            }
-
-            fs = fwrite(self->mBuffer, frameSize, device->UpdateSize, self->mFile);
-            (void)fs;
-            if(ferror(self->mFile))
-            {
-                ERR("Error writing to file\n");
-                ALCdevice_Lock(device);
-                aluHandleDisconnect(device);
-                ALCdevice_Unlock(device);
-                break;
-            }
-        }
-    }
-
-    return 0;
-}
-
-
-static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name)
-{
-    ALCdevice *device;
-    const char *fname;
-
-    fname = GetConfigValue(NULL, "wave", "file", "");
-    if(!fname[0]) return ALC_INVALID_VALUE;
-
-    if(!name)
-        name = waveDevice;
-    else if(strcmp(name, waveDevice) != 0)
-        return ALC_INVALID_VALUE;
-
-    self->mFile = al_fopen(fname, "wb");
-    if(!self->mFile)
-    {
-        ERR("Could not open file '%s': %s\n", fname, strerror(errno));
-        return ALC_INVALID_VALUE;
-    }
-
-    device = STATIC_CAST(ALCbackend, self)->mDevice;
-    alstr_copy_cstr(&device->DeviceName, name);
-
-    return ALC_NO_ERROR;
-}
-
-static void ALCwaveBackend_close(ALCwaveBackend *self)
-{
-    if(self->mFile)
-        fclose(self->mFile);
-    self->mFile = NULL;
-}
-
-static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    ALuint channels=0, bits=0, chanmask=0;
-    int isbformat = 0;
-    size_t val;
-
-    fseek(self->mFile, 0, SEEK_SET);
-    clearerr(self->mFile);
-
-    if(GetConfigValueBool(NULL, "wave", "bformat", 0))
-    {
-        device->FmtChans = DevFmtAmbi3D;
-        device->AmbiOrder = 1;
-    }
-
-    switch(device->FmtType)
-    {
-        case DevFmtByte:
-            device->FmtType = DevFmtUByte;
-            break;
-        case DevFmtUShort:
-            device->FmtType = DevFmtShort;
-            break;
-        case DevFmtUInt:
-            device->FmtType = DevFmtInt;
-            break;
-        case DevFmtUByte:
-        case DevFmtShort:
-        case DevFmtInt:
-        case DevFmtFloat:
-            break;
-    }
-    switch(device->FmtChans)
-    {
-        case DevFmtMono:   chanmask = 0x04; break;
-        case DevFmtStereo: chanmask = 0x01 | 0x02; break;
-        case DevFmtQuad:   chanmask = 0x01 | 0x02 | 0x10 | 0x20; break;
-        case DevFmtX51: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x200 | 0x400; break;
-        case DevFmtX51Rear: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020; break;
-        case DevFmtX61: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x100 | 0x200 | 0x400; break;
-        case DevFmtX71: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020 | 0x200 | 0x400; break;
-        case DevFmtAmbi3D:
-            /* .amb output requires FuMa */
-            device->AmbiLayout = AmbiLayout_FuMa;
-            device->AmbiScale = AmbiNorm_FuMa;
-            isbformat = 1;
-            chanmask = 0;
-            break;
-    }
-    bits = BytesFromDevFmt(device->FmtType) * 8;
-    channels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
-
-    fputs("RIFF", self->mFile);
-    fwrite32le(0xFFFFFFFF, self->mFile); // 'RIFF' header len; filled in at close
-
-    fputs("WAVE", self->mFile);
-
-    fputs("fmt ", self->mFile);
-    fwrite32le(40, self->mFile); // 'fmt ' header len; 40 bytes for EXTENSIBLE
-
-    // 16-bit val, format type id (extensible: 0xFFFE)
-    fwrite16le(0xFFFE, self->mFile);
-    // 16-bit val, channel count
-    fwrite16le(channels, self->mFile);
-    // 32-bit val, frequency
-    fwrite32le(device->Frequency, self->mFile);
-    // 32-bit val, bytes per second
-    fwrite32le(device->Frequency * channels * bits / 8, self->mFile);
-    // 16-bit val, frame size
-    fwrite16le(channels * bits / 8, self->mFile);
-    // 16-bit val, bits per sample
-    fwrite16le(bits, self->mFile);
-    // 16-bit val, extra byte count
-    fwrite16le(22, self->mFile);
-    // 16-bit val, valid bits per sample
-    fwrite16le(bits, self->mFile);
-    // 32-bit val, channel mask
-    fwrite32le(chanmask, self->mFile);
-    // 16 byte GUID, sub-type format
-    val = fwrite((device->FmtType == DevFmtFloat) ?
-                 (isbformat ? SUBTYPE_BFORMAT_FLOAT : SUBTYPE_FLOAT) :
-                 (isbformat ? SUBTYPE_BFORMAT_PCM : SUBTYPE_PCM), 1, 16, self->mFile);
-    (void)val;
-
-    fputs("data", self->mFile);
-    fwrite32le(0xFFFFFFFF, self->mFile); // 'data' header len; filled in at close
-
-    if(ferror(self->mFile))
-    {
-        ERR("Error writing header: %s\n", strerror(errno));
-        return ALC_FALSE;
-    }
-    self->mDataStart = ftell(self->mFile);
-
-    SetDefaultWFXChannelOrder(device);
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-
-    self->mSize = device->UpdateSize * FrameSizeFromDevFmt(
-        device->FmtChans, device->FmtType, device->AmbiOrder
-    );
-    self->mBuffer = malloc(self->mSize);
-    if(!self->mBuffer)
-    {
-        ERR("Buffer malloc failed\n");
-        return ALC_FALSE;
-    }
-
-    self->killNow = 0;
-    if(althrd_create(&self->thread, ALCwaveBackend_mixerProc, self) != althrd_success)
-    {
-        free(self->mBuffer);
-        self->mBuffer = NULL;
-        self->mSize = 0;
-        return ALC_FALSE;
-    }
-
-    return ALC_TRUE;
-}
-
-static void ALCwaveBackend_stop(ALCwaveBackend *self)
-{
-    ALuint dataLen;
-    long size;
-    int res;
-
-    if(self->killNow)
-        return;
-
-    self->killNow = 1;
-    althrd_join(self->thread, &res);
-
-    free(self->mBuffer);
-    self->mBuffer = NULL;
-
-    size = ftell(self->mFile);
-    if(size > 0)
-    {
-        dataLen = size - self->mDataStart;
-        if(fseek(self->mFile, self->mDataStart-4, SEEK_SET) == 0)
-            fwrite32le(dataLen, self->mFile); // 'data' header len
-        if(fseek(self->mFile, 4, SEEK_SET) == 0)
-            fwrite32le(size-8, self->mFile); // 'WAVE' header len
-    }
-}
-
-
-typedef struct ALCwaveBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCwaveBackendFactory;
-#define ALCWAVEBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwaveBackendFactory, ALCbackendFactory) } }
-
-ALCbackendFactory *ALCwaveBackendFactory_getFactory(void);
-
-static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory *self);
-static DECLARE_FORWARD(ALCwaveBackendFactory, ALCbackendFactory, void, deinit)
-static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory *self, ALCbackend_Type type);
-static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwaveBackendFactory);
-
-
-ALCbackendFactory *ALCwaveBackendFactory_getFactory(void)
-{
-    static ALCwaveBackendFactory factory = ALCWAVEBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}
-
-
-static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory* UNUSED(self))
-{
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-        return !!ConfigValueExists(NULL, "wave", "file");
-    return ALC_FALSE;
-}
-
-static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            AppendAllDevicesList(waveDevice);
-            break;
-        case CAPTURE_DEVICE_PROBE:
-            break;
-    }
-}
-
-static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCwaveBackend *backend;
-        NEW_OBJ(backend, ALCwaveBackend)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}

Alc/backends/winmm.c

@@ -1,803 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <memory.h>
-
-#include <windows.h>
-#include <mmsystem.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "threads.h"
-
-#include "backends/base.h"
-
-#ifndef WAVE_FORMAT_IEEE_FLOAT
-#define WAVE_FORMAT_IEEE_FLOAT  0x0003
-#endif
-
-#define DEVNAME_HEAD "OpenAL Soft on "
-
-
-static vector_al_string PlaybackDevices;
-static vector_al_string CaptureDevices;
-
-static void clear_devlist(vector_al_string *list)
-{
-    VECTOR_FOR_EACH(al_string, *list, alstr_reset);
-    VECTOR_RESIZE(*list, 0, 0);
-}
-
-
-static void ProbePlaybackDevices(void)
-{
-    ALuint numdevs;
-    ALuint i;
-
-    clear_devlist(&PlaybackDevices);
-
-    numdevs = waveOutGetNumDevs();
-    VECTOR_RESIZE(PlaybackDevices, 0, numdevs);
-    for(i = 0;i < numdevs;i++)
-    {
-        WAVEOUTCAPSW WaveCaps;
-        const al_string *iter;
-        al_string dname;
-
-        AL_STRING_INIT(dname);
-        if(waveOutGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR)
-        {
-            ALuint count = 0;
-            while(1)
-            {
-                alstr_copy_cstr(&dname, DEVNAME_HEAD);
-                alstr_append_wcstr(&dname, WaveCaps.szPname);
-                if(count != 0)
-                {
-                    char str[64];
-                    snprintf(str, sizeof(str), " #%d", count+1);
-                    alstr_append_cstr(&dname, str);
-                }
-                count++;
-
-#define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0)
-                VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_ENTRY);
-                if(iter == VECTOR_END(PlaybackDevices)) break;
-#undef MATCH_ENTRY
-            }
-
-            TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i);
-        }
-        VECTOR_PUSH_BACK(PlaybackDevices, dname);
-    }
-}
-
-static void ProbeCaptureDevices(void)
-{
-    ALuint numdevs;
-    ALuint i;
-
-    clear_devlist(&CaptureDevices);
-
-    numdevs = waveInGetNumDevs();
-    VECTOR_RESIZE(CaptureDevices, 0, numdevs);
-    for(i = 0;i < numdevs;i++)
-    {
-        WAVEINCAPSW WaveCaps;
-        const al_string *iter;
-        al_string dname;
-
-        AL_STRING_INIT(dname);
-        if(waveInGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR)
-        {
-            ALuint count = 0;
-            while(1)
-            {
-                alstr_copy_cstr(&dname, DEVNAME_HEAD);
-                alstr_append_wcstr(&dname, WaveCaps.szPname);
-                if(count != 0)
-                {
-                    char str[64];
-                    snprintf(str, sizeof(str), " #%d", count+1);
-                    alstr_append_cstr(&dname, str);
-                }
-                count++;
-
-#define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0)
-                VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_ENTRY);
-                if(iter == VECTOR_END(CaptureDevices)) break;
-#undef MATCH_ENTRY
-            }
-
-            TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i);
-        }
-        VECTOR_PUSH_BACK(CaptureDevices, dname);
-    }
-}
-
-
-typedef struct ALCwinmmPlayback {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    RefCount WaveBuffersCommitted;
-    WAVEHDR WaveBuffer[4];
-
-    HWAVEOUT OutHdl;
-
-    WAVEFORMATEX Format;
-
-    volatile ALboolean killNow;
-    althrd_t thread;
-} ALCwinmmPlayback;
-
-static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device);
-static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self);
-
-static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2);
-static int ALCwinmmPlayback_mixerProc(void *arg);
-
-static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *name);
-static void ALCwinmmPlayback_close(ALCwinmmPlayback *self);
-static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self);
-static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self);
-static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self);
-static DECLARE_FORWARD2(ALCwinmmPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
-static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ALCuint, availableSamples)
-static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCwinmmPlayback)
-
-DEFINE_ALCBACKEND_VTABLE(ALCwinmmPlayback);
-
-
-static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCwinmmPlayback, ALCbackend, self);
-
-    InitRef(&self->WaveBuffersCommitted, 0);
-    self->OutHdl = NULL;
-
-    self->killNow = AL_TRUE;
-}
-
-static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self)
-{
-    if(self->OutHdl)
-        waveOutClose(self->OutHdl);
-    self->OutHdl = 0;
-
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-/* ALCwinmmPlayback_waveOutProc
- *
- * Posts a message to 'ALCwinmmPlayback_mixerProc' everytime a WaveOut Buffer
- * is completed and returns to the application (for more data)
- */
-static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2))
-{
-    ALCwinmmPlayback *self = (ALCwinmmPlayback*)instance;
-
-    if(msg != WOM_DONE)
-        return;
-
-    DecrementRef(&self->WaveBuffersCommitted);
-    PostThreadMessage(self->thread, msg, 0, param1);
-}
-
-FORCE_ALIGN static int ALCwinmmPlayback_mixerProc(void *arg)
-{
-    ALCwinmmPlayback *self = arg;
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    WAVEHDR *WaveHdr;
-    MSG msg;
-
-    SetRTPriority();
-    althrd_setname(althrd_current(), MIXER_THREAD_NAME);
-
-    while(GetMessage(&msg, NULL, 0, 0))
-    {
-        if(msg.message != WOM_DONE)
-            continue;
-
-        if(self->killNow)
-        {
-            if(ReadRef(&self->WaveBuffersCommitted) == 0)
-                break;
-            continue;
-        }
-
-        WaveHdr = ((WAVEHDR*)msg.lParam);
-        ALCwinmmPlayback_lock(self);
-        aluMixData(device, WaveHdr->lpData, WaveHdr->dwBufferLength /
-                                            self->Format.nBlockAlign);
-        ALCwinmmPlayback_unlock(self);
-
-        // Send buffer back to play more data
-        waveOutWrite(self->OutHdl, WaveHdr, sizeof(WAVEHDR));
-        IncrementRef(&self->WaveBuffersCommitted);
-    }
-
-    return 0;
-}
-
-
-static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *deviceName)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    const al_string *iter;
-    UINT DeviceID;
-    MMRESULT res;
-
-    if(VECTOR_SIZE(PlaybackDevices) == 0)
-        ProbePlaybackDevices();
-
-    // Find the Device ID matching the deviceName if valid
-#define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && \
-                             (!deviceName || alstr_cmp_cstr(*(iter), deviceName) == 0))
-    VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_DEVNAME);
-    if(iter == VECTOR_END(PlaybackDevices))
-        return ALC_INVALID_VALUE;
-#undef MATCH_DEVNAME
-
-    DeviceID = (UINT)(iter - VECTOR_BEGIN(PlaybackDevices));
-
-retry_open:
-    memset(&self->Format, 0, sizeof(WAVEFORMATEX));
-    if(device->FmtType == DevFmtFloat)
-    {
-        self->Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
-        self->Format.wBitsPerSample = 32;
-    }
-    else
-    {
-        self->Format.wFormatTag = WAVE_FORMAT_PCM;
-        if(device->FmtType == DevFmtUByte || device->FmtType == DevFmtByte)
-            self->Format.wBitsPerSample = 8;
-        else
-            self->Format.wBitsPerSample = 16;
-    }
-    self->Format.nChannels = ((device->FmtChans == DevFmtMono) ? 1 : 2);
-    self->Format.nBlockAlign = self->Format.wBitsPerSample *
-                               self->Format.nChannels / 8;
-    self->Format.nSamplesPerSec = device->Frequency;
-    self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec *
-                                   self->Format.nBlockAlign;
-    self->Format.cbSize = 0;
-
-    if((res=waveOutOpen(&self->OutHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmPlayback_waveOutProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
-    {
-        if(device->FmtType == DevFmtFloat)
-        {
-            device->FmtType = DevFmtShort;
-            goto retry_open;
-        }
-        ERR("waveOutOpen failed: %u\n", res);
-        goto failure;
-    }
-
-    alstr_copy(&device->DeviceName, VECTOR_ELEM(PlaybackDevices, DeviceID));
-    return ALC_NO_ERROR;
-
-failure:
-    if(self->OutHdl)
-        waveOutClose(self->OutHdl);
-    self->OutHdl = NULL;
-
-    return ALC_INVALID_VALUE;
-}
-
-static void ALCwinmmPlayback_close(ALCwinmmPlayback* UNUSED(self))
-{ }
-
-static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-
-    device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
-                                  self->Format.nSamplesPerSec /
-                                  device->Frequency);
-    device->UpdateSize = (device->UpdateSize*device->NumUpdates + 3) / 4;
-    device->NumUpdates = 4;
-    device->Frequency = self->Format.nSamplesPerSec;
-
-    if(self->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
-    {
-        if(self->Format.wBitsPerSample == 32)
-            device->FmtType = DevFmtFloat;
-        else
-        {
-            ERR("Unhandled IEEE float sample depth: %d\n", self->Format.wBitsPerSample);
-            return ALC_FALSE;
-        }
-    }
-    else if(self->Format.wFormatTag == WAVE_FORMAT_PCM)
-    {
-        if(self->Format.wBitsPerSample == 16)
-            device->FmtType = DevFmtShort;
-        else if(self->Format.wBitsPerSample == 8)
-            device->FmtType = DevFmtUByte;
-        else
-        {
-            ERR("Unhandled PCM sample depth: %d\n", self->Format.wBitsPerSample);
-            return ALC_FALSE;
-        }
-    }
-    else
-    {
-        ERR("Unhandled format tag: 0x%04x\n", self->Format.wFormatTag);
-        return ALC_FALSE;
-    }
-
-    if(self->Format.nChannels == 2)
-        device->FmtChans = DevFmtStereo;
-    else if(self->Format.nChannels == 1)
-        device->FmtChans = DevFmtMono;
-    else
-    {
-        ERR("Unhandled channel count: %d\n", self->Format.nChannels);
-        return ALC_FALSE;
-    }
-    SetDefaultWFXChannelOrder(device);
-
-    return ALC_TRUE;
-}
-
-static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    ALbyte *BufferData;
-    ALint BufferSize;
-    ALuint i;
-
-    self->killNow = AL_FALSE;
-    if(althrd_create(&self->thread, ALCwinmmPlayback_mixerProc, self) != althrd_success)
-        return ALC_FALSE;
-
-    InitRef(&self->WaveBuffersCommitted, 0);
-
-    // Create 4 Buffers
-    BufferSize  = device->UpdateSize*device->NumUpdates / 4;
-    BufferSize *= FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
-
-    BufferData = calloc(4, BufferSize);
-    for(i = 0;i < 4;i++)
-    {
-        memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR));
-        self->WaveBuffer[i].dwBufferLength = BufferSize;
-        self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData :
-                                      (self->WaveBuffer[i-1].lpData +
-                                       self->WaveBuffer[i-1].dwBufferLength));
-        waveOutPrepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
-        waveOutWrite(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
-        IncrementRef(&self->WaveBuffersCommitted);
-    }
-
-    return ALC_TRUE;
-}
-
-static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self)
-{
-    void *buffer = NULL;
-    int i;
-
-    if(self->killNow)
-        return;
-
-    // Set flag to stop processing headers
-    self->killNow = AL_TRUE;
-    althrd_join(self->thread, &i);
-
-    // Release the wave buffers
-    for(i = 0;i < 4;i++)
-    {
-        waveOutUnprepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
-        if(i == 0) buffer = self->WaveBuffer[i].lpData;
-        self->WaveBuffer[i].lpData = NULL;
-    }
-    free(buffer);
-}
-
-
-
-typedef struct ALCwinmmCapture {
-    DERIVE_FROM_TYPE(ALCbackend);
-
-    RefCount WaveBuffersCommitted;
-    WAVEHDR WaveBuffer[4];
-
-    HWAVEIN InHdl;
-
-    ll_ringbuffer_t *Ring;
-
-    WAVEFORMATEX Format;
-
-    volatile ALboolean killNow;
-    althrd_t thread;
-} ALCwinmmCapture;
-
-static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device);
-static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self);
-
-static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2);
-static int ALCwinmmCapture_captureProc(void *arg);
-
-static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name);
-static void ALCwinmmCapture_close(ALCwinmmCapture *self);
-static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ALCboolean, reset)
-static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self);
-static void ALCwinmmCapture_stop(ALCwinmmCapture *self);
-static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples);
-static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self);
-static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ClockLatency, getClockLatency)
-static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, lock)
-static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, unlock)
-DECLARE_DEFAULT_ALLOCATORS(ALCwinmmCapture)
-
-DEFINE_ALCBACKEND_VTABLE(ALCwinmmCapture);
-
-
-static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device)
-{
-    ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
-    SET_VTABLE2(ALCwinmmCapture, ALCbackend, self);
-
-    InitRef(&self->WaveBuffersCommitted, 0);
-    self->InHdl = NULL;
-
-    self->killNow = AL_TRUE;
-}
-
-static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self)
-{
-    if(self->InHdl)
-        waveInClose(self->InHdl);
-    self->InHdl = 0;
-
-    ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
-}
-
-
-/* ALCwinmmCapture_waveInProc
- *
- * Posts a message to 'ALCwinmmCapture_captureProc' everytime a WaveIn Buffer
- * is completed and returns to the application (with more data).
- */
-static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2))
-{
-    ALCwinmmCapture *self = (ALCwinmmCapture*)instance;
-
-    if(msg != WIM_DATA)
-        return;
-
-    DecrementRef(&self->WaveBuffersCommitted);
-    PostThreadMessage(self->thread, msg, 0, param1);
-}
-
-static int ALCwinmmCapture_captureProc(void *arg)
-{
-    ALCwinmmCapture *self = arg;
-    WAVEHDR *WaveHdr;
-    MSG msg;
-
-    althrd_setname(althrd_current(), RECORD_THREAD_NAME);
-
-    while(GetMessage(&msg, NULL, 0, 0))
-    {
-        if(msg.message != WIM_DATA)
-            continue;
-        /* Don't wait for other buffers to finish before quitting. We're
-         * closing so we don't need them. */
-        if(self->killNow)
-            break;
-
-        WaveHdr = ((WAVEHDR*)msg.lParam);
-        ll_ringbuffer_write(self->Ring, WaveHdr->lpData,
-            WaveHdr->dwBytesRecorded / self->Format.nBlockAlign
-        );
-
-        // Send buffer back to capture more data
-        waveInAddBuffer(self->InHdl, WaveHdr, sizeof(WAVEHDR));
-        IncrementRef(&self->WaveBuffersCommitted);
-    }
-
-    return 0;
-}
-
-
-static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name)
-{
-    ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
-    const al_string *iter;
-    ALbyte *BufferData = NULL;
-    DWORD CapturedDataSize;
-    ALint BufferSize;
-    UINT DeviceID;
-    MMRESULT res;
-    ALuint i;
-
-    if(VECTOR_SIZE(CaptureDevices) == 0)
-        ProbeCaptureDevices();
-
-    // Find the Device ID matching the deviceName if valid
-#define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && (!name || alstr_cmp_cstr(*iter, name) == 0))
-    VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_DEVNAME);
-    if(iter == VECTOR_END(CaptureDevices))
-        return ALC_INVALID_VALUE;
-#undef MATCH_DEVNAME
-
-    DeviceID = (UINT)(iter - VECTOR_BEGIN(CaptureDevices));
-
-    switch(device->FmtChans)
-    {
-        case DevFmtMono:
-        case DevFmtStereo:
-            break;
-
-        case DevFmtQuad:
-        case DevFmtX51:
-        case DevFmtX51Rear:
-        case DevFmtX61:
-        case DevFmtX71:
-        case DevFmtAmbi3D:
-            return ALC_INVALID_ENUM;
-    }
-
-    switch(device->FmtType)
-    {
-        case DevFmtUByte:
-        case DevFmtShort:
-        case DevFmtInt:
-        case DevFmtFloat:
-            break;
-
-        case DevFmtByte:
-        case DevFmtUShort:
-        case DevFmtUInt:
-            return ALC_INVALID_ENUM;
-    }
-
-    memset(&self->Format, 0, sizeof(WAVEFORMATEX));
-    self->Format.wFormatTag = ((device->FmtType == DevFmtFloat) ?
-                               WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM);
-    self->Format.nChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
-    self->Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8;
-    self->Format.nBlockAlign = self->Format.wBitsPerSample *
-                               self->Format.nChannels / 8;
-    self->Format.nSamplesPerSec = device->Frequency;
-    self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec *
-                                   self->Format.nBlockAlign;
-    self->Format.cbSize = 0;
-
-    if((res=waveInOpen(&self->InHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmCapture_waveInProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
-    {
-        ERR("waveInOpen failed: %u\n", res);
-        goto failure;
-    }
-
-    // Allocate circular memory buffer for the captured audio
-    CapturedDataSize = device->UpdateSize*device->NumUpdates;
-
-    // Make sure circular buffer is at least 100ms in size
-    if(CapturedDataSize < (self->Format.nSamplesPerSec / 10))
-        CapturedDataSize = self->Format.nSamplesPerSec / 10;
-
-    self->Ring = ll_ringbuffer_create(CapturedDataSize+1, self->Format.nBlockAlign);
-    if(!self->Ring) goto failure;
-
-    InitRef(&self->WaveBuffersCommitted, 0);
-
-    // Create 4 Buffers of 50ms each
-    BufferSize = self->Format.nAvgBytesPerSec / 20;
-    BufferSize -= (BufferSize % self->Format.nBlockAlign);
-
-    BufferData = calloc(4, BufferSize);
-    if(!BufferData) goto failure;
-
-    for(i = 0;i < 4;i++)
-    {
-        memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR));
-        self->WaveBuffer[i].dwBufferLength = BufferSize;
-        self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData :
-                                      (self->WaveBuffer[i-1].lpData +
-                                       self->WaveBuffer[i-1].dwBufferLength));
-        self->WaveBuffer[i].dwFlags = 0;
-        self->WaveBuffer[i].dwLoops = 0;
-        waveInPrepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
-        waveInAddBuffer(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
-        IncrementRef(&self->WaveBuffersCommitted);
-    }
-
-    self->killNow = AL_FALSE;
-    if(althrd_create(&self->thread, ALCwinmmCapture_captureProc, self) != althrd_success)
-        goto failure;
-
-    alstr_copy(&device->DeviceName, VECTOR_ELEM(CaptureDevices, DeviceID));
-    return ALC_NO_ERROR;
-
-failure:
-    if(BufferData)
-    {
-        for(i = 0;i < 4;i++)
-            waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
-        free(BufferData);
-    }
-
-    ll_ringbuffer_free(self->Ring);
-    self->Ring = NULL;
-
-    if(self->InHdl)
-        waveInClose(self->InHdl);
-    self->InHdl = NULL;
-
-    return ALC_INVALID_VALUE;
-}
-
-static void ALCwinmmCapture_close(ALCwinmmCapture *self)
-{
-    void *buffer = NULL;
-    int i;
-
-    /* Tell the processing thread to quit and wait for it to do so. */
-    self->killNow = AL_TRUE;
-    PostThreadMessage(self->thread, WM_QUIT, 0, 0);
-
-    althrd_join(self->thread, &i);
-
-    /* Make sure capture is stopped and all pending buffers are flushed. */
-    waveInReset(self->InHdl);
-
-    // Release the wave buffers
-    for(i = 0;i < 4;i++)
-    {
-        waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
-        if(i == 0) buffer = self->WaveBuffer[i].lpData;
-        self->WaveBuffer[i].lpData = NULL;
-    }
-    free(buffer);
-
-    ll_ringbuffer_free(self->Ring);
-    self->Ring = NULL;
-
-    // Close the Wave device
-    waveInClose(self->InHdl);
-    self->InHdl = NULL;
-}
-
-static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self)
-{
-    waveInStart(self->InHdl);
-    return ALC_TRUE;
-}
-
-static void ALCwinmmCapture_stop(ALCwinmmCapture *self)
-{
-    waveInStop(self->InHdl);
-}
-
-static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples)
-{
-    ll_ringbuffer_read(self->Ring, buffer, samples);
-    return ALC_NO_ERROR;
-}
-
-static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self)
-{
-    return ll_ringbuffer_read_space(self->Ring);
-}
-
-
-static inline void AppendAllDevicesList2(const al_string *name)
-{
-    if(!alstr_empty(*name))
-        AppendAllDevicesList(alstr_get_cstr(*name));
-}
-static inline void AppendCaptureDeviceList2(const al_string *name)
-{
-    if(!alstr_empty(*name))
-        AppendCaptureDeviceList(alstr_get_cstr(*name));
-}
-
-typedef struct ALCwinmmBackendFactory {
-    DERIVE_FROM_TYPE(ALCbackendFactory);
-} ALCwinmmBackendFactory;
-#define ALCWINMMBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwinmmBackendFactory, ALCbackendFactory) } }
-
-static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory *self);
-static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory *self);
-static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory *self, ALCbackend_Type type);
-static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory *self, enum DevProbe type);
-static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
-
-DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwinmmBackendFactory);
-
-
-static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory* UNUSED(self))
-{
-    VECTOR_INIT(PlaybackDevices);
-    VECTOR_INIT(CaptureDevices);
-
-    return ALC_TRUE;
-}
-
-static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory* UNUSED(self))
-{
-    clear_devlist(&PlaybackDevices);
-    VECTOR_DEINIT(PlaybackDevices);
-
-    clear_devlist(&CaptureDevices);
-    VECTOR_DEINIT(CaptureDevices);
-}
-
-static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory* UNUSED(self), ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback || type == ALCbackend_Capture)
-        return ALC_TRUE;
-    return ALC_FALSE;
-}
-
-static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory* UNUSED(self), enum DevProbe type)
-{
-    switch(type)
-    {
-        case ALL_DEVICE_PROBE:
-            ProbePlaybackDevices();
-            VECTOR_FOR_EACH(const al_string, PlaybackDevices, AppendAllDevicesList2);
-            break;
-
-        case CAPTURE_DEVICE_PROBE:
-            ProbeCaptureDevices();
-            VECTOR_FOR_EACH(const al_string, CaptureDevices, AppendCaptureDeviceList2);
-            break;
-    }
-}
-
-static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
-{
-    if(type == ALCbackend_Playback)
-    {
-        ALCwinmmPlayback *backend;
-        NEW_OBJ(backend, ALCwinmmPlayback)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-    if(type == ALCbackend_Capture)
-    {
-        ALCwinmmCapture *backend;
-        NEW_OBJ(backend, ALCwinmmCapture)(device);
-        if(!backend) return NULL;
-        return STATIC_CAST(ALCbackend, backend);
-    }
-
-    return NULL;
-}
-
-ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void)
-{
-    static ALCwinmmBackendFactory factory = ALCWINMMBACKENDFACTORY_INITIALIZER;
-    return STATIC_CAST(ALCbackendFactory, &factory);
-}

Alc/bformatdec.c

@@ -1,612 +0,0 @@
-
-#include "config.h"
-
-#include "bformatdec.h"
-#include "ambdec.h"
-#include "mixer_defs.h"
-#include "alu.h"
-
-#include "bool.h"
-#include "threads.h"
-#include "almalloc.h"
-
-
-void bandsplit_init(BandSplitter *splitter, ALfloat freq_mult)
-{
-    ALfloat w = freq_mult * F_TAU;
-    ALfloat cw = cosf(w);
-    if(cw > FLT_EPSILON)
-        splitter->coeff = (sinf(w) - 1.0f) / cw;
-    else
-        splitter->coeff = cw * -0.5f;
-
-    splitter->lp_z1 = 0.0f;
-    splitter->lp_z2 = 0.0f;
-    splitter->hp_z1 = 0.0f;
-}
-
-void bandsplit_clear(BandSplitter *splitter)
-{
-    splitter->lp_z1 = 0.0f;
-    splitter->lp_z2 = 0.0f;
-    splitter->hp_z1 = 0.0f;
-}
-
-void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout,
-                       const ALfloat *input, ALsizei count)
-{
-    ALfloat coeff, d, x;
-    ALfloat z1, z2;
-    ALsizei i;
-
-    coeff = splitter->coeff*0.5f + 0.5f;
-    z1 = splitter->lp_z1;
-    z2 = splitter->lp_z2;
-    for(i = 0;i < count;i++)
-    {
-        x = input[i];
-
-        d = (x - z1) * coeff;
-        x = z1 + d;
-        z1 = x + d;
-
-        d = (x - z2) * coeff;
-        x = z2 + d;
-        z2 = x + d;
-
-        lpout[i] = x;
-    }
-    splitter->lp_z1 = z1;
-    splitter->lp_z2 = z2;
-
-    coeff = splitter->coeff;
-    z1 = splitter->hp_z1;
-    for(i = 0;i < count;i++)
-    {
-        x = input[i];
-
-        d = x - coeff*z1;
-        x = z1 + coeff*d;
-        z1 = d;
-
-        hpout[i] = x - lpout[i];
-    }
-    splitter->hp_z1 = z1;
-}
-
-
-void splitterap_init(SplitterAllpass *splitter, ALfloat freq_mult)
-{
-    ALfloat w = freq_mult * F_TAU;
-    ALfloat cw = cosf(w);
-    if(cw > FLT_EPSILON)
-        splitter->coeff = (sinf(w) - 1.0f) / cw;
-    else
-        splitter->coeff = cw * -0.5f;
-
-    splitter->z1 = 0.0f;
-}
-
-void splitterap_clear(SplitterAllpass *splitter)
-{
-    splitter->z1 = 0.0f;
-}
-
-void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count)
-{
-    ALfloat coeff, d, x;
-    ALfloat z1;
-    ALsizei i;
-
-    coeff = splitter->coeff;
-    z1 = splitter->z1;
-    for(i = 0;i < count;i++)
-    {
-        x = samples[i];
-
-        d = x - coeff*z1;
-        x = z1 + coeff*d;
-        z1 = d;
-
-        samples[i] = x;
-    }
-    splitter->z1 = z1;
-}
-
-
-static const ALfloat UnitScale[MAX_AMBI_COEFFS] = {
-    1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
-    1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f
-};
-static const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS] = {
-    1.000000000f, /* ACN  0 (W), sqrt(1) */
-    1.732050808f, /* ACN  1 (Y), sqrt(3) */
-    1.732050808f, /* ACN  2 (Z), sqrt(3) */
-    1.732050808f, /* ACN  3 (X), sqrt(3) */
-    2.236067978f, /* ACN  4 (V), sqrt(5) */
-    2.236067978f, /* ACN  5 (T), sqrt(5) */
-    2.236067978f, /* ACN  6 (R), sqrt(5) */
-    2.236067978f, /* ACN  7 (S), sqrt(5) */
-    2.236067978f, /* ACN  8 (U), sqrt(5) */
-    2.645751311f, /* ACN  9 (Q), sqrt(7) */
-    2.645751311f, /* ACN 10 (O), sqrt(7) */
-    2.645751311f, /* ACN 11 (M), sqrt(7) */
-    2.645751311f, /* ACN 12 (K), sqrt(7) */
-    2.645751311f, /* ACN 13 (L), sqrt(7) */
-    2.645751311f, /* ACN 14 (N), sqrt(7) */
-    2.645751311f, /* ACN 15 (P), sqrt(7) */
-};
-static const ALfloat FuMa2N3DScale[MAX_AMBI_COEFFS] = {
-    1.414213562f, /* ACN  0 (W), sqrt(2) */
-    1.732050808f, /* ACN  1 (Y), sqrt(3) */
-    1.732050808f, /* ACN  2 (Z), sqrt(3) */
-    1.732050808f, /* ACN  3 (X), sqrt(3) */
-    1.936491673f, /* ACN  4 (V), sqrt(15)/2 */
-    1.936491673f, /* ACN  5 (T), sqrt(15)/2 */
-    2.236067978f, /* ACN  6 (R), sqrt(5) */
-    1.936491673f, /* ACN  7 (S), sqrt(15)/2 */
-    1.936491673f, /* ACN  8 (U), sqrt(15)/2 */
-    2.091650066f, /* ACN  9 (Q), sqrt(35/8) */
-    1.972026594f, /* ACN 10 (O), sqrt(35)/3 */
-    2.231093404f, /* ACN 11 (M), sqrt(224/45) */
-    2.645751311f, /* ACN 12 (K), sqrt(7) */
-    2.231093404f, /* ACN 13 (L), sqrt(224/45) */
-    1.972026594f, /* ACN 14 (N), sqrt(35)/3 */
-    2.091650066f, /* ACN 15 (P), sqrt(35/8) */
-};
-
-
-enum FreqBand {
-    FB_HighFreq,
-    FB_LowFreq,
-    FB_Max
-};
-
-/* These points are in AL coordinates! */
-static const ALfloat Ambi3DPoints[8][3] = {
-    { -0.577350269f,  0.577350269f, -0.577350269f },
-    {  0.577350269f,  0.577350269f, -0.577350269f },
-    { -0.577350269f,  0.577350269f,  0.577350269f },
-    {  0.577350269f,  0.577350269f,  0.577350269f },
-    { -0.577350269f, -0.577350269f, -0.577350269f },
-    {  0.577350269f, -0.577350269f, -0.577350269f },
-    { -0.577350269f, -0.577350269f,  0.577350269f },
-    {  0.577350269f, -0.577350269f,  0.577350269f },
-};
-static const ALfloat Ambi3DDecoder[8][FB_Max][MAX_AMBI_COEFFS] = {
-    { { 0.25f,  0.1443375672f,  0.1443375672f,  0.1443375672f }, { 0.125f,  0.125f,  0.125f,  0.125f } },
-    { { 0.25f, -0.1443375672f,  0.1443375672f,  0.1443375672f }, { 0.125f, -0.125f,  0.125f,  0.125f } },
-    { { 0.25f,  0.1443375672f,  0.1443375672f, -0.1443375672f }, { 0.125f,  0.125f,  0.125f, -0.125f } },
-    { { 0.25f, -0.1443375672f,  0.1443375672f, -0.1443375672f }, { 0.125f, -0.125f,  0.125f, -0.125f } },
-    { { 0.25f,  0.1443375672f, -0.1443375672f,  0.1443375672f }, { 0.125f,  0.125f, -0.125f,  0.125f } },
-    { { 0.25f, -0.1443375672f, -0.1443375672f,  0.1443375672f }, { 0.125f, -0.125f, -0.125f,  0.125f } },
-    { { 0.25f,  0.1443375672f, -0.1443375672f, -0.1443375672f }, { 0.125f,  0.125f, -0.125f, -0.125f } },
-    { { 0.25f, -0.1443375672f, -0.1443375672f, -0.1443375672f }, { 0.125f, -0.125f, -0.125f, -0.125f } },
-};
-
-
-static RowMixerFunc MixMatrixRow = MixRow_C;
-
-
-static alonce_flag bformatdec_inited = AL_ONCE_FLAG_INIT;
-
-static void init_bformatdec(void)
-{
-    MixMatrixRow = SelectRowMixer();
-}
-
-
-/* NOTE: BandSplitter filters are unused with single-band decoding */
-typedef struct BFormatDec {
-    ALboolean Enabled[MAX_OUTPUT_CHANNELS];
-
-    union {
-        alignas(16) ALfloat Dual[MAX_OUTPUT_CHANNELS][FB_Max][MAX_AMBI_COEFFS];
-        alignas(16) ALfloat Single[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
-    } Matrix;
-
-    BandSplitter XOver[MAX_AMBI_COEFFS];
-
-    ALfloat (*Samples)[BUFFERSIZE];
-    /* These two alias into Samples */
-    ALfloat (*SamplesHF)[BUFFERSIZE];
-    ALfloat (*SamplesLF)[BUFFERSIZE];
-
-    alignas(16) ALfloat ChannelMix[BUFFERSIZE];
-
-    struct {
-        BandSplitter XOver;
-        ALfloat Gains[FB_Max];
-    } UpSampler[4];
-
-    ALsizei NumChannels;
-    ALboolean DualBand;
-} BFormatDec;
-
-BFormatDec *bformatdec_alloc()
-{
-    alcall_once(&bformatdec_inited, init_bformatdec);
-    return al_calloc(16, sizeof(BFormatDec));
-}
-
-void bformatdec_free(BFormatDec *dec)
-{
-    if(dec)
-    {
-        al_free(dec->Samples);
-        dec->Samples = NULL;
-        dec->SamplesHF = NULL;
-        dec->SamplesLF = NULL;
-
-        memset(dec, 0, sizeof(*dec));
-        al_free(dec);
-    }
-}
-
-void bformatdec_reset(BFormatDec *dec, const AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS])
-{
-    static const ALsizei map2DTo3D[MAX_AMBI2D_COEFFS] = {
-        0,  1, 3,  4, 8,  9, 15
-    };
-    const ALfloat *coeff_scale = UnitScale;
-    bool periphonic;
-    ALfloat ratio;
-    ALsizei i;
-
-    al_free(dec->Samples);
-    dec->Samples = NULL;
-    dec->SamplesHF = NULL;
-    dec->SamplesLF = NULL;
-
-    dec->NumChannels = chancount;
-    dec->Samples = al_calloc(16, dec->NumChannels*2 * sizeof(dec->Samples[0]));
-    dec->SamplesHF = dec->Samples;
-    dec->SamplesLF = dec->SamplesHF + dec->NumChannels;
-
-    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-        dec->Enabled[i] = AL_FALSE;
-    for(i = 0;i < conf->NumSpeakers;i++)
-        dec->Enabled[chanmap[i]] = AL_TRUE;
-
-    if(conf->CoeffScale == ADS_SN3D)
-        coeff_scale = SN3D2N3DScale;
-    else if(conf->CoeffScale == ADS_FuMa)
-        coeff_scale = FuMa2N3DScale;
-
-    memset(dec->UpSampler, 0, sizeof(dec->UpSampler));
-    ratio = 400.0f / (ALfloat)srate;
-    for(i = 0;i < 4;i++)
-        bandsplit_init(&dec->UpSampler[i].XOver, ratio);
-    if((conf->ChanMask&AMBI_PERIPHONIC_MASK))
-    {
-        periphonic = true;
-
-        dec->UpSampler[0].Gains[FB_HighFreq] = (dec->NumChannels > 9) ? W_SCALE3D_THIRD :
-                                               (dec->NumChannels > 4) ? W_SCALE3D_SECOND : 1.0f;
-        dec->UpSampler[0].Gains[FB_LowFreq] = 1.0f;
-        for(i = 1;i < 4;i++)
-        {
-            dec->UpSampler[i].Gains[FB_HighFreq] = (dec->NumChannels > 9) ? XYZ_SCALE3D_THIRD :
-                                                   (dec->NumChannels > 4) ? XYZ_SCALE3D_SECOND : 1.0f;
-            dec->UpSampler[i].Gains[FB_LowFreq] = 1.0f;
-        }
-    }
-    else
-    {
-        periphonic = false;
-
-        dec->UpSampler[0].Gains[FB_HighFreq] = (dec->NumChannels > 5) ? W_SCALE2D_THIRD :
-                                               (dec->NumChannels > 3) ? W_SCALE2D_SECOND : 1.0f;
-        dec->UpSampler[0].Gains[FB_LowFreq] = 1.0f;
-        for(i = 1;i < 3;i++)
-        {
-            dec->UpSampler[i].Gains[FB_HighFreq] = (dec->NumChannels > 5) ? XYZ_SCALE2D_THIRD :
-                                                   (dec->NumChannels > 3) ? XYZ_SCALE2D_SECOND : 1.0f;
-            dec->UpSampler[i].Gains[FB_LowFreq] = 1.0f;
-        }
-        dec->UpSampler[3].Gains[FB_HighFreq] = 0.0f;
-        dec->UpSampler[3].Gains[FB_LowFreq] = 0.0f;
-    }
-
-    memset(&dec->Matrix, 0, sizeof(dec->Matrix));
-    if(conf->FreqBands == 1)
-    {
-        dec->DualBand = AL_FALSE;
-        for(i = 0;i < conf->NumSpeakers;i++)
-        {
-            ALsizei chan = chanmap[i];
-            ALfloat gain;
-            ALsizei j, k;
-
-            if(!periphonic)
-            {
-                for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
-                {
-                    ALsizei l = map2DTo3D[j];
-                    if(j == 0) gain = conf->HFOrderGain[0];
-                    else if(j == 1) gain = conf->HFOrderGain[1];
-                    else if(j == 3) gain = conf->HFOrderGain[2];
-                    else if(j == 5) gain = conf->HFOrderGain[3];
-                    if((conf->ChanMask&(1<<l)))
-                        dec->Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[l] *
-                                                      gain;
-                }
-            }
-            else
-            {
-                for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
-                {
-                    if(j == 0) gain = conf->HFOrderGain[0];
-                    else if(j == 1) gain = conf->HFOrderGain[1];
-                    else if(j == 4) gain = conf->HFOrderGain[2];
-                    else if(j == 9) gain = conf->HFOrderGain[3];
-                    if((conf->ChanMask&(1<<j)))
-                        dec->Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[j] *
-                                                      gain;
-                }
-            }
-        }
-    }
-    else
-    {
-        dec->DualBand = AL_TRUE;
-
-        ratio = conf->XOverFreq / (ALfloat)srate;
-        for(i = 0;i < MAX_AMBI_COEFFS;i++)
-            bandsplit_init(&dec->XOver[i], ratio);
-
-        ratio = powf(10.0f, conf->XOverRatio / 40.0f);
-        for(i = 0;i < conf->NumSpeakers;i++)
-        {
-            ALsizei chan = chanmap[i];
-            ALfloat gain;
-            ALsizei j, k;
-
-            if(!periphonic)
-            {
-                for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
-                {
-                    ALsizei l = map2DTo3D[j];
-                    if(j == 0) gain = conf->HFOrderGain[0] * ratio;
-                    else if(j == 1) gain = conf->HFOrderGain[1] * ratio;
-                    else if(j == 3) gain = conf->HFOrderGain[2] * ratio;
-                    else if(j == 5) gain = conf->HFOrderGain[3] * ratio;
-                    if((conf->ChanMask&(1<<l)))
-                        dec->Matrix.Dual[chan][FB_HighFreq][j] = conf->HFMatrix[i][k++] /
-                                                                 coeff_scale[l] * gain;
-                }
-                for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
-                {
-                    ALsizei l = map2DTo3D[j];
-                    if(j == 0) gain = conf->LFOrderGain[0] / ratio;
-                    else if(j == 1) gain = conf->LFOrderGain[1] / ratio;
-                    else if(j == 3) gain = conf->LFOrderGain[2] / ratio;
-                    else if(j == 5) gain = conf->LFOrderGain[3] / ratio;
-                    if((conf->ChanMask&(1<<l)))
-                        dec->Matrix.Dual[chan][FB_LowFreq][j] = conf->LFMatrix[i][k++] /
-                                                                coeff_scale[l] * gain;
-                }
-            }
-            else
-            {
-                for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
-                {
-                    if(j == 0) gain = conf->HFOrderGain[0] * ratio;
-                    else if(j == 1) gain = conf->HFOrderGain[1] * ratio;
-                    else if(j == 4) gain = conf->HFOrderGain[2] * ratio;
-                    else if(j == 9) gain = conf->HFOrderGain[3] * ratio;
-                    if((conf->ChanMask&(1<<j)))
-                        dec->Matrix.Dual[chan][FB_HighFreq][j] = conf->HFMatrix[i][k++] /
-                                                                 coeff_scale[j] * gain;
-                }
-                for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
-                {
-                    if(j == 0) gain = conf->LFOrderGain[0] / ratio;
-                    else if(j == 1) gain = conf->LFOrderGain[1] / ratio;
-                    else if(j == 4) gain = conf->LFOrderGain[2] / ratio;
-                    else if(j == 9) gain = conf->LFOrderGain[3] / ratio;
-                    if((conf->ChanMask&(1<<j)))
-                        dec->Matrix.Dual[chan][FB_LowFreq][j] = conf->LFMatrix[i][k++] /
-                                                                coeff_scale[j] * gain;
-                }
-            }
-        }
-    }
-}
-
-
-void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
-{
-    ALsizei chan, i;
-
-    OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
-    if(dec->DualBand)
-    {
-        for(i = 0;i < dec->NumChannels;i++)
-            bandsplit_process(&dec->XOver[i], dec->SamplesHF[i], dec->SamplesLF[i],
-                              InSamples[i], SamplesToDo);
-
-        for(chan = 0;chan < OutChannels;chan++)
-        {
-            if(!dec->Enabled[chan])
-                continue;
-
-            memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
-            MixMatrixRow(dec->ChannelMix, dec->Matrix.Dual[chan][FB_HighFreq],
-                SAFE_CONST(ALfloatBUFFERSIZE*,dec->SamplesHF), dec->NumChannels, 0,
-                SamplesToDo
-            );
-            MixMatrixRow(dec->ChannelMix, dec->Matrix.Dual[chan][FB_LowFreq],
-                SAFE_CONST(ALfloatBUFFERSIZE*,dec->SamplesLF), dec->NumChannels, 0,
-                SamplesToDo
-            );
-
-            for(i = 0;i < SamplesToDo;i++)
-                OutBuffer[chan][i] += dec->ChannelMix[i];
-        }
-    }
-    else
-    {
-        for(chan = 0;chan < OutChannels;chan++)
-        {
-            if(!dec->Enabled[chan])
-                continue;
-
-            memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
-            MixMatrixRow(dec->ChannelMix, dec->Matrix.Single[chan], InSamples,
-                         dec->NumChannels, 0, SamplesToDo);
-
-            for(i = 0;i < SamplesToDo;i++)
-                OutBuffer[chan][i] += dec->ChannelMix[i];
-        }
-    }
-}
-
-
-void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo)
-{
-    ALsizei i;
-
-    /* This up-sampler leverages the differences observed in dual-band second-
-     * and third-order decoder matrices compared to first-order. For the same
-     * output channel configuration, the low-frequency matrix has identical
-     * coefficients in the shared input channels, while the high-frequency
-     * matrix has extra scalars applied to the W channel and X/Y/Z channels.
-     * Mixing the first-order content into the higher-order stream with the
-     * appropriate counter-scales applied to the HF response results in the
-     * subsequent higher-order decode generating the same response as a first-
-     * order decode.
-     */
-    for(i = 0;i < InChannels;i++)
-    {
-        /* First, split the first-order components into low and high frequency
-         * bands.
-         */
-        bandsplit_process(&dec->UpSampler[i].XOver,
-            dec->Samples[FB_HighFreq], dec->Samples[FB_LowFreq],
-            InSamples[i], SamplesToDo
-        );
-
-        /* Now write each band to the output. */
-        MixMatrixRow(OutBuffer[i], dec->UpSampler[i].Gains,
-            SAFE_CONST(ALfloatBUFFERSIZE*,dec->Samples), FB_Max, 0,
-            SamplesToDo
-        );
-    }
-}
-
-
-#define INVALID_UPSAMPLE_INDEX INT_MAX
-
-static ALsizei GetACNIndex(const BFChannelConfig *chans, ALsizei numchans, ALsizei acn)
-{
-    ALsizei i;
-    for(i = 0;i < numchans;i++)
-    {
-        if(chans[i].Index == acn)
-            return i;
-    }
-    return INVALID_UPSAMPLE_INDEX;
-}
-#define GetChannelForACN(b, a) GetACNIndex((b).Ambi.Map, (b).NumChannels, (a))
-
-typedef struct AmbiUpsampler {
-    alignas(16) ALfloat Samples[FB_Max][BUFFERSIZE];
-
-    BandSplitter XOver[4];
-
-    ALfloat Gains[4][MAX_OUTPUT_CHANNELS][FB_Max];
-} AmbiUpsampler;
-
-AmbiUpsampler *ambiup_alloc()
-{
-    alcall_once(&bformatdec_inited, init_bformatdec);
-    return al_calloc(16, sizeof(AmbiUpsampler));
-}
-
-void ambiup_free(struct AmbiUpsampler *ambiup)
-{
-    al_free(ambiup);
-}
-
-void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device)
-{
-    ALfloat ratio;
-    size_t i;
-
-    ratio = 400.0f / (ALfloat)device->Frequency;
-    for(i = 0;i < 4;i++)
-        bandsplit_init(&ambiup->XOver[i], ratio);
-
-    memset(ambiup->Gains, 0, sizeof(ambiup->Gains));
-    if(device->Dry.CoeffCount > 0)
-    {
-        ALfloat encgains[8][MAX_OUTPUT_CHANNELS];
-        ALsizei j;
-        size_t k;
-
-        for(i = 0;i < COUNTOF(Ambi3DPoints);i++)
-        {
-            ALfloat coeffs[MAX_AMBI_COEFFS] = { 0.0f };
-            CalcDirectionCoeffs(Ambi3DPoints[i], 0.0f, coeffs);
-            ComputePanningGains(device->Dry, coeffs, 1.0f, encgains[i]);
-        }
-
-        /* Combine the matrices that do the in->virt and virt->out conversions
-         * so we get a single in->out conversion. NOTE: the Encoder matrix
-         * (encgains) and output are transposed, so the input channels line up
-         * with the rows and the output channels line up with the columns.
-         */
-        for(i = 0;i < 4;i++)
-        {
-            for(j = 0;j < device->Dry.NumChannels;j++)
-            {
-                ALfloat hfgain=0.0f, lfgain=0.0f;
-                for(k = 0;k < COUNTOF(Ambi3DDecoder);k++)
-                {
-                    hfgain += Ambi3DDecoder[k][FB_HighFreq][i]*encgains[k][j];
-                    lfgain += Ambi3DDecoder[k][FB_LowFreq][i]*encgains[k][j];
-                }
-                ambiup->Gains[i][j][FB_HighFreq] = hfgain;
-                ambiup->Gains[i][j][FB_LowFreq] = lfgain;
-            }
-        }
-    }
-    else
-    {
-        /* Assumes full 3D/periphonic on the input and output mixes! */
-        ALfloat w_scale = (device->Dry.NumChannels > 9) ? W_SCALE3D_THIRD :
-                          (device->Dry.NumChannels > 4) ? W_SCALE3D_SECOND : 1.0f;
-        ALfloat xyz_scale = (device->Dry.NumChannels > 9) ? XYZ_SCALE3D_THIRD :
-                            (device->Dry.NumChannels > 4) ? XYZ_SCALE3D_SECOND : 1.0f;
-        for(i = 0;i < 4;i++)
-        {
-            ALsizei index = GetChannelForACN(device->Dry, i);
-            if(index != INVALID_UPSAMPLE_INDEX)
-            {
-                ALfloat scale = device->Dry.Ambi.Map[index].Scale;
-                ambiup->Gains[i][index][FB_HighFreq] = scale * ((i==0) ? w_scale : xyz_scale);
-                ambiup->Gains[i][index][FB_LowFreq] = scale;
-            }
-        }
-    }
-}
-
-void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
-{
-    ALsizei i, j;
-
-    for(i = 0;i < 4;i++)
-    {
-        bandsplit_process(&ambiup->XOver[i],
-            ambiup->Samples[FB_HighFreq], ambiup->Samples[FB_LowFreq],
-            InSamples[i], SamplesToDo
-        );
-
-        for(j = 0;j < OutChannels;j++)
-            MixMatrixRow(OutBuffer[j], ambiup->Gains[i][j],
-                SAFE_CONST(ALfloatBUFFERSIZE*,ambiup->Samples), FB_Max, 0,
-                SamplesToDo
-            );
-    }
-}

Alc/bformatdec.h

@@ -1,75 +0,0 @@
-#ifndef BFORMATDEC_H
-#define BFORMATDEC_H
-
-#include "alMain.h"
-
-
-/* These are the necessary scales for first-order HF responses to play over
- * higher-order 2D (non-periphonic) decoders.
- */
-#define W_SCALE2D_SECOND   1.224744871f /* sqrt(1.5) */
-#define XYZ_SCALE2D_SECOND 1.0f
-#define W_SCALE2D_THIRD   1.414213562f /* sqrt(2) */
-#define XYZ_SCALE2D_THIRD 1.082392196f
-
-/* These are the necessary scales for first-order HF responses to play over
- * higher-order 3D (periphonic) decoders.
- */
-#define W_SCALE3D_SECOND   1.341640787f /* sqrt(1.8) */
-#define XYZ_SCALE3D_SECOND 1.0f
-#define W_SCALE3D_THIRD   1.695486018f
-#define XYZ_SCALE3D_THIRD 1.136697713f
-
-
-struct AmbDecConf;
-struct BFormatDec;
-struct AmbiUpsampler;
-
-
-struct BFormatDec *bformatdec_alloc();
-void bformatdec_free(struct BFormatDec *dec);
-void bformatdec_reset(struct BFormatDec *dec, const struct AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS]);
-
-/* Decodes the ambisonic input to the given output channels. */
-void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
-
-/* Up-samples a first-order input to the decoder's configuration. */
-void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo);
-
-
-/* Stand-alone first-order upsampler. Kept here because it shares some stuff
- * with bformatdec.
- */
-struct AmbiUpsampler *ambiup_alloc();
-void ambiup_free(struct AmbiUpsampler *ambiup);
-void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device);
-
-void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
-
-
-/* Band splitter. Splits a signal into two phase-matching frequency bands. */
-typedef struct BandSplitter {
-    ALfloat coeff;
-    ALfloat lp_z1;
-    ALfloat lp_z2;
-    ALfloat hp_z1;
-} BandSplitter;
-
-void bandsplit_init(BandSplitter *splitter, ALfloat freq_mult);
-void bandsplit_clear(BandSplitter *splitter);
-void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout,
-                       const ALfloat *input, ALsizei count);
-
-/* The all-pass portion of the band splitter. Applies the same phase shift
- * without splitting the signal.
- */
-typedef struct SplitterAllpass {
-    ALfloat coeff;
-    ALfloat z1;
-} SplitterAllpass;
-
-void splitterap_init(SplitterAllpass *splitter, ALfloat freq_mult);
-void splitterap_clear(SplitterAllpass *splitter);
-void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count);
-
-#endif /* BFORMATDEC_H */

Alc/bs2b.c

@@ -1,187 +0,0 @@
-/*-
- * Copyright (c) 2005 Boris Mikhaylov
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject to
- * the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <string.h>
-
-#include "bs2b.h"
-#include "alu.h"
-
-
-/* Set up all data. */
-static void init(struct bs2b *bs2b)
-{
-    float Fc_lo, Fc_hi;
-    float G_lo,  G_hi;
-    float x, g;
-
-    switch(bs2b->level)
-    {
-    case BS2B_LOW_CLEVEL: /* Low crossfeed level */
-        Fc_lo = 360.0f;
-        Fc_hi = 501.0f;
-        G_lo  = 0.398107170553497f;
-        G_hi  = 0.205671765275719f;
-        break;
-
-    case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */
-        Fc_lo = 500.0f;
-        Fc_hi = 711.0f;
-        G_lo  = 0.459726988530872f;
-        G_hi  = 0.228208484414988f;
-        break;
-
-    case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */
-        Fc_lo = 700.0f;
-        Fc_hi = 1021.0f;
-        G_lo  = 0.530884444230988f;
-        G_hi  = 0.250105790667544f;
-        break;
-
-    case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */
-        Fc_lo = 360.0f;
-        Fc_hi = 494.0f;
-        G_lo  = 0.316227766016838f;
-        G_hi  = 0.168236228897329f;
-        break;
-
-    case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */
-        Fc_lo = 500.0f;
-        Fc_hi = 689.0f;
-        G_lo  = 0.354813389233575f;
-        G_hi  = 0.187169483835901f;
-        break;
-
-    default: /* High easy crossfeed level */
-        bs2b->level = BS2B_HIGH_ECLEVEL;
-
-        Fc_lo = 700.0f;
-        Fc_hi = 975.0f;
-        G_lo  = 0.398107170553497f;
-        G_hi  = 0.205671765275719f;
-        break;
-    } /* switch */
-
-    g = 1.0f / (1.0f - G_hi + G_lo);
-
-    /* $fc = $Fc / $s;
-     * $d  = 1 / 2 / pi / $fc;
-     * $x  = exp(-1 / $d);
-     */
-    x           = expf(-2.0f * F_PI * Fc_lo / bs2b->srate);
-    bs2b->b1_lo = x;
-    bs2b->a0_lo = G_lo * (1.0f - x) * g;
-
-    x           = expf(-2.0f * F_PI * Fc_hi / bs2b->srate);
-    bs2b->b1_hi = x;
-    bs2b->a0_hi = (1.0f - G_hi * (1.0f - x)) * g;
-    bs2b->a1_hi = -x * g;
-} /* init */
-
-
-/* Exported functions.
- * See descriptions in "bs2b.h"
- */
-
-void bs2b_set_params(struct bs2b *bs2b, int level, int srate)
-{
-    if(srate <= 0) srate = 1;
-
-    bs2b->level = level;
-    bs2b->srate = srate;
-    init(bs2b);
-} /* bs2b_set_params */
-
-int bs2b_get_level(struct bs2b *bs2b)
-{
-    return bs2b->level;
-} /* bs2b_get_level */
-
-int bs2b_get_srate(struct bs2b *bs2b)
-{
-    return bs2b->srate;
-} /* bs2b_get_srate */
-
-void bs2b_clear(struct bs2b *bs2b)
-{
-    memset(&bs2b->last_sample, 0, sizeof(bs2b->last_sample));
-} /* bs2b_clear */
-
-void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo)
-{
-    float lsamples[128][2];
-    float rsamples[128][2];
-    int base;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        int todo = mini(128, SamplesToDo-base);
-        int i;
-
-        /* Process left input */
-        lsamples[0][0] = bs2b->a0_lo*Left[0] +
-                         bs2b->b1_lo*bs2b->last_sample[0].lo;
-        lsamples[0][1] = bs2b->a0_hi*Left[0] +
-                         bs2b->a1_hi*bs2b->last_sample[0].asis +
-                         bs2b->b1_hi*bs2b->last_sample[0].hi;
-        for(i = 1;i < todo;i++)
-        {
-            lsamples[i][0] = bs2b->a0_lo*Left[i] +
-                             bs2b->b1_lo*lsamples[i-1][0];
-            lsamples[i][1] = bs2b->a0_hi*Left[i] +
-                             bs2b->a1_hi*Left[i-1] +
-                             bs2b->b1_hi*lsamples[i-1][1];
-        }
-        bs2b->last_sample[0].asis = Left[i-1];
-        bs2b->last_sample[0].lo = lsamples[i-1][0];
-        bs2b->last_sample[0].hi = lsamples[i-1][1];
-
-        /* Process right input */
-        rsamples[0][0] = bs2b->a0_lo*Right[0] +
-                         bs2b->b1_lo*bs2b->last_sample[1].lo;
-        rsamples[0][1] = bs2b->a0_hi*Right[0] +
-                         bs2b->a1_hi*bs2b->last_sample[1].asis +
-                         bs2b->b1_hi*bs2b->last_sample[1].hi;
-        for(i = 1;i < todo;i++)
-        {
-            rsamples[i][0] = bs2b->a0_lo*Right[i] +
-                             bs2b->b1_lo*rsamples[i-1][0];
-            rsamples[i][1] = bs2b->a0_hi*Right[i] +
-                             bs2b->a1_hi*Right[i-1] +
-                             bs2b->b1_hi*rsamples[i-1][1];
-        }
-        bs2b->last_sample[1].asis = Right[i-1];
-        bs2b->last_sample[1].lo = rsamples[i-1][0];
-        bs2b->last_sample[1].hi = rsamples[i-1][1];
-
-        /* Crossfeed */
-        for(i = 0;i < todo;i++)
-            *(Left++) = lsamples[i][1] + rsamples[i][0];
-        for(i = 0;i < todo;i++)
-            *(Right++) = rsamples[i][1] + lsamples[i][0];
-
-        base += todo;
-    }
-} /* bs2b_cross_feed */

Alc/compat.h

@@ -1,57 +0,0 @@
-#ifndef AL_COMPAT_H
-#define AL_COMPAT_H
-
-#include "alstring.h"
-
-#ifdef _WIN32
-
-#define WIN32_LEAN_AND_MEAN
-#include <windows.h>
-
-WCHAR *strdupW(const WCHAR *str);
-
-/* Opens a file with standard I/O. The filename is expected to be UTF-8. */
-FILE *al_fopen(const char *fname, const char *mode);
-
-#define HAVE_DYNLOAD 1
-
-#else
-
-#define al_fopen fopen
-
-#if defined(HAVE_DLFCN_H) && !defined(IN_IDE_PARSER)
-#define HAVE_DYNLOAD 1
-#endif
-
-#endif
-
-struct FileMapping {
-#ifdef _WIN32
-    HANDLE file;
-    HANDLE fmap;
-#else
-    int fd;
-#endif
-    void *ptr;
-    size_t len;
-};
-struct FileMapping MapFileToMem(const char *fname);
-void UnmapFileMem(const struct FileMapping *mapping);
-
-al_string GetProcPath(void);
-
-#ifdef HAVE_DYNLOAD
-void *LoadLib(const char *name);
-void CloseLib(void *handle);
-void *GetSymbol(void *handle, const char *name);
-#endif
-
-#ifdef __ANDROID__
-#define JCALL(obj, func)  ((*(obj))->func((obj), EXTRACT_VCALL_ARGS
-#define JCALL0(obj, func)  ((*(obj))->func((obj) EXTRACT_VCALL_ARGS
-
-/** Returns a JNIEnv*. */
-void *Android_GetJNIEnv(void);
-#endif
-
-#endif /* AL_COMPAT_H */

Alc/converter.c

@@ -1,466 +0,0 @@
-
-#include "config.h"
-
-#include "converter.h"
-
-#include "mixer_defs.h"
-
-
-SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate)
-{
-    SampleConverter *converter;
-    ALsizei step;
-
-    if(numchans <= 0 || srcRate <= 0 || dstRate <= 0)
-        return NULL;
-
-    converter = al_calloc(16, FAM_SIZE(SampleConverter, Chan, numchans));
-    converter->mSrcType = srcType;
-    converter->mDstType = dstType;
-    converter->mNumChannels = numchans;
-    converter->mSrcTypeSize = BytesFromDevFmt(srcType);
-    converter->mDstTypeSize = BytesFromDevFmt(dstType);
-
-    converter->mSrcPrepCount = 0;
-    converter->mFracOffset = 0;
-
-    /* Have to set the mixer FPU mode since that's what the resampler code expects. */
-    START_MIXER_MODE();
-    step = fastf2i(minf((ALdouble)srcRate / dstRate, MAX_PITCH)*FRACTIONONE + 0.5f);
-    converter->mIncrement = maxi(step, 1);
-    if(converter->mIncrement == FRACTIONONE)
-        converter->mResample = Resample_copy32_C;
-    else
-    {
-        /* TODO: Allow other resamplers. */
-        BsincPrepare(converter->mIncrement, &converter->mState.bsinc);
-        converter->mResample = SelectResampler(BSincResampler);
-    }
-    END_MIXER_MODE();
-
-    return converter;
-}
-
-void DestroySampleConverter(SampleConverter **converter)
-{
-    if(converter)
-    {
-        al_free(*converter);
-        *converter = NULL;
-    }
-}
-
-
-static inline ALfloat Sample_ALbyte(ALbyte val)
-{ return val * (1.0f/128.0f); }
-static inline ALfloat Sample_ALubyte(ALubyte val)
-{ return Sample_ALbyte((ALint)val - 128); }
-
-static inline ALfloat Sample_ALshort(ALshort val)
-{ return val * (1.0f/32768.0f); }
-static inline ALfloat Sample_ALushort(ALushort val)
-{ return Sample_ALshort((ALint)val - 32768); }
-
-static inline ALfloat Sample_ALint(ALint val)
-{ return (val>>7) * (1.0f/16777216.0f); }
-static inline ALfloat Sample_ALuint(ALuint val)
-{ return Sample_ALint(val - INT_MAX - 1); }
-
-static inline ALfloat Sample_ALfloat(ALfloat val)
-{ return val; }
-
-#define DECL_TEMPLATE(T)                                                      \
-static inline void Load_##T(ALfloat *restrict dst, const T *restrict src,     \
-                            ALint srcstep, ALsizei samples)                   \
-{                                                                             \
-    ALsizei i;                                                                \
-    for(i = 0;i < samples;i++)                                                \
-        dst[i] = Sample_##T(src[i*srcstep]);                                  \
-}
-
-DECL_TEMPLATE(ALbyte)
-DECL_TEMPLATE(ALubyte)
-DECL_TEMPLATE(ALshort)
-DECL_TEMPLATE(ALushort)
-DECL_TEMPLATE(ALint)
-DECL_TEMPLATE(ALuint)
-DECL_TEMPLATE(ALfloat)
-
-#undef DECL_TEMPLATE
-
-static void LoadSamples(ALfloat *dst, const ALvoid *src, ALint srcstep, enum DevFmtType srctype, ALsizei samples)
-{
-    switch(srctype)
-    {
-        case DevFmtByte:
-            Load_ALbyte(dst, src, srcstep, samples);
-            break;
-        case DevFmtUByte:
-            Load_ALubyte(dst, src, srcstep, samples);
-            break;
-        case DevFmtShort:
-            Load_ALshort(dst, src, srcstep, samples);
-            break;
-        case DevFmtUShort:
-            Load_ALushort(dst, src, srcstep, samples);
-            break;
-        case DevFmtInt:
-            Load_ALint(dst, src, srcstep, samples);
-            break;
-        case DevFmtUInt:
-            Load_ALuint(dst, src, srcstep, samples);
-            break;
-        case DevFmtFloat:
-            Load_ALfloat(dst, src, srcstep, samples);
-            break;
-    }
-}
-
-
-static inline ALbyte ALbyte_Sample(ALfloat val)
-{ return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); }
-static inline ALubyte ALubyte_Sample(ALfloat val)
-{ return ALbyte_Sample(val)+128; }
-
-static inline ALshort ALshort_Sample(ALfloat val)
-{ return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); }
-static inline ALushort ALushort_Sample(ALfloat val)
-{ return ALshort_Sample(val)+32768; }
-
-static inline ALint ALint_Sample(ALfloat val)
-{ return fastf2i(clampf(val*16777216.0f, -16777216.0f, 16777215.0f)) << 7; }
-static inline ALuint ALuint_Sample(ALfloat val)
-{ return ALint_Sample(val)+INT_MAX+1; }
-
-static inline ALfloat ALfloat_Sample(ALfloat val)
-{ return val; }
-
-#define DECL_TEMPLATE(T)                                                      \
-static inline void Store_##T(T *restrict dst, const ALfloat *restrict src,    \
-                             ALint dststep, ALsizei samples)                  \
-{                                                                             \
-    ALsizei i;                                                                \
-    for(i = 0;i < samples;i++)                                                \
-        dst[i*dststep] = T##_Sample(src[i]);                                  \
-}
-
-DECL_TEMPLATE(ALbyte)
-DECL_TEMPLATE(ALubyte)
-DECL_TEMPLATE(ALshort)
-DECL_TEMPLATE(ALushort)
-DECL_TEMPLATE(ALint)
-DECL_TEMPLATE(ALuint)
-DECL_TEMPLATE(ALfloat)
-
-#undef DECL_TEMPLATE
-
-static void StoreSamples(ALvoid *dst, const ALfloat *src, ALint dststep, enum DevFmtType dsttype, ALsizei samples)
-{
-    switch(dsttype)
-    {
-        case DevFmtByte:
-            Store_ALbyte(dst, src, dststep, samples);
-            break;
-        case DevFmtUByte:
-            Store_ALubyte(dst, src, dststep, samples);
-            break;
-        case DevFmtShort:
-            Store_ALshort(dst, src, dststep, samples);
-            break;
-        case DevFmtUShort:
-            Store_ALushort(dst, src, dststep, samples);
-            break;
-        case DevFmtInt:
-            Store_ALint(dst, src, dststep, samples);
-            break;
-        case DevFmtUInt:
-            Store_ALuint(dst, src, dststep, samples);
-            break;
-        case DevFmtFloat:
-            Store_ALfloat(dst, src, dststep, samples);
-            break;
-    }
-}
-
-
-ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes)
-{
-    ALint prepcount = converter->mSrcPrepCount;
-    ALsizei increment = converter->mIncrement;
-    ALsizei DataPosFrac = converter->mFracOffset;
-    ALuint64 DataSize64;
-
-    if(prepcount < 0)
-    {
-        /* Negative prepcount means we need to skip that many input samples. */
-        if(-prepcount >= srcframes)
-            return 0;
-        srcframes += prepcount;
-        prepcount = 0;
-    }
-
-    if(srcframes < 1)
-    {
-        /* No output samples if there's no input samples. */
-        return 0;
-    }
-
-    if(prepcount < MAX_POST_SAMPLES+MAX_PRE_SAMPLES &&
-       MAX_POST_SAMPLES+MAX_PRE_SAMPLES-prepcount >= srcframes)
-    {
-        /* Not enough input samples to generate an output sample. */
-        return 0;
-    }
-
-    DataSize64  = prepcount;
-    DataSize64 += srcframes;
-    DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
-    DataSize64 <<= FRACTIONBITS;
-    DataSize64 -= DataPosFrac;
-
-    /* If we have a full prep, we can generate at least one sample. */
-    return (ALsizei)clampu64((DataSize64 + increment-1)/increment, 1, BUFFERSIZE);
-}
-
-
-ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes)
-{
-    const ALsizei SrcFrameSize = converter->mNumChannels * converter->mSrcTypeSize;
-    const ALsizei DstFrameSize = converter->mNumChannels * converter->mDstTypeSize;
-    const ALsizei increment = converter->mIncrement;
-    ALsizei pos = 0;
-
-    START_MIXER_MODE();
-    while(pos < dstframes && *srcframes > 0)
-    {
-        ALfloat *restrict SrcData = ASSUME_ALIGNED(converter->mSrcSamples, 16);
-        ALfloat *restrict DstData = ASSUME_ALIGNED(converter->mDstSamples, 16);
-        ALint prepcount = converter->mSrcPrepCount;
-        ALsizei DataPosFrac = converter->mFracOffset;
-        ALuint64 DataSize64;
-        ALsizei DstSize;
-        ALint toread;
-        ALsizei chan;
-
-        if(prepcount < 0)
-        {
-            /* Negative prepcount means we need to skip that many input samples. */
-            if(-prepcount >= *srcframes)
-            {
-                converter->mSrcPrepCount = prepcount + *srcframes;
-                *srcframes = 0;
-                break;
-            }
-            *src = (const ALbyte*)*src + SrcFrameSize*-prepcount;
-            *srcframes += prepcount;
-            converter->mSrcPrepCount = 0;
-            continue;
-        }
-        toread = mini(*srcframes, BUFFERSIZE-(MAX_POST_SAMPLES+MAX_PRE_SAMPLES));
-
-        if(prepcount < MAX_POST_SAMPLES+MAX_PRE_SAMPLES &&
-           MAX_POST_SAMPLES+MAX_PRE_SAMPLES-prepcount >= toread)
-        {
-            /* Not enough input samples to generate an output sample. Store
-             * what we're given for later.
-             */
-            for(chan = 0;chan < converter->mNumChannels;chan++)
-                LoadSamples(&converter->Chan[chan].mPrevSamples[prepcount],
-                    (const ALbyte*)*src + converter->mSrcTypeSize*chan,
-                    converter->mNumChannels, converter->mSrcType, toread
-                );
-
-            converter->mSrcPrepCount = prepcount + toread;
-            *srcframes = 0;
-            break;
-        }
-
-        DataSize64  = prepcount;
-        DataSize64 += toread;
-        DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
-        DataSize64 <<= FRACTIONBITS;
-        DataSize64 -= DataPosFrac;
-
-        /* If we have a full prep, we can generate at least one sample. */
-        DstSize = (ALsizei)clampu64((DataSize64 + increment-1)/increment, 1, BUFFERSIZE);
-        DstSize = mini(DstSize, dstframes-pos);
-
-        for(chan = 0;chan < converter->mNumChannels;chan++)
-        {
-            const ALbyte *SrcSamples = (const ALbyte*)*src + converter->mSrcTypeSize*chan;
-            ALbyte *DstSamples = (ALbyte*)dst + converter->mDstTypeSize*chan;
-            const ALfloat *ResampledData;
-            ALsizei SrcDataEnd;
-
-            /* Load the previous samples into the source data first, then the
-             * new samples from the input buffer.
-             */
-            memcpy(SrcData, converter->Chan[chan].mPrevSamples,
-                   prepcount*sizeof(ALfloat));
-            LoadSamples(SrcData + prepcount, SrcSamples,
-                converter->mNumChannels, converter->mSrcType, toread
-            );
-
-            /* Store as many prep samples for next time as possible, given the
-             * number of output samples being generated.
-             */
-            SrcDataEnd = (DataPosFrac + increment*DstSize)>>FRACTIONBITS;
-            if(SrcDataEnd >= prepcount+toread)
-                memset(converter->Chan[chan].mPrevSamples, 0,
-                       sizeof(converter->Chan[chan].mPrevSamples));
-            else
-            {
-                size_t len = mini(MAX_PRE_SAMPLES+MAX_POST_SAMPLES, prepcount+toread-SrcDataEnd);
-                memcpy(converter->Chan[chan].mPrevSamples, &SrcData[SrcDataEnd],
-                       len*sizeof(ALfloat));
-                memset(converter->Chan[chan].mPrevSamples+len, 0,
-                       sizeof(converter->Chan[chan].mPrevSamples) - len*sizeof(ALfloat));
-            }
-
-            /* Now resample, and store the result in the output buffer. */
-            ResampledData = converter->mResample(&converter->mState,
-                SrcData+MAX_PRE_SAMPLES, DataPosFrac, increment,
-                DstData, DstSize
-            );
-
-            StoreSamples(DstSamples, ResampledData, converter->mNumChannels,
-                         converter->mDstType, DstSize);
-        }
-
-        /* Update the number of prep samples still available, as well as the
-         * fractional offset.
-         */
-        DataPosFrac += increment*DstSize;
-        converter->mSrcPrepCount = mini(MAX_PRE_SAMPLES+MAX_POST_SAMPLES,
-                                        prepcount+toread-(DataPosFrac>>FRACTIONBITS));
-        converter->mFracOffset = DataPosFrac & FRACTIONMASK;
-
-        /* Update the src and dst pointers in case there's still more to do. */
-        *src = (const ALbyte*)*src + SrcFrameSize*(DataPosFrac>>FRACTIONBITS);
-        *srcframes -= mini(*srcframes, (DataPosFrac>>FRACTIONBITS));
-
-        dst = (ALbyte*)dst + DstFrameSize*DstSize;
-        pos += DstSize;
-    }
-    END_MIXER_MODE();
-
-    return pos;
-}
-
-
-ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans)
-{
-    ChannelConverter *converter;
-
-    if(srcChans != dstChans && !((srcChans == DevFmtMono && dstChans == DevFmtStereo) ||
-                                 (srcChans == DevFmtStereo && dstChans == DevFmtMono)))
-        return NULL;
-
-    converter = al_calloc(DEF_ALIGN, sizeof(*converter));
-    converter->mSrcType = srcType;
-    converter->mSrcChans = srcChans;
-    converter->mDstChans = dstChans;
-
-    return converter;
-}
-
-void DestroyChannelConverter(ChannelConverter **converter)
-{
-    if(converter)
-    {
-        al_free(*converter);
-        *converter = NULL;
-    }
-}
-
-
-#define DECL_TEMPLATE(T)                                                       \
-static void Mono2Stereo##T(ALfloat *restrict dst, const T *src, ALsizei frames)\
-{                                                                              \
-    ALsizei i;                                                                 \
-    for(i = 0;i < frames;i++)                                                  \
-        dst[i*2 + 1] = dst[i*2 + 0] = Sample_##T(src[i]) * 0.707106781187f;    \
-}                                                                              \
-                                                                               \
-static void Stereo2Mono##T(ALfloat *restrict dst, const T *src, ALsizei frames)\
-{                                                                              \
-    ALsizei i;                                                                 \
-    for(i = 0;i < frames;i++)                                                  \
-        dst[i] = (Sample_##T(src[i*2 + 0])+Sample_##T(src[i*2 + 1])) *         \
-                 0.707106781187f;                                              \
-}
-
-DECL_TEMPLATE(ALbyte)
-DECL_TEMPLATE(ALubyte)
-DECL_TEMPLATE(ALshort)
-DECL_TEMPLATE(ALushort)
-DECL_TEMPLATE(ALint)
-DECL_TEMPLATE(ALuint)
-DECL_TEMPLATE(ALfloat)
-
-#undef DECL_TEMPLATE
-
-void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames)
-{
-    if(converter->mSrcChans == converter->mDstChans)
-    {
-        LoadSamples(dst, src, 1, converter->mSrcType,
-                    frames*ChannelsFromDevFmt(converter->mSrcChans, 0));
-        return;
-    }
-
-    if(converter->mSrcChans == DevFmtStereo && converter->mDstChans == DevFmtMono)
-    {
-        switch(converter->mSrcType)
-        {
-            case DevFmtByte:
-                Stereo2MonoALbyte(dst, src, frames);
-                break;
-            case DevFmtUByte:
-                Stereo2MonoALubyte(dst, src, frames);
-                break;
-            case DevFmtShort:
-                Stereo2MonoALshort(dst, src, frames);
-                break;
-            case DevFmtUShort:
-                Stereo2MonoALushort(dst, src, frames);
-                break;
-            case DevFmtInt:
-                Stereo2MonoALint(dst, src, frames);
-                break;
-            case DevFmtUInt:
-                Stereo2MonoALuint(dst, src, frames);
-                break;
-            case DevFmtFloat:
-                Stereo2MonoALfloat(dst, src, frames);
-                break;
-        }
-    }
-    else /*if(converter->mSrcChans == DevFmtMono && converter->mDstChans == DevFmtStereo)*/
-    {
-        switch(converter->mSrcType)
-        {
-            case DevFmtByte:
-                Mono2StereoALbyte(dst, src, frames);
-                break;
-            case DevFmtUByte:
-                Mono2StereoALubyte(dst, src, frames);
-                break;
-            case DevFmtShort:
-                Mono2StereoALshort(dst, src, frames);
-                break;
-            case DevFmtUShort:
-                Mono2StereoALushort(dst, src, frames);
-                break;
-            case DevFmtInt:
-                Mono2StereoALint(dst, src, frames);
-                break;
-            case DevFmtUInt:
-                Mono2StereoALuint(dst, src, frames);
-                break;
-            case DevFmtFloat:
-                Mono2StereoALfloat(dst, src, frames);
-                break;
-        }
-    }
-}

Alc/converter.h

@@ -1,55 +0,0 @@
-#ifndef CONVERTER_H
-#define CONVERTER_H
-
-#include "alMain.h"
-#include "alu.h"
-
-#ifdef __cpluspluc
-extern "C" {
-#endif
-
-typedef struct SampleConverter {
-    enum DevFmtType mSrcType;
-    enum DevFmtType mDstType;
-    ALsizei mNumChannels;
-    ALsizei mSrcTypeSize;
-    ALsizei mDstTypeSize;
-
-    ALint mSrcPrepCount;
-
-    ALsizei mFracOffset;
-    ALsizei mIncrement;
-    InterpState mState;
-    ResamplerFunc mResample;
-
-    alignas(16) ALfloat mSrcSamples[BUFFERSIZE];
-    alignas(16) ALfloat mDstSamples[BUFFERSIZE];
-
-    struct {
-        alignas(16) ALfloat mPrevSamples[MAX_PRE_SAMPLES+MAX_POST_SAMPLES];
-    } Chan[];
-} SampleConverter;
-
-SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate);
-void DestroySampleConverter(SampleConverter **converter);
-
-ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes);
-ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes);
-
-
-typedef struct ChannelConverter {
-    enum DevFmtType mSrcType;
-    enum DevFmtChannels mSrcChans;
-    enum DevFmtChannels mDstChans;
-} ChannelConverter;
-
-ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans);
-void DestroyChannelConverter(ChannelConverter **converter);
-
-void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames);
-
-#ifdef __cpluspluc
-}
-#endif
-
-#endif /* CONVERTER_H */

Alc/effects/chorus.c

@@ -1,410 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2013 by Mike Gorchak
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-enum ChorusWaveForm {
-    CWF_Triangle = AL_CHORUS_WAVEFORM_TRIANGLE,
-    CWF_Sinusoid = AL_CHORUS_WAVEFORM_SINUSOID
-};
-
-typedef struct ALchorusState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    ALfloat *SampleBuffer[2];
-    ALsizei BufferLength;
-    ALsizei offset;
-    ALsizei lfo_range;
-    ALfloat lfo_scale;
-    ALint lfo_disp;
-
-    /* Gains for left and right sides */
-    ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
-
-    /* effect parameters */
-    enum ChorusWaveForm waveform;
-    ALint delay;
-    ALfloat depth;
-    ALfloat feedback;
-} ALchorusState;
-
-static ALvoid ALchorusState_Destruct(ALchorusState *state);
-static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device);
-static ALvoid ALchorusState_update(ALchorusState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
-static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALchorusState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALchorusState);
-
-
-static void ALchorusState_Construct(ALchorusState *state)
-{
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALchorusState, ALeffectState, state);
-
-    state->BufferLength = 0;
-    state->SampleBuffer[0] = NULL;
-    state->SampleBuffer[1] = NULL;
-    state->offset = 0;
-    state->lfo_range = 1;
-    state->waveform = CWF_Triangle;
-}
-
-static ALvoid ALchorusState_Destruct(ALchorusState *state)
-{
-    al_free(state->SampleBuffer[0]);
-    state->SampleBuffer[0] = NULL;
-    state->SampleBuffer[1] = NULL;
-
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device)
-{
-    ALsizei maxlen;
-    ALsizei it;
-
-    maxlen = fastf2i(AL_CHORUS_MAX_DELAY * 2.0f * Device->Frequency) + 1;
-    maxlen = NextPowerOf2(maxlen);
-
-    if(maxlen != state->BufferLength)
-    {
-        void *temp = al_calloc(16, maxlen * sizeof(ALfloat) * 2);
-        if(!temp) return AL_FALSE;
-
-        al_free(state->SampleBuffer[0]);
-        state->SampleBuffer[0] = temp;
-        state->SampleBuffer[1] = state->SampleBuffer[0] + maxlen;
-
-        state->BufferLength = maxlen;
-    }
-
-    for(it = 0;it < state->BufferLength;it++)
-    {
-        state->SampleBuffer[0][it] = 0.0f;
-        state->SampleBuffer[1][it] = 0.0f;
-    }
-
-    return AL_TRUE;
-}
-
-static ALvoid ALchorusState_update(ALchorusState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
-{
-    ALfloat frequency = (ALfloat)Device->Frequency;
-    ALfloat coeffs[MAX_AMBI_COEFFS];
-    ALfloat rate;
-    ALint phase;
-
-    switch(props->Chorus.Waveform)
-    {
-        case AL_CHORUS_WAVEFORM_TRIANGLE:
-            state->waveform = CWF_Triangle;
-            break;
-        case AL_CHORUS_WAVEFORM_SINUSOID:
-            state->waveform = CWF_Sinusoid;
-            break;
-    }
-    state->feedback = props->Chorus.Feedback;
-    state->delay = fastf2i(props->Chorus.Delay * frequency);
-    /* The LFO depth is scaled to be relative to the sample delay. */
-    state->depth = props->Chorus.Depth * state->delay;
-
-    /* Gains for left and right sides */
-    CalcAngleCoeffs(-F_PI_2, 0.0f, 0.0f, coeffs);
-    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[0]);
-    CalcAngleCoeffs( F_PI_2, 0.0f, 0.0f, coeffs);
-    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[1]);
-
-    phase = props->Chorus.Phase;
-    rate = props->Chorus.Rate;
-    if(!(rate > 0.0f))
-    {
-        state->lfo_scale = 0.0f;
-        state->lfo_range = 1;
-        state->lfo_disp = 0;
-    }
-    else
-    {
-        /* Calculate LFO coefficient */
-        state->lfo_range = fastf2i(frequency/rate + 0.5f);
-        switch(state->waveform)
-        {
-            case CWF_Triangle:
-                state->lfo_scale = 4.0f / state->lfo_range;
-                break;
-            case CWF_Sinusoid:
-                state->lfo_scale = F_TAU / state->lfo_range;
-                break;
-        }
-
-        /* Calculate lfo phase displacement */
-        if(phase >= 0)
-            state->lfo_disp = fastf2i(state->lfo_range * (phase/360.0f));
-        else
-            state->lfo_disp = fastf2i(state->lfo_range * ((360+phase)/360.0f));
-    }
-}
-
-static void GetTriangleDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
-                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
-                              const ALsizei todo)
-{
-    ALsizei i;
-    for(i = 0;i < todo;i++)
-    {
-        delays[i] = fastf2i((1.0f - fabsf(2.0f - lfo_scale*offset)) * depth) + delay;
-        offset = (offset+1)%lfo_range;
-    }
-}
-
-static void GetSinusoidDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
-                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
-                              const ALsizei todo)
-{
-    ALsizei i;
-    for(i = 0;i < todo;i++)
-    {
-        delays[i] = fastf2i(sinf(lfo_scale*offset) * depth) + delay;
-        offset = (offset+1)%lfo_range;
-    }
-}
-
-
-static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    ALfloat *restrict leftbuf = state->SampleBuffer[0];
-    ALfloat *restrict rightbuf = state->SampleBuffer[1];
-    const ALsizei bufmask = state->BufferLength-1;
-    const ALfloat feedback = state->feedback;
-    ALsizei offset = state->offset;
-    ALsizei i, c;
-    ALsizei base;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        const ALsizei todo = mini(128, SamplesToDo-base);
-        ALfloat temps[128][2];
-        ALint moddelays[2][128];
-
-        switch(state->waveform)
-        {
-            case CWF_Triangle:
-                GetTriangleDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
-                                  state->lfo_scale, state->depth, state->delay, todo);
-                GetTriangleDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
-                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
-                                  todo);
-                break;
-            case CWF_Sinusoid:
-                GetSinusoidDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
-                                  state->lfo_scale, state->depth, state->delay, todo);
-                GetSinusoidDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
-                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
-                                  todo);
-                break;
-        }
-
-        for(i = 0;i < todo;i++)
-        {
-            leftbuf[offset&bufmask] = SamplesIn[0][base+i];
-            temps[i][0] = leftbuf[(offset-moddelays[0][i])&bufmask] * feedback;
-            leftbuf[offset&bufmask] += temps[i][0];
-
-            rightbuf[offset&bufmask] = SamplesIn[0][base+i];
-            temps[i][1] = rightbuf[(offset-moddelays[1][i])&bufmask] * feedback;
-            rightbuf[offset&bufmask] += temps[i][1];
-
-            offset++;
-        }
-
-        for(c = 0;c < NumChannels;c++)
-        {
-            ALfloat gain = state->Gain[0][c];
-            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
-            {
-                for(i = 0;i < todo;i++)
-                    SamplesOut[c][i+base] += temps[i][0] * gain;
-            }
-
-            gain = state->Gain[1][c];
-            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
-            {
-                for(i = 0;i < todo;i++)
-                    SamplesOut[c][i+base] += temps[i][1] * gain;
-            }
-        }
-
-        base += todo;
-    }
-
-    state->offset = offset;
-}
-
-
-typedef struct ALchorusStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALchorusStateFactory;
-
-static ALeffectState *ALchorusStateFactory_create(ALchorusStateFactory *UNUSED(factory))
-{
-    ALchorusState *state;
-
-    NEW_OBJ0(state, ALchorusState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALchorusStateFactory);
-
-
-ALeffectStateFactory *ALchorusStateFactory_getFactory(void)
-{
-    static ALchorusStateFactory ChorusFactory = { { GET_VTABLE2(ALchorusStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &ChorusFactory);
-}
-
-
-void ALchorus_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_CHORUS_WAVEFORM:
-            if(!(val >= AL_CHORUS_MIN_WAVEFORM && val <= AL_CHORUS_MAX_WAVEFORM))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Chorus.Waveform = val;
-            break;
-
-        case AL_CHORUS_PHASE:
-            if(!(val >= AL_CHORUS_MIN_PHASE && val <= AL_CHORUS_MAX_PHASE))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Chorus.Phase = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALchorus_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALchorus_setParami(effect, context, param, vals[0]);
-}
-void ALchorus_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_CHORUS_RATE:
-            if(!(val >= AL_CHORUS_MIN_RATE && val <= AL_CHORUS_MAX_RATE))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Chorus.Rate = val;
-            break;
-
-        case AL_CHORUS_DEPTH:
-            if(!(val >= AL_CHORUS_MIN_DEPTH && val <= AL_CHORUS_MAX_DEPTH))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Chorus.Depth = val;
-            break;
-
-        case AL_CHORUS_FEEDBACK:
-            if(!(val >= AL_CHORUS_MIN_FEEDBACK && val <= AL_CHORUS_MAX_FEEDBACK))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Chorus.Feedback = val;
-            break;
-
-        case AL_CHORUS_DELAY:
-            if(!(val >= AL_CHORUS_MIN_DELAY && val <= AL_CHORUS_MAX_DELAY))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Chorus.Delay = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALchorus_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALchorus_setParamf(effect, context, param, vals[0]);
-}
-
-void ALchorus_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_CHORUS_WAVEFORM:
-            *val = props->Chorus.Waveform;
-            break;
-
-        case AL_CHORUS_PHASE:
-            *val = props->Chorus.Phase;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALchorus_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALchorus_getParami(effect, context, param, vals);
-}
-void ALchorus_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_CHORUS_RATE:
-            *val = props->Chorus.Rate;
-            break;
-
-        case AL_CHORUS_DEPTH:
-            *val = props->Chorus.Depth;
-            break;
-
-        case AL_CHORUS_FEEDBACK:
-            *val = props->Chorus.Feedback;
-            break;
-
-        case AL_CHORUS_DELAY:
-            *val = props->Chorus.Delay;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALchorus_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALchorus_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALchorus);

Alc/effects/compressor.c

@@ -1,254 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2013 by Anis A. Hireche
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include <stdlib.h>
-
-#include "config.h"
-#include "alError.h"
-#include "alMain.h"
-#include "alAuxEffectSlot.h"
-#include "alu.h"
-
-
-typedef struct ALcompressorState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    /* Effect gains for each channel */
-    ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
-
-    /* Effect parameters */
-    ALboolean Enabled;
-    ALfloat AttackRate;
-    ALfloat ReleaseRate;
-    ALfloat GainCtrl;
-} ALcompressorState;
-
-static ALvoid ALcompressorState_Destruct(ALcompressorState *state);
-static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device);
-static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props);
-static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALcompressorState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALcompressorState);
-
-
-static void ALcompressorState_Construct(ALcompressorState *state)
-{
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALcompressorState, ALeffectState, state);
-
-    state->Enabled = AL_TRUE;
-    state->AttackRate = 0.0f;
-    state->ReleaseRate = 0.0f;
-    state->GainCtrl = 1.0f;
-}
-
-static ALvoid ALcompressorState_Destruct(ALcompressorState *state)
-{
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device)
-{
-    const ALfloat attackTime = device->Frequency * 0.2f; /* 200ms Attack */
-    const ALfloat releaseTime = device->Frequency * 0.4f; /* 400ms Release */
-
-    state->AttackRate = 1.0f / attackTime;
-    state->ReleaseRate = 1.0f / releaseTime;
-
-    return AL_TRUE;
-}
-
-static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props)
-{
-    ALuint i;
-
-    state->Enabled = props->Compressor.OnOff;
-
-    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
-    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
-    for(i = 0;i < 4;i++)
-        ComputeFirstOrderGains(device->FOAOut, IdentityMatrixf.m[i],
-                               slot->Params.Gain, state->Gain[i]);
-}
-
-static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    ALsizei i, j, k;
-    ALsizei base;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        ALfloat temps[64][4];
-        ALsizei td = mini(64, SamplesToDo-base);
-
-        /* Load samples into the temp buffer first. */
-        for(j = 0;j < 4;j++)
-        {
-            for(i = 0;i < td;i++)
-                temps[i][j] = SamplesIn[j][i+base];
-        }
-
-        if(state->Enabled)
-        {
-            ALfloat gain = state->GainCtrl;
-            ALfloat output, amplitude;
-
-            for(i = 0;i < td;i++)
-            {
-                /* Roughly calculate the maximum amplitude from the 4-channel
-                 * signal, and attack or release the gain control to reach it.
-                 */
-                amplitude = fabsf(temps[i][0]);
-                amplitude = maxf(amplitude + fabsf(temps[i][1]),
-                                 maxf(amplitude + fabsf(temps[i][2]),
-                                      amplitude + fabsf(temps[i][3])));
-                if(amplitude > gain)
-                    gain = minf(gain+state->AttackRate, amplitude);
-                else if(amplitude < gain)
-                    gain = maxf(gain-state->ReleaseRate, amplitude);
-
-                /* Apply the inverse of the gain control to normalize/compress
-                 * the volume. */
-                output = 1.0f / clampf(gain, 0.5f, 2.0f);
-                for(j = 0;j < 4;j++)
-                    temps[i][j] *= output;
-            }
-
-            state->GainCtrl = gain;
-        }
-        else
-        {
-            ALfloat gain = state->GainCtrl;
-            ALfloat output, amplitude;
-
-            for(i = 0;i < td;i++)
-            {
-                /* Same as above, except the amplitude is forced to 1. This
-                 * helps ensure smooth gain changes when the compressor is
-                 * turned on and off.
-                 */
-                amplitude = 1.0f;
-                if(amplitude > gain)
-                    gain = minf(gain+state->AttackRate, amplitude);
-                else if(amplitude < gain)
-                    gain = maxf(gain-state->ReleaseRate, amplitude);
-
-                output = 1.0f / clampf(gain, 0.5f, 2.0f);
-                for(j = 0;j < 4;j++)
-                    temps[i][j] *= output;
-            }
-
-            state->GainCtrl = gain;
-        }
-
-        /* Now mix to the output. */
-        for(j = 0;j < 4;j++)
-        {
-            for(k = 0;k < NumChannels;k++)
-            {
-                ALfloat gain = state->Gain[j][k];
-                if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-                    continue;
-
-                for(i = 0;i < td;i++)
-                    SamplesOut[k][base+i] += gain * temps[i][j];
-            }
-        }
-
-        base += td;
-    }
-}
-
-
-typedef struct ALcompressorStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALcompressorStateFactory;
-
-static ALeffectState *ALcompressorStateFactory_create(ALcompressorStateFactory *UNUSED(factory))
-{
-    ALcompressorState *state;
-
-    NEW_OBJ0(state, ALcompressorState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALcompressorStateFactory);
-
-ALeffectStateFactory *ALcompressorStateFactory_getFactory(void)
-{
-    static ALcompressorStateFactory CompressorFactory = { { GET_VTABLE2(ALcompressorStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &CompressorFactory);
-}
-
-
-void ALcompressor_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_COMPRESSOR_ONOFF:
-            if(!(val >= AL_COMPRESSOR_MIN_ONOFF && val <= AL_COMPRESSOR_MAX_ONOFF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Compressor.OnOff = val;
-            break;
-
-    default:
-        SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALcompressor_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALcompressor_setParami(effect, context, param, vals[0]);
-}
-void ALcompressor_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALfloat UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALcompressor_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALcompressor_setParamf(effect, context, param, vals[0]);
-}
-
-void ALcompressor_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
-{ 
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_COMPRESSOR_ONOFF:
-            *val = props->Compressor.OnOff;
-            break;
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALcompressor_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALcompressor_getParami(effect, context, param, vals);
-}
-void ALcompressor_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALfloat *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALcompressor_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALcompressor_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALcompressor);

Alc/effects/dedicated.c

@@ -1,203 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2011 by Chris Robinson.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALdedicatedState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    ALfloat gains[MAX_OUTPUT_CHANNELS];
-} ALdedicatedState;
-
-static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state);
-static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *state, ALCdevice *device);
-static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCdevice *device, const ALeffectslot *Slot, const ALeffectProps *props);
-static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALdedicatedState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALdedicatedState);
-
-
-static void ALdedicatedState_Construct(ALdedicatedState *state)
-{
-    ALsizei s;
-
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALdedicatedState, ALeffectState, state);
-
-    for(s = 0;s < MAX_OUTPUT_CHANNELS;s++)
-        state->gains[s] = 0.0f;
-}
-
-static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state)
-{
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *UNUSED(state), ALCdevice *UNUSED(device))
-{
-    return AL_TRUE;
-}
-
-static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCdevice *device, const ALeffectslot *Slot, const ALeffectProps *props)
-{
-    ALfloat Gain;
-    ALuint i;
-
-    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
-        state->gains[i] = 0.0f;
-
-    Gain = Slot->Params.Gain * props->Dedicated.Gain;
-    if(Slot->Params.EffectType == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
-    {
-        int idx;
-        if((idx=GetChannelIdxByName(device->RealOut, LFE)) != -1)
-        {
-            STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer;
-            STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels;
-            state->gains[idx] = Gain;
-        }
-    }
-    else if(Slot->Params.EffectType == AL_EFFECT_DEDICATED_DIALOGUE)
-    {
-        int idx;
-        /* Dialog goes to the front-center speaker if it exists, otherwise it
-         * plays from the front-center location. */
-        if((idx=GetChannelIdxByName(device->RealOut, FrontCenter)) != -1)
-        {
-            STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer;
-            STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels;
-            state->gains[idx] = Gain;
-        }
-        else
-        {
-            ALfloat coeffs[MAX_AMBI_COEFFS];
-            CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
-
-            STATIC_CAST(ALeffectState,state)->OutBuffer = device->Dry.Buffer;
-            STATIC_CAST(ALeffectState,state)->OutChannels = device->Dry.NumChannels;
-            ComputePanningGains(device->Dry, coeffs, Gain, state->gains);
-        }
-    }
-}
-
-static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    ALsizei i, c;
-
-    SamplesIn = ASSUME_ALIGNED(SamplesIn, 16);
-    SamplesOut = ASSUME_ALIGNED(SamplesOut, 16);
-    for(c = 0;c < NumChannels;c++)
-    {
-        const ALfloat gain = state->gains[c];
-        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-            continue;
-
-        for(i = 0;i < SamplesToDo;i++)
-            SamplesOut[c][i] += SamplesIn[0][i] * gain;
-    }
-}
-
-
-typedef struct ALdedicatedStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALdedicatedStateFactory;
-
-ALeffectState *ALdedicatedStateFactory_create(ALdedicatedStateFactory *UNUSED(factory))
-{
-    ALdedicatedState *state;
-
-    NEW_OBJ0(state, ALdedicatedState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALdedicatedStateFactory);
-
-
-ALeffectStateFactory *ALdedicatedStateFactory_getFactory(void)
-{
-    static ALdedicatedStateFactory DedicatedFactory = { { GET_VTABLE2(ALdedicatedStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &DedicatedFactory);
-}
-
-
-void ALdedicated_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALdedicated_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALdedicated_setParami(effect, context, param, vals[0]);
-}
-void ALdedicated_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_DEDICATED_GAIN:
-            if(!(val >= 0.0f && isfinite(val)))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Dedicated.Gain = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALdedicated_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALdedicated_setParamf(effect, context, param, vals[0]);
-}
-
-void ALdedicated_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALdedicated_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALdedicated_getParami(effect, context, param, vals);
-}
-void ALdedicated_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_DEDICATED_GAIN:
-            *val = props->Dedicated.Gain;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALdedicated_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALdedicated_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALdedicated);

Alc/effects/distortion.c

@@ -1,297 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2013 by Mike Gorchak
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALdistortionState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    /* Effect gains for each channel */
-    ALfloat Gain[MAX_OUTPUT_CHANNELS];
-
-    /* Effect parameters */
-    ALfilterState lowpass;
-    ALfilterState bandpass;
-    ALfloat attenuation;
-    ALfloat edge_coeff;
-} ALdistortionState;
-
-static ALvoid ALdistortionState_Destruct(ALdistortionState *state);
-static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *device);
-static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
-static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALdistortionState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALdistortionState);
-
-
-static void ALdistortionState_Construct(ALdistortionState *state)
-{
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALdistortionState, ALeffectState, state);
-
-    ALfilterState_clear(&state->lowpass);
-    ALfilterState_clear(&state->bandpass);
-}
-
-static ALvoid ALdistortionState_Destruct(ALdistortionState *state)
-{
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *UNUSED(state), ALCdevice *UNUSED(device))
-{
-    return AL_TRUE;
-}
-
-static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
-{
-    ALfloat frequency = (ALfloat)Device->Frequency;
-    ALfloat bandwidth;
-    ALfloat cutoff;
-    ALfloat edge;
-
-    /* Store distorted signal attenuation settings. */
-    state->attenuation = props->Distortion.Gain;
-
-    /* Store waveshaper edge settings. */
-    edge = sinf(props->Distortion.Edge * (F_PI_2));
-    edge = minf(edge, 0.99f);
-    state->edge_coeff = 2.0f * edge / (1.0f-edge);
-
-    cutoff = props->Distortion.LowpassCutoff;
-    /* Bandwidth value is constant in octaves. */
-    bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
-    /* Multiply sampling frequency by the amount of oversampling done during
-     * processing.
-     */
-    ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f,
-        cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
-    );
-
-    cutoff = props->Distortion.EQCenter;
-    /* Convert bandwidth in Hz to octaves. */
-    bandwidth = props->Distortion.EQBandwidth / (cutoff * 0.67f);
-    ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f,
-        cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
-    );
-
-    ComputeAmbientGains(Device->Dry, Slot->Params.Gain, state->Gain);
-}
-
-static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    const ALfloat fc = state->edge_coeff;
-    ALsizei it, kt;
-    ALsizei base;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        float buffer[2][64 * 4];
-        ALsizei td = mini(64, SamplesToDo-base);
-
-        /* Perform 4x oversampling to avoid aliasing. Oversampling greatly
-         * improves distortion quality and allows to implement lowpass and
-         * bandpass filters using high frequencies, at which classic IIR
-         * filters became unstable.
-         */
-
-        /* Fill oversample buffer using zero stuffing. */
-        for(it = 0;it < td;it++)
-        {
-            /* Multiply the sample by the amount of oversampling to maintain
-             * the signal's power.
-             */
-            buffer[0][it*4 + 0] = SamplesIn[0][it+base] * 4.0f;
-            buffer[0][it*4 + 1] = 0.0f;
-            buffer[0][it*4 + 2] = 0.0f;
-            buffer[0][it*4 + 3] = 0.0f;
-        }
-
-        /* First step, do lowpass filtering of original signal. Additionally
-         * perform buffer interpolation and lowpass cutoff for oversampling
-         * (which is fortunately first step of distortion). So combine three
-         * operations into the one.
-         */
-        ALfilterState_process(&state->lowpass, buffer[1], buffer[0], td*4);
-
-        /* Second step, do distortion using waveshaper function to emulate
-         * signal processing during tube overdriving. Three steps of
-         * waveshaping are intended to modify waveform without boost/clipping/
-         * attenuation process.
-         */
-        for(it = 0;it < td*4;it++)
-        {
-            ALfloat smp = buffer[1][it];
-
-            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
-            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f;
-            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
-
-            buffer[0][it] = smp;
-        }
-
-        /* Third step, do bandpass filtering of distorted signal. */
-        ALfilterState_process(&state->bandpass, buffer[1], buffer[0], td*4);
-
-        for(kt = 0;kt < NumChannels;kt++)
-        {
-            /* Fourth step, final, do attenuation and perform decimation,
-             * store only one sample out of 4.
-             */
-            ALfloat gain = state->Gain[kt] * state->attenuation;
-            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-                continue;
-
-            for(it = 0;it < td;it++)
-                SamplesOut[kt][base+it] += gain * buffer[1][it*4];
-        }
-
-        base += td;
-    }
-}
-
-
-typedef struct ALdistortionStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALdistortionStateFactory;
-
-static ALeffectState *ALdistortionStateFactory_create(ALdistortionStateFactory *UNUSED(factory))
-{
-    ALdistortionState *state;
-
-    NEW_OBJ0(state, ALdistortionState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALdistortionStateFactory);
-
-
-ALeffectStateFactory *ALdistortionStateFactory_getFactory(void)
-{
-    static ALdistortionStateFactory DistortionFactory = { { GET_VTABLE2(ALdistortionStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &DistortionFactory);
-}
-
-
-void ALdistortion_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALdistortion_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALdistortion_setParami(effect, context, param, vals[0]);
-}
-void ALdistortion_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_DISTORTION_EDGE:
-            if(!(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Distortion.Edge = val;
-            break;
-
-        case AL_DISTORTION_GAIN:
-            if(!(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Distortion.Gain = val;
-            break;
-
-        case AL_DISTORTION_LOWPASS_CUTOFF:
-            if(!(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Distortion.LowpassCutoff = val;
-            break;
-
-        case AL_DISTORTION_EQCENTER:
-            if(!(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Distortion.EQCenter = val;
-            break;
-
-        case AL_DISTORTION_EQBANDWIDTH:
-            if(!(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Distortion.EQBandwidth = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALdistortion_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALdistortion_setParamf(effect, context, param, vals[0]);
-}
-
-void ALdistortion_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALdistortion_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALdistortion_getParami(effect, context, param, vals);
-}
-void ALdistortion_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_DISTORTION_EDGE:
-            *val = props->Distortion.Edge;
-            break;
-
-        case AL_DISTORTION_GAIN:
-            *val = props->Distortion.Gain;
-            break;
-
-        case AL_DISTORTION_LOWPASS_CUTOFF:
-            *val = props->Distortion.LowpassCutoff;
-            break;
-
-        case AL_DISTORTION_EQCENTER:
-            *val = props->Distortion.EQCenter;
-            break;
-
-        case AL_DISTORTION_EQBANDWIDTH:
-            *val = props->Distortion.EQBandwidth;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALdistortion_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALdistortion_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALdistortion);

Alc/effects/echo.c

@@ -1,326 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2009 by Chris Robinson.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALechoState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    ALfloat *SampleBuffer;
-    ALsizei BufferLength;
-
-    // The echo is two tap. The delay is the number of samples from before the
-    // current offset
-    struct {
-        ALsizei delay;
-    } Tap[2];
-    ALsizei Offset;
-    /* The panning gains for the two taps */
-    ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
-
-    ALfloat FeedGain;
-
-    ALfilterState Filter;
-} ALechoState;
-
-static ALvoid ALechoState_Destruct(ALechoState *state);
-static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device);
-static ALvoid ALechoState_update(ALechoState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
-static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALechoState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALechoState);
-
-
-static void ALechoState_Construct(ALechoState *state)
-{
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALechoState, ALeffectState, state);
-
-    state->BufferLength = 0;
-    state->SampleBuffer = NULL;
-
-    state->Tap[0].delay = 0;
-    state->Tap[1].delay = 0;
-    state->Offset = 0;
-
-    ALfilterState_clear(&state->Filter);
-}
-
-static ALvoid ALechoState_Destruct(ALechoState *state)
-{
-    al_free(state->SampleBuffer);
-    state->SampleBuffer = NULL;
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device)
-{
-    ALsizei maxlen, i;
-
-    // Use the next power of 2 for the buffer length, so the tap offsets can be
-    // wrapped using a mask instead of a modulo
-    maxlen  = fastf2i(AL_ECHO_MAX_DELAY * Device->Frequency) + 1;
-    maxlen += fastf2i(AL_ECHO_MAX_LRDELAY * Device->Frequency) + 1;
-    maxlen  = NextPowerOf2(maxlen);
-
-    if(maxlen != state->BufferLength)
-    {
-        void *temp = al_calloc(16, maxlen * sizeof(ALfloat));
-        if(!temp) return AL_FALSE;
-
-        al_free(state->SampleBuffer);
-        state->SampleBuffer = temp;
-        state->BufferLength = maxlen;
-    }
-    for(i = 0;i < state->BufferLength;i++)
-        state->SampleBuffer[i] = 0.0f;
-
-    return AL_TRUE;
-}
-
-static ALvoid ALechoState_update(ALechoState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
-{
-    ALuint frequency = Device->Frequency;
-    ALfloat coeffs[MAX_AMBI_COEFFS];
-    ALfloat gain, lrpan, spread;
-
-    state->Tap[0].delay = fastf2i(props->Echo.Delay * frequency) + 1;
-    state->Tap[1].delay = fastf2i(props->Echo.LRDelay * frequency);
-    state->Tap[1].delay += state->Tap[0].delay;
-
-    spread = props->Echo.Spread;
-    if(spread < 0.0f) lrpan = -1.0f;
-    else lrpan = 1.0f;
-    /* Convert echo spread (where 0 = omni, +/-1 = directional) to coverage
-     * spread (where 0 = point, tau = omni).
-     */
-    spread = asinf(1.0f - fabsf(spread))*4.0f;
-
-    state->FeedGain = props->Echo.Feedback;
-
-    gain = maxf(1.0f - props->Echo.Damping, 0.0625f); /* Limit -24dB */
-    ALfilterState_setParams(&state->Filter, ALfilterType_HighShelf,
-                            gain, LOWPASSFREQREF/frequency,
-                            calc_rcpQ_from_slope(gain, 1.0f));
-
-    gain = Slot->Params.Gain;
-
-    /* First tap panning */
-    CalcAngleCoeffs(-F_PI_2*lrpan, 0.0f, spread, coeffs);
-    ComputePanningGains(Device->Dry, coeffs, gain, state->Gain[0]);
-
-    /* Second tap panning */
-    CalcAngleCoeffs( F_PI_2*lrpan, 0.0f, spread, coeffs);
-    ComputePanningGains(Device->Dry, coeffs, gain, state->Gain[1]);
-}
-
-static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    const ALsizei mask = state->BufferLength-1;
-    const ALsizei tap1 = state->Tap[0].delay;
-    const ALsizei tap2 = state->Tap[1].delay;
-    ALsizei offset = state->Offset;
-    ALfloat x[2], y[2], in, out;
-    ALsizei base, k;
-    ALsizei i;
-
-    x[0] = state->Filter.x[0];
-    x[1] = state->Filter.x[1];
-    y[0] = state->Filter.y[0];
-    y[1] = state->Filter.y[1];
-    for(base = 0;base < SamplesToDo;)
-    {
-        ALfloat temps[128][2];
-        ALsizei td = mini(128, SamplesToDo-base);
-
-        for(i = 0;i < td;i++)
-        {
-            /* First tap */
-            temps[i][0] = state->SampleBuffer[(offset-tap1) & mask];
-            /* Second tap */
-            temps[i][1] = state->SampleBuffer[(offset-tap2) & mask];
-
-            // Apply damping and feedback gain to the second tap, and mix in the
-            // new sample
-            in = temps[i][1] + SamplesIn[0][i+base];
-            out = in*state->Filter.b0 +
-                  x[0]*state->Filter.b1 + x[1]*state->Filter.b2 -
-                  y[0]*state->Filter.a1 - y[1]*state->Filter.a2;
-            x[1] = x[0]; x[0] = in;
-            y[1] = y[0]; y[0] = out;
-
-            state->SampleBuffer[offset&mask] = out * state->FeedGain;
-            offset++;
-        }
-
-        for(k = 0;k < NumChannels;k++)
-        {
-            ALfloat gain = state->Gain[0][k];
-            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
-            {
-                for(i = 0;i < td;i++)
-                    SamplesOut[k][i+base] += temps[i][0] * gain;
-            }
-
-            gain = state->Gain[1][k];
-            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
-            {
-                for(i = 0;i < td;i++)
-                    SamplesOut[k][i+base] += temps[i][1] * gain;
-            }
-        }
-
-        base += td;
-    }
-    state->Filter.x[0] = x[0];
-    state->Filter.x[1] = x[1];
-    state->Filter.y[0] = y[0];
-    state->Filter.y[1] = y[1];
-
-    state->Offset = offset;
-}
-
-
-typedef struct ALechoStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALechoStateFactory;
-
-ALeffectState *ALechoStateFactory_create(ALechoStateFactory *UNUSED(factory))
-{
-    ALechoState *state;
-
-    NEW_OBJ0(state, ALechoState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALechoStateFactory);
-
-ALeffectStateFactory *ALechoStateFactory_getFactory(void)
-{
-    static ALechoStateFactory EchoFactory = { { GET_VTABLE2(ALechoStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &EchoFactory);
-}
-
-
-void ALecho_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALecho_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALecho_setParami(effect, context, param, vals[0]);
-}
-void ALecho_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_ECHO_DELAY:
-            if(!(val >= AL_ECHO_MIN_DELAY && val <= AL_ECHO_MAX_DELAY))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Echo.Delay = val;
-            break;
-
-        case AL_ECHO_LRDELAY:
-            if(!(val >= AL_ECHO_MIN_LRDELAY && val <= AL_ECHO_MAX_LRDELAY))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Echo.LRDelay = val;
-            break;
-
-        case AL_ECHO_DAMPING:
-            if(!(val >= AL_ECHO_MIN_DAMPING && val <= AL_ECHO_MAX_DAMPING))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Echo.Damping = val;
-            break;
-
-        case AL_ECHO_FEEDBACK:
-            if(!(val >= AL_ECHO_MIN_FEEDBACK && val <= AL_ECHO_MAX_FEEDBACK))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Echo.Feedback = val;
-            break;
-
-        case AL_ECHO_SPREAD:
-            if(!(val >= AL_ECHO_MIN_SPREAD && val <= AL_ECHO_MAX_SPREAD))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Echo.Spread = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALecho_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALecho_setParamf(effect, context, param, vals[0]);
-}
-
-void ALecho_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALecho_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALecho_getParami(effect, context, param, vals);
-}
-void ALecho_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_ECHO_DELAY:
-            *val = props->Echo.Delay;
-            break;
-
-        case AL_ECHO_LRDELAY:
-            *val = props->Echo.LRDelay;
-            break;
-
-        case AL_ECHO_DAMPING:
-            *val = props->Echo.Damping;
-            break;
-
-        case AL_ECHO_FEEDBACK:
-            *val = props->Echo.Feedback;
-            break;
-
-        case AL_ECHO_SPREAD:
-            *val = props->Echo.Spread;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALecho_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALecho_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALecho);

Alc/effects/equalizer.c

@@ -1,380 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2013 by Mike Gorchak
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-/*  The document  "Effects Extension Guide.pdf"  says that low and high  *
- *  frequencies are cutoff frequencies. This is not fully correct, they  *
- *  are corner frequencies for low and high shelf filters. If they were  *
- *  just cutoff frequencies, there would be no need in cutoff frequency  *
- *  gains, which are present.  Documentation for  "Creative Proteus X2"  *
- *  software describes  4-band equalizer functionality in a much better  *
- *  way.  This equalizer seems  to be a predecessor  of  OpenAL  4-band  *
- *  equalizer.  With low and high  shelf filters  we are able to cutoff  *
- *  frequencies below and/or above corner frequencies using attenuation  *
- *  gains (below 1.0) and amplify all low and/or high frequencies using  *
- *  gains above 1.0.                                                     *
- *                                                                       *
- *     Low-shelf       Low Mid Band      High Mid Band     High-shelf    *
- *      corner            center             center          corner      *
- *     frequency        frequency          frequency       frequency     *
- *    50Hz..800Hz     200Hz..3000Hz      1000Hz..8000Hz  4000Hz..16000Hz *
- *                                                                       *
- *          |               |                  |               |         *
- *          |               |                  |               |         *
- *   B -----+            /--+--\            /--+--\            +-----    *
- *   O      |\          |   |   |          |   |   |          /|         *
- *   O      | \        -    |    -        -    |    -        / |         *
- *   S +    |  \      |     |     |      |     |     |      /  |         *
- *   T      |   |    |      |      |    |      |      |    |   |         *
- * ---------+---------------+------------------+---------------+-------- *
- *   C      |   |    |      |      |    |      |      |    |   |         *
- *   U -    |  /      |     |     |      |     |     |      \  |         *
- *   T      | /        -    |    -        -    |    -        \ |         *
- *   O      |/          |   |   |          |   |   |          \|         *
- *   F -----+            \--+--/            \--+--/            +-----    *
- *   F      |               |                  |               |         *
- *          |               |                  |               |         *
- *                                                                       *
- * Gains vary from 0.126 up to 7.943, which means from -18dB attenuation *
- * up to +18dB amplification. Band width varies from 0.01 up to 1.0 in   *
- * octaves for two mid bands.                                            *
- *                                                                       *
- * Implementation is based on the "Cookbook formulae for audio EQ biquad *
- * filter coefficients" by Robert Bristow-Johnson                        *
- * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt                   */
-
-
-/* The maximum number of sample frames per update. */
-#define MAX_UPDATE_SAMPLES 256
-
-typedef struct ALequalizerState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    /* Effect gains for each channel */
-    ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
-
-    /* Effect parameters */
-    ALfilterState filter[4][MAX_EFFECT_CHANNELS];
-
-    ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES];
-} ALequalizerState;
-
-static ALvoid ALequalizerState_Destruct(ALequalizerState *state);
-static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *device);
-static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props);
-static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALequalizerState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
-
-
-static void ALequalizerState_Construct(ALequalizerState *state)
-{
-    int it, ft;
-
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALequalizerState, ALeffectState, state);
-
-    /* Initialize sample history only on filter creation to avoid */
-    /* sound clicks if filter settings were changed in runtime.   */
-    for(it = 0; it < 4; it++)
-    {
-        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
-            ALfilterState_clear(&state->filter[it][ft]);
-    }
-}
-
-static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
-{
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), ALCdevice *UNUSED(device))
-{
-    return AL_TRUE;
-}
-
-static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props)
-{
-    ALfloat frequency = (ALfloat)device->Frequency;
-    ALfloat gain, freq_mult;
-    ALuint i;
-
-    STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
-    STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
-    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
-        ComputeFirstOrderGains(device->FOAOut, IdentityMatrixf.m[i],
-                               slot->Params.Gain, state->Gain[i]);
-
-    /* Calculate coefficients for the each type of filter. Note that the shelf
-     * filters' gain is for the reference frequency, which is the centerpoint
-     * of the transition band.
-     */
-    gain = maxf(sqrtf(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */
-    freq_mult = props->Equalizer.LowCutoff/frequency;
-    ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf,
-        gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
-    );
-    /* Copy the filter coefficients for the other input channels. */
-    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
-        ALfilterState_copyParams(&state->filter[0][i], &state->filter[0][0]);
-
-    gain = maxf(props->Equalizer.Mid1Gain, 0.0625f);
-    freq_mult = props->Equalizer.Mid1Center/frequency;
-    ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking,
-        gain, freq_mult, calc_rcpQ_from_bandwidth(
-            freq_mult, props->Equalizer.Mid1Width
-        )
-    );
-    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
-        ALfilterState_copyParams(&state->filter[1][i], &state->filter[1][0]);
-
-    gain = maxf(props->Equalizer.Mid2Gain, 0.0625f);
-    freq_mult = props->Equalizer.Mid2Center/frequency;
-    ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking,
-        gain, freq_mult, calc_rcpQ_from_bandwidth(
-            freq_mult, props->Equalizer.Mid2Width
-        )
-    );
-    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
-        ALfilterState_copyParams(&state->filter[2][i], &state->filter[2][0]);
-
-    gain = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f);
-    freq_mult = props->Equalizer.HighCutoff/frequency;
-    ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf,
-        gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
-    );
-    for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
-        ALfilterState_copyParams(&state->filter[3][i], &state->filter[3][0]);
-}
-
-static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer;
-    ALsizei it, kt, ft;
-    ALsizei base;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base);
-
-        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
-            ALfilterState_process(&state->filter[0][ft], Samples[0][ft], &SamplesIn[ft][base], td);
-        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
-            ALfilterState_process(&state->filter[1][ft], Samples[1][ft], Samples[0][ft], td);
-        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
-            ALfilterState_process(&state->filter[2][ft], Samples[2][ft], Samples[1][ft], td);
-        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
-            ALfilterState_process(&state->filter[3][ft], Samples[3][ft], Samples[2][ft], td);
-
-        for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
-        {
-            for(kt = 0;kt < NumChannels;kt++)
-            {
-                ALfloat gain = state->Gain[ft][kt];
-                if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-                    continue;
-
-                for(it = 0;it < td;it++)
-                    SamplesOut[kt][base+it] += gain * Samples[3][ft][it];
-            }
-        }
-
-        base += td;
-    }
-}
-
-
-typedef struct ALequalizerStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALequalizerStateFactory;
-
-ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory))
-{
-    ALequalizerState *state;
-
-    NEW_OBJ0(state, ALequalizerState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALequalizerStateFactory);
-
-ALeffectStateFactory *ALequalizerStateFactory_getFactory(void)
-{
-    static ALequalizerStateFactory EqualizerFactory = { { GET_VTABLE2(ALequalizerStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
-}
-
-
-void ALequalizer_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALequalizer_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALequalizer_setParami(effect, context, param, vals[0]);
-}
-void ALequalizer_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_EQUALIZER_LOW_GAIN:
-            if(!(val >= AL_EQUALIZER_MIN_LOW_GAIN && val <= AL_EQUALIZER_MAX_LOW_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.LowGain = val;
-            break;
-
-        case AL_EQUALIZER_LOW_CUTOFF:
-            if(!(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.LowCutoff = val;
-            break;
-
-        case AL_EQUALIZER_MID1_GAIN:
-            if(!(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.Mid1Gain = val;
-            break;
-
-        case AL_EQUALIZER_MID1_CENTER:
-            if(!(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.Mid1Center = val;
-            break;
-
-        case AL_EQUALIZER_MID1_WIDTH:
-            if(!(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.Mid1Width = val;
-            break;
-
-        case AL_EQUALIZER_MID2_GAIN:
-            if(!(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.Mid2Gain = val;
-            break;
-
-        case AL_EQUALIZER_MID2_CENTER:
-            if(!(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.Mid2Center = val;
-            break;
-
-        case AL_EQUALIZER_MID2_WIDTH:
-            if(!(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.Mid2Width = val;
-            break;
-
-        case AL_EQUALIZER_HIGH_GAIN:
-            if(!(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.HighGain = val;
-            break;
-
-        case AL_EQUALIZER_HIGH_CUTOFF:
-            if(!(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Equalizer.HighCutoff = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALequalizer_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALequalizer_setParamf(effect, context, param, vals[0]);
-}
-
-void ALequalizer_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-void ALequalizer_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALequalizer_getParami(effect, context, param, vals);
-}
-void ALequalizer_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_EQUALIZER_LOW_GAIN:
-            *val = props->Equalizer.LowGain;
-            break;
-
-        case AL_EQUALIZER_LOW_CUTOFF:
-            *val = props->Equalizer.LowCutoff;
-            break;
-
-        case AL_EQUALIZER_MID1_GAIN:
-            *val = props->Equalizer.Mid1Gain;
-            break;
-
-        case AL_EQUALIZER_MID1_CENTER:
-            *val = props->Equalizer.Mid1Center;
-            break;
-
-        case AL_EQUALIZER_MID1_WIDTH:
-            *val = props->Equalizer.Mid1Width;
-            break;
-
-        case AL_EQUALIZER_MID2_GAIN:
-            *val = props->Equalizer.Mid2Gain;
-            break;
-
-        case AL_EQUALIZER_MID2_CENTER:
-            *val = props->Equalizer.Mid2Center;
-            break;
-
-        case AL_EQUALIZER_MID2_WIDTH:
-            *val = props->Equalizer.Mid2Width;
-            break;
-
-        case AL_EQUALIZER_HIGH_GAIN:
-            *val = props->Equalizer.HighGain;
-            break;
-
-        case AL_EQUALIZER_HIGH_CUTOFF:
-            *val = props->Equalizer.HighCutoff;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALequalizer_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALequalizer_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALequalizer);

Alc/effects/flanger.c

@@ -1,408 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2013 by Mike Gorchak
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-enum FlangerWaveForm {
-    FWF_Triangle = AL_FLANGER_WAVEFORM_TRIANGLE,
-    FWF_Sinusoid = AL_FLANGER_WAVEFORM_SINUSOID
-};
-
-typedef struct ALflangerState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    ALfloat *SampleBuffer[2];
-    ALsizei BufferLength;
-    ALsizei offset;
-    ALsizei lfo_range;
-    ALfloat lfo_scale;
-    ALint lfo_disp;
-
-    /* Gains for left and right sides */
-    ALfloat Gain[2][MAX_OUTPUT_CHANNELS];
-
-    /* effect parameters */
-    enum FlangerWaveForm waveform;
-    ALint delay;
-    ALfloat depth;
-    ALfloat feedback;
-} ALflangerState;
-
-static ALvoid ALflangerState_Destruct(ALflangerState *state);
-static ALboolean ALflangerState_deviceUpdate(ALflangerState *state, ALCdevice *Device);
-static ALvoid ALflangerState_update(ALflangerState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
-static ALvoid ALflangerState_process(ALflangerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALflangerState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALflangerState);
-
-
-static void ALflangerState_Construct(ALflangerState *state)
-{
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALflangerState, ALeffectState, state);
-
-    state->BufferLength = 0;
-    state->SampleBuffer[0] = NULL;
-    state->SampleBuffer[1] = NULL;
-    state->offset = 0;
-    state->lfo_range = 1;
-    state->waveform = FWF_Triangle;
-}
-
-static ALvoid ALflangerState_Destruct(ALflangerState *state)
-{
-    al_free(state->SampleBuffer[0]);
-    state->SampleBuffer[0] = NULL;
-    state->SampleBuffer[1] = NULL;
-
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALflangerState_deviceUpdate(ALflangerState *state, ALCdevice *Device)
-{
-    ALsizei maxlen;
-    ALsizei it;
-
-    maxlen = fastf2i(AL_FLANGER_MAX_DELAY * 2.0f * Device->Frequency) + 1;
-    maxlen = NextPowerOf2(maxlen);
-
-    if(maxlen != state->BufferLength)
-    {
-        void *temp = al_calloc(16, maxlen * sizeof(ALfloat) * 2);
-        if(!temp) return AL_FALSE;
-
-        al_free(state->SampleBuffer[0]);
-        state->SampleBuffer[0] = temp;
-        state->SampleBuffer[1] = state->SampleBuffer[0] + maxlen;
-
-        state->BufferLength = maxlen;
-    }
-
-    for(it = 0;it < state->BufferLength;it++)
-    {
-        state->SampleBuffer[0][it] = 0.0f;
-        state->SampleBuffer[1][it] = 0.0f;
-    }
-
-    return AL_TRUE;
-}
-
-static ALvoid ALflangerState_update(ALflangerState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
-{
-    ALfloat frequency = (ALfloat)Device->Frequency;
-    ALfloat coeffs[MAX_AMBI_COEFFS];
-    ALfloat rate;
-    ALint phase;
-
-    switch(props->Flanger.Waveform)
-    {
-        case AL_FLANGER_WAVEFORM_TRIANGLE:
-            state->waveform = FWF_Triangle;
-            break;
-        case AL_FLANGER_WAVEFORM_SINUSOID:
-            state->waveform = FWF_Sinusoid;
-            break;
-    }
-    state->feedback = props->Flanger.Feedback;
-    state->delay = fastf2i(props->Flanger.Delay * frequency);
-    /* The LFO depth is scaled to be relative to the sample delay. */
-    state->depth = props->Flanger.Depth * state->delay;
-
-    /* Gains for left and right sides */
-    CalcAngleCoeffs(-F_PI_2, 0.0f, 0.0f, coeffs);
-    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[0]);
-    CalcAngleCoeffs( F_PI_2, 0.0f, 0.0f, coeffs);
-    ComputePanningGains(Device->Dry, coeffs, Slot->Params.Gain, state->Gain[1]);
-
-    phase = props->Flanger.Phase;
-    rate = props->Flanger.Rate;
-    if(!(rate > 0.0f))
-    {
-        state->lfo_scale = 0.0f;
-        state->lfo_range = 1;
-        state->lfo_disp = 0;
-    }
-    else
-    {
-        /* Calculate LFO coefficient */
-        state->lfo_range = fastf2i(frequency/rate + 0.5f);
-        switch(state->waveform)
-        {
-            case FWF_Triangle:
-                state->lfo_scale = 4.0f / state->lfo_range;
-                break;
-            case FWF_Sinusoid:
-                state->lfo_scale = F_TAU / state->lfo_range;
-                break;
-        }
-
-        /* Calculate lfo phase displacement */
-        if(phase >= 0)
-            state->lfo_disp = fastf2i(state->lfo_range * (phase/360.0f));
-        else
-            state->lfo_disp = fastf2i(state->lfo_range * ((360+phase)/360.0f));
-    }
-}
-
-static void GetTriangleDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
-                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
-                              const ALsizei todo)
-{
-    ALsizei i;
-    for(i = 0;i < todo;i++)
-    {
-        delays[i] = fastf2i((1.0f - fabsf(2.0f - lfo_scale*offset)) * depth) + delay;
-        offset = (offset+1)%lfo_range;
-    }
-}
-
-static void GetSinusoidDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
-                              const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
-                              const ALsizei todo)
-{
-    ALsizei i;
-    for(i = 0;i < todo;i++)
-    {
-        delays[i] = fastf2i(sinf(lfo_scale*offset) * depth) + delay;
-        offset = (offset+1)%lfo_range;
-    }
-}
-
-static ALvoid ALflangerState_process(ALflangerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    ALfloat *restrict leftbuf = state->SampleBuffer[0];
-    ALfloat *restrict rightbuf = state->SampleBuffer[1];
-    const ALsizei bufmask = state->BufferLength-1;
-    const ALfloat feedback = state->feedback;
-    ALsizei offset = state->offset;
-    ALsizei i, c;
-    ALsizei base;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        const ALsizei todo = mini(128, SamplesToDo-base);
-        ALfloat temps[128][2];
-        ALint moddelays[2][128];
-
-        switch(state->waveform)
-        {
-            case FWF_Triangle:
-                GetTriangleDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
-                                  state->lfo_scale, state->depth, state->delay, todo);
-                GetTriangleDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
-                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
-                                  todo);
-                break;
-            case FWF_Sinusoid:
-                GetSinusoidDelays(moddelays[0], offset%state->lfo_range, state->lfo_range,
-                                  state->lfo_scale, state->depth, state->delay, todo);
-                GetSinusoidDelays(moddelays[1], (offset+state->lfo_disp)%state->lfo_range,
-                                  state->lfo_range, state->lfo_scale, state->depth, state->delay,
-                                  todo);
-                break;
-        }
-
-        for(i = 0;i < todo;i++)
-        {
-            leftbuf[offset&bufmask] = SamplesIn[0][base+i];
-            temps[i][0] = leftbuf[(offset-moddelays[0][i])&bufmask] * feedback;
-            leftbuf[offset&bufmask] += temps[i][0];
-
-            rightbuf[offset&bufmask] = SamplesIn[0][base+i];
-            temps[i][1] = rightbuf[(offset-moddelays[1][i])&bufmask] * feedback;
-            rightbuf[offset&bufmask] += temps[i][1];
-
-            offset++;
-        }
-
-        for(c = 0;c < NumChannels;c++)
-        {
-            ALfloat gain = state->Gain[0][c];
-            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
-            {
-                for(i = 0;i < todo;i++)
-                    SamplesOut[c][i+base] += temps[i][0] * gain;
-            }
-
-            gain = state->Gain[1][c];
-            if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
-            {
-                for(i = 0;i < todo;i++)
-                    SamplesOut[c][i+base] += temps[i][1] * gain;
-            }
-        }
-
-        base += todo;
-    }
-
-    state->offset = offset;
-}
-
-
-typedef struct ALflangerStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALflangerStateFactory;
-
-ALeffectState *ALflangerStateFactory_create(ALflangerStateFactory *UNUSED(factory))
-{
-    ALflangerState *state;
-
-    NEW_OBJ0(state, ALflangerState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALflangerStateFactory);
-
-ALeffectStateFactory *ALflangerStateFactory_getFactory(void)
-{
-    static ALflangerStateFactory FlangerFactory = { { GET_VTABLE2(ALflangerStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &FlangerFactory);
-}
-
-
-void ALflanger_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_FLANGER_WAVEFORM:
-            if(!(val >= AL_FLANGER_MIN_WAVEFORM && val <= AL_FLANGER_MAX_WAVEFORM))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Flanger.Waveform = val;
-            break;
-
-        case AL_FLANGER_PHASE:
-            if(!(val >= AL_FLANGER_MIN_PHASE && val <= AL_FLANGER_MAX_PHASE))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Flanger.Phase = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALflanger_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALflanger_setParami(effect, context, param, vals[0]);
-}
-void ALflanger_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_FLANGER_RATE:
-            if(!(val >= AL_FLANGER_MIN_RATE && val <= AL_FLANGER_MAX_RATE))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Flanger.Rate = val;
-            break;
-
-        case AL_FLANGER_DEPTH:
-            if(!(val >= AL_FLANGER_MIN_DEPTH && val <= AL_FLANGER_MAX_DEPTH))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Flanger.Depth = val;
-            break;
-
-        case AL_FLANGER_FEEDBACK:
-            if(!(val >= AL_FLANGER_MIN_FEEDBACK && val <= AL_FLANGER_MAX_FEEDBACK))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Flanger.Feedback = val;
-            break;
-
-        case AL_FLANGER_DELAY:
-            if(!(val >= AL_FLANGER_MIN_DELAY && val <= AL_FLANGER_MAX_DELAY))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Flanger.Delay = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALflanger_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALflanger_setParamf(effect, context, param, vals[0]);
-}
-
-void ALflanger_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_FLANGER_WAVEFORM:
-            *val = props->Flanger.Waveform;
-            break;
-
-        case AL_FLANGER_PHASE:
-            *val = props->Flanger.Phase;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALflanger_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALflanger_getParami(effect, context, param, vals);
-}
-void ALflanger_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_FLANGER_RATE:
-            *val = props->Flanger.Rate;
-            break;
-
-        case AL_FLANGER_DEPTH:
-            *val = props->Flanger.Depth;
-            break;
-
-        case AL_FLANGER_FEEDBACK:
-            *val = props->Flanger.Feedback;
-            break;
-
-        case AL_FLANGER_DELAY:
-            *val = props->Flanger.Delay;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALflanger_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALflanger_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALflanger);

Alc/effects/modulator.c

@@ -1,311 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2009 by Chris Robinson.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALmodulatorState {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    void (*Process)(ALfloat*, const ALfloat*, ALsizei, const ALsizei, ALsizei);
-
-    ALsizei index;
-    ALsizei step;
-
-    ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
-
-    ALfilterState Filter[MAX_EFFECT_CHANNELS];
-} ALmodulatorState;
-
-static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state);
-static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *device);
-static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
-static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALmodulatorState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALmodulatorState);
-
-
-#define WAVEFORM_FRACBITS  24
-#define WAVEFORM_FRACONE   (1<<WAVEFORM_FRACBITS)
-#define WAVEFORM_FRACMASK  (WAVEFORM_FRACONE-1)
-
-static inline ALfloat Sin(ALsizei index)
-{
-    return sinf(index*(F_TAU/WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f;
-}
-
-static inline ALfloat Saw(ALsizei index)
-{
-    return (ALfloat)index / WAVEFORM_FRACONE;
-}
-
-static inline ALfloat Square(ALsizei index)
-{
-    return (ALfloat)((index >> (WAVEFORM_FRACBITS - 1)) & 1);
-}
-
-#define DECL_TEMPLATE(func)                                                   \
-static void Modulate##func(ALfloat *restrict dst, const ALfloat *restrict src,\
-                           ALsizei index, const ALsizei step, ALsizei todo)   \
-{                                                                             \
-    ALsizei i;                                                                \
-    for(i = 0;i < todo;i++)                                                   \
-    {                                                                         \
-        index += step;                                                        \
-        index &= WAVEFORM_FRACMASK;                                           \
-        dst[i] = src[i] * func(index);                                        \
-    }                                                                         \
-}
-
-DECL_TEMPLATE(Sin)
-DECL_TEMPLATE(Saw)
-DECL_TEMPLATE(Square)
-
-#undef DECL_TEMPLATE
-
-
-static void ALmodulatorState_Construct(ALmodulatorState *state)
-{
-    ALuint i;
-
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALmodulatorState, ALeffectState, state);
-
-    state->index = 0;
-    state->step = 1;
-
-    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
-        ALfilterState_clear(&state->Filter[i]);
-}
-
-static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state)
-{
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *UNUSED(state), ALCdevice *UNUSED(device))
-{
-    return AL_TRUE;
-}
-
-static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props)
-{
-    ALfloat cw, a;
-    ALsizei i;
-
-    if(props->Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
-        state->Process = ModulateSin;
-    else if(props->Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
-        state->Process = ModulateSaw;
-    else /*if(Slot->Params.EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)*/
-        state->Process = ModulateSquare;
-
-    state->step = fastf2i(props->Modulator.Frequency*WAVEFORM_FRACONE /
-                          Device->Frequency);
-    if(state->step == 0) state->step = 1;
-
-    /* Custom filter coeffs, which match the old version instead of a low-shelf. */
-    cw = cosf(F_TAU * props->Modulator.HighPassCutoff / Device->Frequency);
-    a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
-
-    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
-    {
-        state->Filter[i].b0 = a;
-        state->Filter[i].b1 = -a;
-        state->Filter[i].b2 = 0.0f;
-        state->Filter[i].a1 = -a;
-        state->Filter[i].a2 = 0.0f;
-    }
-
-    STATIC_CAST(ALeffectState,state)->OutBuffer = Device->FOAOut.Buffer;
-    STATIC_CAST(ALeffectState,state)->OutChannels = Device->FOAOut.NumChannels;
-    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
-        ComputeFirstOrderGains(Device->FOAOut, IdentityMatrixf.m[i],
-                               Slot->Params.Gain, state->Gain[i]);
-}
-
-static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
-    const ALsizei step = state->step;
-    ALsizei index = state->index;
-    ALsizei base;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        ALfloat temps[2][128];
-        ALsizei td = mini(128, SamplesToDo-base);
-        ALsizei i, j, k;
-
-        for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
-        {
-            ALfilterState_process(&state->Filter[j], temps[0], &SamplesIn[j][base], td);
-            state->Process(temps[1], temps[0], index, step, td);
-
-            for(k = 0;k < NumChannels;k++)
-            {
-                ALfloat gain = state->Gain[j][k];
-                if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-                    continue;
-
-                for(i = 0;i < td;i++)
-                    SamplesOut[k][base+i] += gain * temps[1][i];
-            }
-        }
-
-        for(i = 0;i < td;i++)
-        {
-            index += step;
-            index &= WAVEFORM_FRACMASK;
-        }
-        base += td;
-    }
-    state->index = index;
-}
-
-
-typedef struct ALmodulatorStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALmodulatorStateFactory;
-
-static ALeffectState *ALmodulatorStateFactory_create(ALmodulatorStateFactory *UNUSED(factory))
-{
-    ALmodulatorState *state;
-
-    NEW_OBJ0(state, ALmodulatorState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALmodulatorStateFactory);
-
-ALeffectStateFactory *ALmodulatorStateFactory_getFactory(void)
-{
-    static ALmodulatorStateFactory ModulatorFactory = { { GET_VTABLE2(ALmodulatorStateFactory, ALeffectStateFactory) } };
-
-    return STATIC_CAST(ALeffectStateFactory, &ModulatorFactory);
-}
-
-
-void ALmodulator_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_RING_MODULATOR_FREQUENCY:
-            if(!(val >= AL_RING_MODULATOR_MIN_FREQUENCY && val <= AL_RING_MODULATOR_MAX_FREQUENCY))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Modulator.Frequency = val;
-            break;
-
-        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
-            if(!(val >= AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF && val <= AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Modulator.HighPassCutoff = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALmodulator_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    ALmodulator_setParamf(effect, context, param, vals[0]);
-}
-void ALmodulator_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
-{
-    ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_RING_MODULATOR_FREQUENCY:
-        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
-            ALmodulator_setParamf(effect, context, param, (ALfloat)val);
-            break;
-
-        case AL_RING_MODULATOR_WAVEFORM:
-            if(!(val >= AL_RING_MODULATOR_MIN_WAVEFORM && val <= AL_RING_MODULATOR_MAX_WAVEFORM))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            props->Modulator.Waveform = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALmodulator_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
-    ALmodulator_setParami(effect, context, param, vals[0]);
-}
-
-void ALmodulator_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_RING_MODULATOR_FREQUENCY:
-            *val = (ALint)props->Modulator.Frequency;
-            break;
-        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
-            *val = (ALint)props->Modulator.HighPassCutoff;
-            break;
-        case AL_RING_MODULATOR_WAVEFORM:
-            *val = props->Modulator.Waveform;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALmodulator_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
-    ALmodulator_getParami(effect, context, param, vals);
-}
-void ALmodulator_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    const ALeffectProps *props = &effect->Props;
-    switch(param)
-    {
-        case AL_RING_MODULATOR_FREQUENCY:
-            *val = props->Modulator.Frequency;
-            break;
-        case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
-            *val = props->Modulator.HighPassCutoff;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALmodulator_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    ALmodulator_getParamf(effect, context, param, vals);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALmodulator);

Alc/effects/null.c

@@ -1,179 +0,0 @@
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-
-
-typedef struct ALnullState {
-    DERIVE_FROM_TYPE(ALeffectState);
-} ALnullState;
-
-/* Forward-declare "virtual" functions to define the vtable with. */
-static ALvoid ALnullState_Destruct(ALnullState *state);
-static ALboolean ALnullState_deviceUpdate(ALnullState *state, ALCdevice *device);
-static ALvoid ALnullState_update(ALnullState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props);
-static ALvoid ALnullState_process(ALnullState *state, ALsizei samplesToDo, const ALfloatBUFFERSIZE*restrict samplesIn, ALfloatBUFFERSIZE*restrict samplesOut, ALsizei NumChannels);
-static void *ALnullState_New(size_t size);
-static void ALnullState_Delete(void *ptr);
-
-/* Define the ALeffectState vtable for this type. */
-DEFINE_ALEFFECTSTATE_VTABLE(ALnullState);
-
-
-/* This constructs the effect state. It's called when the object is first
- * created. Make sure to call the parent Construct function first, and set the
- * vtable!
- */
-static void ALnullState_Construct(ALnullState *state)
-{
-    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
-    SET_VTABLE2(ALnullState, ALeffectState, state);
-}
-
-/* This destructs (not free!) the effect state. It's called only when the
- * effect slot is no longer used. Make sure to call the parent Destruct
- * function before returning!
- */
-static ALvoid ALnullState_Destruct(ALnullState *state)
-{
-    ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-/* This updates the device-dependant effect state. This is called on
- * initialization and any time the device parameters (eg. playback frequency,
- * format) have been changed.
- */
-static ALboolean ALnullState_deviceUpdate(ALnullState* UNUSED(state), ALCdevice* UNUSED(device))
-{
-    return AL_TRUE;
-}
-
-/* This updates the effect state. This is called any time the effect is
- * (re)loaded into a slot.
- */
-static ALvoid ALnullState_update(ALnullState* UNUSED(state), const ALCdevice* UNUSED(device), const ALeffectslot* UNUSED(slot), const ALeffectProps* UNUSED(props))
-{
-}
-
-/* This processes the effect state, for the given number of samples from the
- * input to the output buffer. The result should be added to the output buffer,
- * not replace it.
- */
-static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALsizei UNUSED(samplesToDo), const ALfloatBUFFERSIZE*restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALsizei UNUSED(NumChannels))
-{
-}
-
-/* This allocates memory to store the object, before it gets constructed.
- * DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method.
- */
-static void *ALnullState_New(size_t size)
-{
-    return al_malloc(16, size);
-}
-
-/* This frees the memory used by the object, after it has been destructed.
- * DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method.
- */
-static void ALnullState_Delete(void *ptr)
-{
-    al_free(ptr);
-}
-
-
-typedef struct ALnullStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALnullStateFactory;
-
-/* Creates ALeffectState objects of the appropriate type. */
-ALeffectState *ALnullStateFactory_create(ALnullStateFactory *UNUSED(factory))
-{
-    ALnullState *state;
-
-    NEW_OBJ0(state, ALnullState)();
-    if(!state) return NULL;
-
-    return STATIC_CAST(ALeffectState, state);
-}
-
-/* Define the ALeffectStateFactory vtable for this type. */
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALnullStateFactory);
-
-ALeffectStateFactory *ALnullStateFactory_getFactory(void)
-{
-    static ALnullStateFactory NullFactory = { { GET_VTABLE2(ALnullStateFactory, ALeffectStateFactory) } };
-    return STATIC_CAST(ALeffectStateFactory, &NullFactory);
-}
-
-
-void ALnull_setParami(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALnull_setParamiv(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, const ALint* UNUSED(vals))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALnull_setParamf(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALnull_setParamfv(ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat* UNUSED(vals))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-
-void ALnull_getParami(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(val))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALnull_getParamiv(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(vals))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALnull_getParamf(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(val))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-void ALnull_getParamfv(const ALeffect* UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(vals))
-{
-    switch(param)
-    {
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-
-DEFINE_ALEFFECT_VTABLE(ALnull);

Alc/hrtf.h

@@ -1,52 +0,0 @@
-#ifndef ALC_HRTF_H
-#define ALC_HRTF_H
-
-#include "AL/al.h"
-#include "AL/alc.h"
-
-#include "alMain.h"
-#include "alstring.h"
-#include "atomic.h"
-
-
-/* The maximum number of virtual speakers used to generate HRTF coefficients
- * for decoding B-Format.
- */
-#define HRTF_AMBI_MAX_CHANNELS 16
-
-
-struct HrtfEntry;
-
-struct Hrtf {
-    RefCount ref;
-
-    ALuint sampleRate;
-    ALsizei irSize;
-    ALubyte evCount;
-
-    const ALubyte *azCount;
-    const ALushort *evOffset;
-    const ALfloat (*coeffs)[2];
-    const ALubyte (*delays)[2];
-};
-
-
-void FreeHrtfs(void);
-
-vector_EnumeratedHrtf EnumerateHrtf(const_al_string devname);
-void FreeHrtfList(vector_EnumeratedHrtf *list);
-struct Hrtf *GetLoadedHrtf(struct HrtfEntry *entry);
-void Hrtf_IncRef(struct Hrtf *hrtf);
-void Hrtf_DecRef(struct Hrtf *hrtf);
-
-void GetHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat spread, ALfloat (*coeffs)[2], ALsizei *delays);
-
-/**
- * Produces HRTF filter coefficients for decoding B-Format, given a set of
- * virtual speaker positions and HF/LF matrices for decoding to them. The
- * returned coefficients are ordered and scaled according to the matrices.
- * Returns the maximum impulse-response length of the generated coefficients.
- */
-ALsizei BuildBFormatHrtf(const struct Hrtf *Hrtf, DirectHrtfState *state, ALsizei NumChannels, const ALfloat (*restrict AmbiPoints)[2], const ALfloat (*restrict AmbiMatrix)[2][MAX_AMBI_COEFFS], ALsizei AmbiCount);
-
-#endif /* ALC_HRTF_H */

Alc/mastering.c

@@ -1,255 +0,0 @@
-#include "config.h"
-
-#include <math.h>
-
-#include "alu.h"
-#include "almalloc.h"
-
-#define RMS_WINDOW_SIZE (1<<7)
-#define RMS_WINDOW_MASK (RMS_WINDOW_SIZE-1)
-#define RMS_VALUE_MAX  (1<<24)
-
-#define LOOKAHEAD_SIZE (1<<13)
-#define LOOKAHEAD_MASK (LOOKAHEAD_SIZE-1)
-
-static_assert(RMS_VALUE_MAX < (UINT_MAX / RMS_WINDOW_SIZE), "RMS_VALUE_MAX is too big");
-
-typedef struct Compressor {
-    ALfloat PreGain;
-    ALfloat PostGain;
-    ALboolean SummedLink;
-    ALfloat AttackMin;
-    ALfloat AttackMax;
-    ALfloat ReleaseMin;
-    ALfloat ReleaseMax;
-    ALfloat Ratio;
-    ALfloat Threshold;
-    ALfloat Knee;
-    ALuint SampleRate;
-
-    ALuint RmsSum;
-    ALuint *RmsWindow;
-    ALsizei RmsIndex;
-    ALfloat Envelope[BUFFERSIZE];
-    ALfloat EnvLast;
-} Compressor;
-
-/* Multichannel compression is linked via one of two modes:
- *
- *   Summed - Absolute sum of all channels.
- *   Maxed  - Absolute maximum of any channel.
- */
-static void SumChannels(Compressor *Comp, const ALsizei NumChans, const ALsizei SamplesToDo,
-                        ALfloat (*restrict OutBuffer)[BUFFERSIZE])
-{
-    ALsizei c, i;
-
-    for(i = 0;i < SamplesToDo;i++)
-        Comp->Envelope[i] = 0.0f;
-
-    for(c = 0;c < NumChans;c++)
-    {
-        for(i = 0;i < SamplesToDo;i++)
-            Comp->Envelope[i] += OutBuffer[c][i];
-    }
-
-    for(i = 0;i < SamplesToDo;i++)
-        Comp->Envelope[i] = fabsf(Comp->Envelope[i]);
-}
-
-static void MaxChannels(Compressor *Comp, const ALsizei NumChans, const ALsizei SamplesToDo,
-                        ALfloat (*restrict OutBuffer)[BUFFERSIZE])
-{
-    ALsizei c, i;
-
-    for(i = 0;i < SamplesToDo;i++)
-        Comp->Envelope[i] = 0.0f;
-
-    for(c = 0;c < NumChans;c++)
-    {
-        for(i = 0;i < SamplesToDo;i++)
-            Comp->Envelope[i] = maxf(Comp->Envelope[i], fabsf(OutBuffer[c][i]));
-    }
-}
-
-/* Envelope detection/sensing can be done via:
- *
- *   RMS  - Rectangular windowed root mean square of linking stage.
- *   Peak - Implicit output from linking stage.
- */
-static void RmsDetection(Compressor *Comp, const ALsizei SamplesToDo)
-{
-    ALuint sum = Comp->RmsSum;
-    ALuint *window = Comp->RmsWindow;
-    ALsizei index = Comp->RmsIndex;
-    ALsizei i;
-
-    for(i = 0;i < SamplesToDo;i++)
-    {
-        ALfloat sig = Comp->Envelope[i];
-
-        sum -= window[index];
-        window[index] = fastf2i(minf(sig * sig * 65536.0f, RMS_VALUE_MAX));
-        sum += window[index];
-        index = (index + 1) & RMS_WINDOW_MASK;
-
-        Comp->Envelope[i] = sqrtf(sum / 65536.0f / RMS_WINDOW_SIZE);
-    }
-
-    Comp->RmsSum = sum;
-    Comp->RmsIndex = index;
-}
-
-/* This isn't a very sophisticated envelope follower, but it gets the job
- * done.  First, it operates at logarithmic scales to keep transitions
- * appropriate for human hearing.  Second, it can apply adaptive (automated)
- * attack/release adjustments based on the signal.
- */
-static void FollowEnvelope(Compressor *Comp, const ALsizei SamplesToDo)
-{
-    ALfloat attackMin = Comp->AttackMin;
-    ALfloat attackMax = Comp->AttackMax;
-    ALfloat releaseMin = Comp->ReleaseMin;
-    ALfloat releaseMax = Comp->ReleaseMax;
-    ALfloat last = Comp->EnvLast;
-    ALsizei i;
-
-    for(i = 0;i < SamplesToDo;i++)
-    {
-        ALfloat env = maxf(-6.0f, log10f(Comp->Envelope[i]));
-        ALfloat slope = minf(1.0f, fabsf(env - last) / 4.5f);
-
-        if(env > last)
-            last = minf(env, last + lerp(attackMin, attackMax, 1.0f - (slope * slope)));
-        else
-            last = maxf(env, last + lerp(releaseMin, releaseMax, 1.0f - (slope * slope)));
-
-        Comp->Envelope[i] = last;
-    }
-
-    Comp->EnvLast = last;
-}
-
-/* The envelope is converted to control gain with an optional soft knee. */
-static void EnvelopeGain(Compressor *Comp, const ALsizei SamplesToDo, const ALfloat Slope)
-{
-    const ALfloat threshold = Comp->Threshold;
-    const ALfloat knee = Comp->Knee;
-    ALsizei i;
-
-    if(!(knee > 0.0f))
-    {
-        for(i = 0;i < SamplesToDo;i++)
-        {
-            ALfloat gain = Slope * (threshold - Comp->Envelope[i]);
-            Comp->Envelope[i] = powf(10.0f, minf(0.0f, gain));
-        }
-    }
-    else
-    {
-        const ALfloat lower = threshold - (0.5f * knee);
-        const ALfloat upper = threshold + (0.5f * knee);
-        const ALfloat m = 0.5f * Slope / knee;
-
-        for(i = 0;i < SamplesToDo;i++)
-        {
-            ALfloat env = Comp->Envelope[i];
-            ALfloat gain;
-
-            if(env > lower && env < upper)
-                gain = m * (env - lower) * (lower - env);
-            else
-                gain = Slope * (threshold - env);
-
-            Comp->Envelope[i] = powf(10.0f, minf(0.0f, gain));
-        }
-    }
-}
-
-
-Compressor *CompressorInit(const ALfloat PreGainDb, const ALfloat PostGainDb,
-                           const ALboolean SummedLink, const ALboolean RmsSensing,
-                           const ALfloat AttackTimeMin, const ALfloat AttackTimeMax,
-                           const ALfloat ReleaseTimeMin, const ALfloat ReleaseTimeMax,
-                           const ALfloat Ratio, const ALfloat ThresholdDb,
-                           const ALfloat KneeDb, const ALuint SampleRate)
-{
-    Compressor *Comp;
-    size_t size;
-    ALsizei i;
-
-    size = sizeof(*Comp);
-    if(RmsSensing)
-        size += sizeof(Comp->RmsWindow[0]) * RMS_WINDOW_SIZE;
-    Comp = al_calloc(16, size);
-
-    Comp->PreGain = powf(10.0f, PreGainDb / 20.0f);
-    Comp->PostGain = powf(10.0f, PostGainDb / 20.0f);
-    Comp->SummedLink = SummedLink;
-    Comp->AttackMin = 1.0f / maxf(0.000001f, AttackTimeMin * SampleRate * logf(10.0f));
-    Comp->AttackMax = 1.0f / maxf(0.000001f, AttackTimeMax * SampleRate * logf(10.0f));
-    Comp->ReleaseMin = -1.0f / maxf(0.000001f, ReleaseTimeMin * SampleRate * logf(10.0f));
-    Comp->ReleaseMax = -1.0f / maxf(0.000001f, ReleaseTimeMax * SampleRate * logf(10.0f));
-    Comp->Ratio = Ratio;
-    Comp->Threshold = ThresholdDb / 20.0f;
-    Comp->Knee = maxf(0.0f, KneeDb / 20.0f);
-    Comp->SampleRate = SampleRate;
-
-    Comp->RmsSum = 0;
-    if(RmsSensing)
-        Comp->RmsWindow = (ALuint*)(Comp+1);
-    else
-        Comp->RmsWindow = NULL;
-    Comp->RmsIndex = 0;
-
-    for(i = 0;i < BUFFERSIZE;i++)
-        Comp->Envelope[i] = 0.0f;
-    Comp->EnvLast = -6.0f;
-
-    return Comp;
-}
-
-ALuint GetCompressorSampleRate(const Compressor *Comp)
-{
-    return Comp->SampleRate;
-}
-
-void ApplyCompression(Compressor *Comp, const ALsizei NumChans, const ALsizei SamplesToDo,
-                      ALfloat (*restrict OutBuffer)[BUFFERSIZE])
-{
-    ALsizei c, i;
-
-    if(Comp->PreGain != 1.0f)
-    {
-        for(c = 0;c < NumChans;c++)
-        {
-            for(i = 0;i < SamplesToDo;i++)
-                OutBuffer[c][i] *= Comp->PreGain;
-        }
-    }
-
-    if(Comp->SummedLink)
-        SumChannels(Comp, NumChans, SamplesToDo, OutBuffer);
-    else
-        MaxChannels(Comp, NumChans, SamplesToDo, OutBuffer);
-
-    if(Comp->RmsWindow)
-        RmsDetection(Comp, SamplesToDo);
-    FollowEnvelope(Comp, SamplesToDo);
-
-    if(Comp->Ratio > 0.0f)
-        EnvelopeGain(Comp, SamplesToDo, 1.0f - (1.0f / Comp->Ratio));
-    else
-        EnvelopeGain(Comp, SamplesToDo, 1.0f);
-
-    if(Comp->PostGain != 1.0f)
-    {
-        for(i = 0;i < SamplesToDo;i++)
-            Comp->Envelope[i] *= Comp->PostGain;
-    }
-    for(c = 0;c < NumChans;c++)
-    {
-        for(i = 0;i < SamplesToDo;i++)
-            OutBuffer[c][i] *= Comp->Envelope[i];
-    }
-}

Alc/mixer.c

@@ -1,671 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-#include <ctype.h>
-#include <assert.h>
-
-#include "alMain.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alSource.h"
-#include "alBuffer.h"
-#include "alListener.h"
-#include "alAuxEffectSlot.h"
-#include "alu.h"
-
-#include "mixer_defs.h"
-
-
-static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
-              "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
-
-extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALint *restrict pos_arr, ALsizei size);
-
-
-/* BSinc requires up to 11 extra samples before the current position, and 12 after. */
-static_assert(MAX_PRE_SAMPLES >= 11, "MAX_PRE_SAMPLES must be at least 11!");
-static_assert(MAX_POST_SAMPLES >= 12, "MAX_POST_SAMPLES must be at least 12!");
-
-
-enum Resampler ResamplerDefault = LinearResampler;
-
-static MixerFunc MixSamples = Mix_C;
-static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
-HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
-
-MixerFunc SelectMixer(void)
-{
-#ifdef HAVE_NEON
-    if((CPUCapFlags&CPU_CAP_NEON))
-        return Mix_Neon;
-#endif
-#ifdef HAVE_SSE
-    if((CPUCapFlags&CPU_CAP_SSE))
-        return Mix_SSE;
-#endif
-    return Mix_C;
-}
-
-RowMixerFunc SelectRowMixer(void)
-{
-#ifdef HAVE_NEON
-    if((CPUCapFlags&CPU_CAP_NEON))
-        return MixRow_Neon;
-#endif
-#ifdef HAVE_SSE
-    if((CPUCapFlags&CPU_CAP_SSE))
-        return MixRow_SSE;
-#endif
-    return MixRow_C;
-}
-
-static inline HrtfMixerFunc SelectHrtfMixer(void)
-{
-#ifdef HAVE_NEON
-    if((CPUCapFlags&CPU_CAP_NEON))
-        return MixHrtf_Neon;
-#endif
-#ifdef HAVE_SSE
-    if((CPUCapFlags&CPU_CAP_SSE))
-        return MixHrtf_SSE;
-#endif
-    return MixHrtf_C;
-}
-
-static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
-{
-#ifdef HAVE_NEON
-    if((CPUCapFlags&CPU_CAP_NEON))
-        return MixHrtfBlend_Neon;
-#endif
-#ifdef HAVE_SSE
-    if((CPUCapFlags&CPU_CAP_SSE))
-        return MixHrtfBlend_SSE;
-#endif
-    return MixHrtfBlend_C;
-}
-
-ResamplerFunc SelectResampler(enum Resampler resampler)
-{
-    switch(resampler)
-    {
-        case PointResampler:
-            return Resample_point32_C;
-        case LinearResampler:
-#ifdef HAVE_NEON
-            if((CPUCapFlags&CPU_CAP_NEON))
-                return Resample_lerp32_Neon;
-#endif
-#ifdef HAVE_SSE4_1
-            if((CPUCapFlags&CPU_CAP_SSE4_1))
-                return Resample_lerp32_SSE41;
-#endif
-#ifdef HAVE_SSE2
-            if((CPUCapFlags&CPU_CAP_SSE2))
-                return Resample_lerp32_SSE2;
-#endif
-            return Resample_lerp32_C;
-        case FIR4Resampler:
-#ifdef HAVE_NEON
-            if((CPUCapFlags&CPU_CAP_NEON))
-                return Resample_fir4_32_Neon;
-#endif
-#ifdef HAVE_SSE4_1
-            if((CPUCapFlags&CPU_CAP_SSE4_1))
-                return Resample_fir4_32_SSE41;
-#endif
-#ifdef HAVE_SSE3
-            if((CPUCapFlags&CPU_CAP_SSE3))
-                return Resample_fir4_32_SSE3;
-#endif
-            return Resample_fir4_32_C;
-        case BSincResampler:
-#ifdef HAVE_NEON
-            if((CPUCapFlags&CPU_CAP_NEON))
-                return Resample_bsinc32_Neon;
-#endif
-#ifdef HAVE_SSE
-            if((CPUCapFlags&CPU_CAP_SSE))
-                return Resample_bsinc32_SSE;
-#endif
-            return Resample_bsinc32_C;
-    }
-
-    return Resample_point32_C;
-}
-
-
-void aluInitMixer(void)
-{
-    const char *str;
-
-    if(ConfigValueStr(NULL, NULL, "resampler", &str))
-    {
-        if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
-            ResamplerDefault = PointResampler;
-        else if(strcasecmp(str, "linear") == 0)
-            ResamplerDefault = LinearResampler;
-        else if(strcasecmp(str, "sinc4") == 0)
-            ResamplerDefault = FIR4Resampler;
-        else if(strcasecmp(str, "bsinc") == 0)
-            ResamplerDefault = BSincResampler;
-        else if(strcasecmp(str, "cubic") == 0 || strcasecmp(str, "sinc8") == 0)
-        {
-            WARN("Resampler option \"%s\" is deprecated, using sinc4\n", str);
-            ResamplerDefault = FIR4Resampler;
-        }
-        else
-        {
-            char *end;
-            long n = strtol(str, &end, 0);
-            if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
-                ResamplerDefault = n;
-            else
-                WARN("Invalid resampler: %s\n", str);
-        }
-    }
-
-    MixHrtfBlendSamples = SelectHrtfBlendMixer();
-    MixHrtfSamples = SelectHrtfMixer();
-    MixSamples = SelectMixer();
-}
-
-
-static inline ALfloat Sample_ALbyte(ALbyte val)
-{ return val * (1.0f/128.0f); }
-
-static inline ALfloat Sample_ALshort(ALshort val)
-{ return val * (1.0f/32768.0f); }
-
-static inline ALfloat Sample_ALfloat(ALfloat val)
-{ return val; }
-
-#define DECL_TEMPLATE(T)                                                      \
-static inline void Load_##T(ALfloat *dst, const T *src, ALint srcstep, ALsizei samples)\
-{                                                                             \
-    ALsizei i;                                                                \
-    for(i = 0;i < samples;i++)                                                \
-        dst[i] = Sample_##T(src[i*srcstep]);                                  \
-}
-
-DECL_TEMPLATE(ALbyte)
-DECL_TEMPLATE(ALshort)
-DECL_TEMPLATE(ALfloat)
-
-#undef DECL_TEMPLATE
-
-static void LoadSamples(ALfloat *dst, const ALvoid *src, ALint srcstep, enum FmtType srctype, ALsizei samples)
-{
-    switch(srctype)
-    {
-        case FmtByte:
-            Load_ALbyte(dst, src, srcstep, samples);
-            break;
-        case FmtShort:
-            Load_ALshort(dst, src, srcstep, samples);
-            break;
-        case FmtFloat:
-            Load_ALfloat(dst, src, srcstep, samples);
-            break;
-    }
-}
-
-static inline void SilenceSamples(ALfloat *dst, ALsizei samples)
-{
-    ALsizei i;
-    for(i = 0;i < samples;i++)
-        dst[i] = 0.0f;
-}
-
-
-static const ALfloat *DoFilters(ALfilterState *lpfilter, ALfilterState *hpfilter,
-                                ALfloat *restrict dst, const ALfloat *restrict src,
-                                ALsizei numsamples, enum ActiveFilters type)
-{
-    ALsizei i;
-    switch(type)
-    {
-        case AF_None:
-            ALfilterState_processPassthru(lpfilter, src, numsamples);
-            ALfilterState_processPassthru(hpfilter, src, numsamples);
-            break;
-
-        case AF_LowPass:
-            ALfilterState_process(lpfilter, dst, src, numsamples);
-            ALfilterState_processPassthru(hpfilter, dst, numsamples);
-            return dst;
-        case AF_HighPass:
-            ALfilterState_processPassthru(lpfilter, src, numsamples);
-            ALfilterState_process(hpfilter, dst, src, numsamples);
-            return dst;
-
-        case AF_BandPass:
-            for(i = 0;i < numsamples;)
-            {
-                ALfloat temp[256];
-                ALsizei todo = mini(256, numsamples-i);
-
-                ALfilterState_process(lpfilter, temp, src+i, todo);
-                ALfilterState_process(hpfilter, dst+i, temp, todo);
-                i += todo;
-            }
-            return dst;
-    }
-    return src;
-}
-
-
-ALboolean MixSource(ALvoice *voice, ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo)
-{
-    ALbufferlistitem *BufferListItem;
-    ALbufferlistitem *BufferLoopItem;
-    ALsizei NumChannels, SampleSize;
-    ResamplerFunc Resample;
-    ALsizei DataPosInt;
-    ALsizei DataPosFrac;
-    ALint64 DataSize64;
-    ALint increment;
-    ALsizei Counter;
-    ALsizei OutPos;
-    ALsizei IrSize;
-    bool isplaying;
-    bool firstpass;
-    ALsizei chan;
-    ALsizei send;
-
-    /* Get source info */
-    isplaying      = true; /* Will only be called while playing. */
-    DataPosInt     = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
-    DataPosFrac    = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
-    BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
-    BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
-    NumChannels    = voice->NumChannels;
-    SampleSize     = voice->SampleSize;
-    increment      = voice->Step;
-
-    IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
-
-    Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
-                Resample_copy32_C : voice->Resampler);
-
-    Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
-    firstpass = true;
-    OutPos = 0;
-
-    do {
-        ALsizei SrcBufferSize, DstBufferSize;
-
-        /* Figure out how many buffer samples will be needed */
-        DataSize64  = SamplesToDo-OutPos;
-        DataSize64 *= increment;
-        DataSize64 += DataPosFrac+FRACTIONMASK;
-        DataSize64 >>= FRACTIONBITS;
-        DataSize64 += MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
-
-        SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
-
-        /* Figure out how many samples we can actually mix from this. */
-        DataSize64  = SrcBufferSize;
-        DataSize64 -= MAX_POST_SAMPLES+MAX_PRE_SAMPLES;
-        DataSize64 <<= FRACTIONBITS;
-        DataSize64 -= DataPosFrac;
-
-        DstBufferSize = (ALsizei)((DataSize64+(increment-1)) / increment);
-        DstBufferSize = mini(DstBufferSize, (SamplesToDo-OutPos));
-
-        /* Some mixers like having a multiple of 4, so try to give that unless
-         * this is the last update. */
-        if(OutPos+DstBufferSize < SamplesToDo)
-            DstBufferSize &= ~3;
-
-        for(chan = 0;chan < NumChannels;chan++)
-        {
-            const ALfloat *ResampledData;
-            ALfloat *SrcData = Device->SourceData;
-            ALsizei SrcDataSize;
-
-            /* Load the previous samples into the source data first. */
-            memcpy(SrcData, voice->PrevSamples[chan], MAX_PRE_SAMPLES*sizeof(ALfloat));
-            SrcDataSize = MAX_PRE_SAMPLES;
-
-            if(Source->SourceType == AL_STATIC)
-            {
-                const ALbuffer *ALBuffer = BufferListItem->buffer;
-                const ALubyte *Data = ALBuffer->data;
-                ALsizei DataSize;
-
-                /* Offset buffer data to current channel */
-                Data += chan*SampleSize;
-
-                /* If current pos is beyond the loop range, do not loop */
-                if(!BufferLoopItem || DataPosInt >= ALBuffer->LoopEnd)
-                {
-                    BufferLoopItem = NULL;
-
-                    /* Load what's left to play from the source buffer, and
-                     * clear the rest of the temp buffer */
-                    DataSize = minu(SrcBufferSize - SrcDataSize,
-                                    ALBuffer->SampleLen - DataPosInt);
-
-                    LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
-                                NumChannels, ALBuffer->FmtType, DataSize);
-                    SrcDataSize += DataSize;
-
-                    SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
-                    SrcDataSize += SrcBufferSize - SrcDataSize;
-                }
-                else
-                {
-                    ALsizei LoopStart = ALBuffer->LoopStart;
-                    ALsizei LoopEnd   = ALBuffer->LoopEnd;
-
-                    /* Load what's left of this loop iteration, then load
-                     * repeats of the loop section */
-                    DataSize = minu(SrcBufferSize - SrcDataSize, LoopEnd - DataPosInt);
-
-                    LoadSamples(&SrcData[SrcDataSize], &Data[DataPosInt * NumChannels*SampleSize],
-                                NumChannels, ALBuffer->FmtType, DataSize);
-                    SrcDataSize += DataSize;
-
-                    DataSize = LoopEnd-LoopStart;
-                    while(SrcBufferSize > SrcDataSize)
-                    {
-                        DataSize = mini(SrcBufferSize - SrcDataSize, DataSize);
-
-                        LoadSamples(&SrcData[SrcDataSize], &Data[LoopStart * NumChannels*SampleSize],
-                                    NumChannels, ALBuffer->FmtType, DataSize);
-                        SrcDataSize += DataSize;
-                    }
-                }
-            }
-            else
-            {
-                /* Crawl the buffer queue to fill in the temp buffer */
-                ALbufferlistitem *tmpiter = BufferListItem;
-                ALsizei pos = DataPosInt;
-
-                while(tmpiter && SrcBufferSize > SrcDataSize)
-                {
-                    const ALbuffer *ALBuffer;
-                    if((ALBuffer=tmpiter->buffer) != NULL)
-                    {
-                        const ALubyte *Data = ALBuffer->data;
-                        ALsizei DataSize = ALBuffer->SampleLen;
-
-                        /* Skip the data already played */
-                        if(DataSize <= pos)
-                            pos -= DataSize;
-                        else
-                        {
-                            Data += (pos*NumChannels + chan)*SampleSize;
-                            DataSize -= pos;
-                            pos -= pos;
-
-                            DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
-                            LoadSamples(&SrcData[SrcDataSize], Data, NumChannels,
-                                        ALBuffer->FmtType, DataSize);
-                            SrcDataSize += DataSize;
-                        }
-                    }
-                    tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
-                    if(!tmpiter && BufferLoopItem)
-                        tmpiter = BufferLoopItem;
-                    else if(!tmpiter)
-                    {
-                        SilenceSamples(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
-                        SrcDataSize += SrcBufferSize - SrcDataSize;
-                    }
-                }
-            }
-
-            /* Store the last source samples used for next time. */
-            memcpy(voice->PrevSamples[chan],
-                &SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
-                MAX_PRE_SAMPLES*sizeof(ALfloat)
-            );
-
-            /* Now resample, then filter and mix to the appropriate outputs. */
-            ResampledData = Resample(&voice->ResampleState,
-                &SrcData[MAX_PRE_SAMPLES], DataPosFrac, increment,
-                Device->ResampledData, DstBufferSize
-            );
-            {
-                DirectParams *parms = &voice->Direct.Params[chan];
-                const ALfloat *samples;
-
-                samples = DoFilters(
-                    &parms->LowPass, &parms->HighPass, Device->FilteredData,
-                    ResampledData, DstBufferSize, voice->Direct.FilterType
-                );
-                if(!(voice->Flags&VOICE_HAS_HRTF))
-                {
-                    if(!Counter)
-                        memcpy(parms->Gains.Current, parms->Gains.Target,
-                               sizeof(parms->Gains.Current));
-                    if(!(voice->Flags&VOICE_HAS_NFC))
-                        MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
-                            parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
-                            DstBufferSize
-                        );
-                    else
-                    {
-                        ALfloat *nfcsamples = Device->NFCtrlData;
-                        ALsizei chanoffset = 0;
-
-                        MixSamples(samples,
-                            voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
-                            parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
-                            DstBufferSize
-                        );
-                        chanoffset += voice->Direct.ChannelsPerOrder[0];
-#define APPLY_NFC_MIX(order)                                                  \
-    if(voice->Direct.ChannelsPerOrder[order] > 0)                             \
-    {                                                                         \
-        NfcFilterUpdate##order(&parms->NFCtrlFilter[order-1], nfcsamples,     \
-                               samples, DstBufferSize);                       \
-        MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order],         \
-            voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
-            parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize    \
-        );                                                                    \
-        chanoffset += voice->Direct.ChannelsPerOrder[order];                  \
-    }
-                        APPLY_NFC_MIX(1)
-                        APPLY_NFC_MIX(2)
-                        APPLY_NFC_MIX(3)
-#undef APPLY_NFC_MIX
-                    }
-                }
-                else
-                {
-                    MixHrtfParams hrtfparams;
-                    ALsizei fademix = 0;
-                    int lidx, ridx;
-
-                    lidx = GetChannelIdxByName(Device->RealOut, FrontLeft);
-                    ridx = GetChannelIdxByName(Device->RealOut, FrontRight);
-                    assert(lidx != -1 && ridx != -1);
-
-                    if(!Counter)
-                    {
-                        /* No fading, just overwrite the old HRTF params. */
-                        parms->Hrtf.Old = parms->Hrtf.Target;
-                    }
-                    else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
-                    {
-                        /* The old HRTF params are silent, so overwrite the old
-                         * coefficients with the new, and reset the old gain to
-                         * 0. The future mix will then fade from silence.
-                         */
-                        parms->Hrtf.Old = parms->Hrtf.Target;
-                        parms->Hrtf.Old.Gain = 0.0f;
-                    }
-                    else if(firstpass)
-                    {
-                        ALfloat gain;
-
-                        /* Fade between the coefficients over 128 samples. */
-                        fademix = mini(DstBufferSize, 128);
-
-                        /* The new coefficients need to fade in completely
-                         * since they're replacing the old ones. To keep the
-                         * gain fading consistent, interpolate between the old
-                         * and new target gains given how much of the fade time
-                         * this mix handles.
-                         */
-                        gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
-                                    minf(1.0f, (ALfloat)fademix/Counter));
-                        hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
-                        hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
-                        hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
-                        hrtfparams.Gain = 0.0f;
-                        hrtfparams.GainStep = gain / (ALfloat)fademix;
-
-                        MixHrtfBlendSamples(
-                            voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
-                            samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
-                            &hrtfparams, &parms->Hrtf.State, fademix
-                        );
-                        /* Update the old parameters with the result. */
-                        parms->Hrtf.Old = parms->Hrtf.Target;
-                        if(fademix < Counter)
-                            parms->Hrtf.Old.Gain = hrtfparams.Gain;
-                    }
-
-                    if(fademix < DstBufferSize)
-                    {
-                        ALsizei todo = DstBufferSize - fademix;
-                        ALfloat gain = parms->Hrtf.Target.Gain;
-
-                        /* Interpolate the target gain if the gain fading lasts
-                         * longer than this mix.
-                         */
-                        if(Counter > DstBufferSize)
-                            gain = lerp(parms->Hrtf.Old.Gain, gain,
-                                        (ALfloat)todo/(Counter-fademix));
-
-                        hrtfparams.Coeffs = SAFE_CONST(ALfloat2*,parms->Hrtf.Target.Coeffs);
-                        hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
-                        hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
-                        hrtfparams.Gain = parms->Hrtf.Old.Gain;
-                        hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
-                        MixHrtfSamples(
-                            voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
-                            samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
-                            &hrtfparams, &parms->Hrtf.State, todo
-                        );
-                        /* Store the interpolated gain or the final target gain
-                         * depending if the fade is done.
-                         */
-                        if(DstBufferSize < Counter)
-                            parms->Hrtf.Old.Gain = gain;
-                        else
-                            parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
-                    }
-                }
-            }
-
-            for(send = 0;send < Device->NumAuxSends;send++)
-            {
-                SendParams *parms = &voice->Send[send].Params[chan];
-                const ALfloat *samples;
-
-                if(!voice->Send[send].Buffer)
-                    continue;
-
-                samples = DoFilters(
-                    &parms->LowPass, &parms->HighPass, Device->FilteredData,
-                    ResampledData, DstBufferSize, voice->Send[send].FilterType
-                );
-
-                if(!Counter)
-                    memcpy(parms->Gains.Current, parms->Gains.Target,
-                           sizeof(parms->Gains.Current));
-                MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
-                    parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
-                );
-            }
-        }
-        /* Update positions */
-        DataPosFrac += increment*DstBufferSize;
-        DataPosInt  += DataPosFrac>>FRACTIONBITS;
-        DataPosFrac &= FRACTIONMASK;
-
-        OutPos += DstBufferSize;
-        voice->Offset += DstBufferSize;
-        Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
-        firstpass = false;
-
-        /* Handle looping sources */
-        while(1)
-        {
-            const ALbuffer *ALBuffer;
-            ALsizei DataSize = 0;
-            ALsizei LoopStart = 0;
-            ALsizei LoopEnd = 0;
-
-            if((ALBuffer=BufferListItem->buffer) != NULL)
-            {
-                DataSize = ALBuffer->SampleLen;
-                LoopStart = ALBuffer->LoopStart;
-                LoopEnd = ALBuffer->LoopEnd;
-                if(LoopEnd > DataPosInt)
-                    break;
-            }
-
-            if(BufferLoopItem && Source->SourceType == AL_STATIC)
-            {
-                assert(LoopEnd > LoopStart);
-                DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
-                break;
-            }
-
-            if(DataSize > DataPosInt)
-                break;
-
-            BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
-            if(!BufferListItem)
-            {
-                BufferListItem = BufferLoopItem;
-                if(!BufferListItem)
-                {
-                    isplaying = false;
-                    DataPosInt = 0;
-                    DataPosFrac = 0;
-                    break;
-                }
-            }
-
-            DataPosInt -= DataSize;
-        }
-    } while(isplaying && OutPos < SamplesToDo);
-
-    voice->Flags |= VOICE_IS_FADING;
-
-    /* Update source info */
-    ATOMIC_STORE(&voice->position,          DataPosInt, almemory_order_relaxed);
-    ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
-    ATOMIC_STORE(&voice->current_buffer,    BufferListItem, almemory_order_release);
-    return isplaying;
-}

Alc/mixer_c.c

@@ -1,208 +0,0 @@
-#include "config.h"
-
-#include <assert.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "alSource.h"
-#include "alAuxEffectSlot.h"
-
-
-static inline ALfloat point32(const ALfloat *restrict vals, ALsizei UNUSED(frac))
-{ return vals[0]; }
-static inline ALfloat lerp32(const ALfloat *restrict vals, ALsizei frac)
-{ return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); }
-static inline ALfloat fir4_32(const ALfloat *restrict vals, ALsizei frac)
-{ return resample_fir4(vals[-1], vals[0], vals[1], vals[2], frac); }
-
-
-const ALfloat *Resample_copy32_C(const InterpState* UNUSED(state),
-  const ALfloat *restrict src, ALsizei UNUSED(frac), ALint UNUSED(increment),
-  ALfloat *restrict dst, ALsizei numsamples)
-{
-#if defined(HAVE_SSE) || defined(HAVE_NEON)
-    /* Avoid copying the source data if it's aligned like the destination. */
-    if((((intptr_t)src)&15) == (((intptr_t)dst)&15))
-        return src;
-#endif
-    memcpy(dst, src, numsamples*sizeof(ALfloat));
-    return dst;
-}
-
-#define DECL_TEMPLATE(Sampler)                                                \
-const ALfloat *Resample_##Sampler##_C(const InterpState* UNUSED(state),       \
-  const ALfloat *restrict src, ALsizei frac, ALint increment,                 \
-  ALfloat *restrict dst, ALsizei numsamples)                                  \
-{                                                                             \
-    ALsizei i;                                                                \
-    for(i = 0;i < numsamples;i++)                                             \
-    {                                                                         \
-        dst[i] = Sampler(src, frac);                                          \
-                                                                              \
-        frac += increment;                                                    \
-        src  += frac>>FRACTIONBITS;                                           \
-        frac &= FRACTIONMASK;                                                 \
-    }                                                                         \
-    return dst;                                                               \
-}
-
-DECL_TEMPLATE(point32)
-DECL_TEMPLATE(lerp32)
-DECL_TEMPLATE(fir4_32)
-
-#undef DECL_TEMPLATE
-
-const ALfloat *Resample_bsinc32_C(const InterpState *state, const ALfloat *restrict src,
-                                  ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                  ALsizei dstlen)
-{
-    const ALfloat *fil, *scd, *phd, *spd;
-    const ALfloat sf = state->bsinc.sf;
-    const ALsizei m = state->bsinc.m;
-    ALsizei j_f, pi, i;
-    ALfloat pf, r;
-
-    src += state->bsinc.l;
-    for(i = 0;i < dstlen;i++)
-    {
-        // Calculate the phase index and factor.
-#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
-        pi = frac >> FRAC_PHASE_BITDIFF;
-        pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
-#undef FRAC_PHASE_BITDIFF
-
-        fil = ASSUME_ALIGNED(state->bsinc.coeffs[pi].filter, 16);
-        scd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].scDelta, 16);
-        phd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].phDelta, 16);
-        spd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].spDelta, 16);
-
-        // Apply the scale and phase interpolated filter.
-        r = 0.0f;
-        for(j_f = 0;j_f < m;j_f++)
-            r += (fil[j_f] + sf*scd[j_f] + pf*(phd[j_f] + sf*spd[j_f])) * src[j_f];
-        dst[i] = r;
-
-        frac += increment;
-        src  += frac>>FRACTIONBITS;
-        frac &= FRACTIONMASK;
-    }
-    return dst;
-}
-
-
-void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples)
-{
-    ALsizei i;
-    if(numsamples > 1)
-    {
-        dst[0] = filter->b0 * src[0] +
-                 filter->b1 * filter->x[0] +
-                 filter->b2 * filter->x[1] -
-                 filter->a1 * filter->y[0] -
-                 filter->a2 * filter->y[1];
-        dst[1] = filter->b0 * src[1] +
-                 filter->b1 * src[0] +
-                 filter->b2 * filter->x[0] -
-                 filter->a1 * dst[0] -
-                 filter->a2 * filter->y[0];
-        for(i = 2;i < numsamples;i++)
-            dst[i] = filter->b0 * src[i] +
-                     filter->b1 * src[i-1] +
-                     filter->b2 * src[i-2] -
-                     filter->a1 * dst[i-1] -
-                     filter->a2 * dst[i-2];
-        filter->x[0] = src[i-1];
-        filter->x[1] = src[i-2];
-        filter->y[0] = dst[i-1];
-        filter->y[1] = dst[i-2];
-    }
-    else if(numsamples == 1)
-    {
-        dst[0] = filter->b0 * src[0] +
-                 filter->b1 * filter->x[0] +
-                 filter->b2 * filter->x[1] -
-                 filter->a1 * filter->y[0] -
-                 filter->a2 * filter->y[1];
-        filter->x[1] = filter->x[0];
-        filter->x[0] = src[0];
-        filter->y[1] = filter->y[0];
-        filter->y[0] = dst[0];
-    }
-}
-
-
-static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
-                               const ALsizei IrSize,
-                               const ALfloat (*restrict Coeffs)[2],
-                               ALfloat left, ALfloat right)
-{
-    ALsizei c;
-    for(c = 0;c < IrSize;c++)
-    {
-        const ALsizei off = (Offset+c)&HRIR_MASK;
-        Values[off][0] += Coeffs[c][0] * left;
-        Values[off][1] += Coeffs[c][1] * right;
-    }
-}
-
-#define MixHrtf MixHrtf_C
-#define MixHrtfBlend MixHrtfBlend_C
-#define MixDirectHrtf MixDirectHrtf_C
-#include "mixer_inc.c"
-#undef MixHrtf
-
-
-void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
-           ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
-           ALsizei BufferSize)
-{
-    ALfloat gain, delta, step;
-    ALsizei c;
-
-    delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
-
-    for(c = 0;c < OutChans;c++)
-    {
-        ALsizei pos = 0;
-        gain = CurrentGains[c];
-        step = (TargetGains[c] - gain) * delta;
-        if(fabsf(step) > FLT_EPSILON)
-        {
-            ALsizei minsize = mini(BufferSize, Counter);
-            for(;pos < minsize;pos++)
-            {
-                OutBuffer[c][OutPos+pos] += data[pos]*gain;
-                gain += step;
-            }
-            if(pos == Counter)
-                gain = TargetGains[c];
-            CurrentGains[c] = gain;
-        }
-
-        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-            continue;
-        for(;pos < BufferSize;pos++)
-            OutBuffer[c][OutPos+pos] += data[pos]*gain;
-    }
-}
-
-/* Basically the inverse of the above. Rather than one input going to multiple
- * outputs (each with its own gain), it's multiple inputs (each with its own
- * gain) going to one output. This applies one row (vs one column) of a matrix
- * transform. And as the matrices are more or less static once set up, no
- * stepping is necessary.
- */
-void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
-{
-    ALsizei c, i;
-
-    for(c = 0;c < InChans;c++)
-    {
-        ALfloat gain = Gains[c];
-        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-            continue;
-
-        for(i = 0;i < BufferSize;i++)
-            OutBuffer[i] += data[c][InPos+i] * gain;
-    }
-}

Alc/mixer_defs.h

@@ -1,129 +0,0 @@
-#ifndef MIXER_DEFS_H
-#define MIXER_DEFS_H
-
-#include "AL/alc.h"
-#include "AL/al.h"
-#include "alMain.h"
-#include "alu.h"
-
-struct MixGains;
-
-struct MixHrtfParams;
-struct HrtfState;
-
-/* C resamplers */
-const ALfloat *Resample_copy32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
-const ALfloat *Resample_point32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
-const ALfloat *Resample_lerp32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
-const ALfloat *Resample_fir4_32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
-const ALfloat *Resample_bsinc32_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
-
-
-/* C mixers */
-void MixHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-               const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-               const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
-               struct HrtfState *hrtfstate, ALsizei BufferSize);
-void MixHrtfBlend_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                    const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                    const ALsizei IrSize, const HrtfParams *oldparams,
-                    MixHrtfParams *newparams, HrtfState *hrtfstate,
-                    ALsizei BufferSize);
-void MixDirectHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                     const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
-                     const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
-                     ALsizei BufferSize);
-void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
-           ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
-           ALsizei BufferSize);
-void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains,
-              const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
-              ALsizei InPos, ALsizei BufferSize);
-
-/* SSE mixers */
-void MixHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                 const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                 const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
-                 struct HrtfState *hrtfstate, ALsizei BufferSize);
-void MixHrtfBlend_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                      const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                      const ALsizei IrSize, const HrtfParams *oldparams,
-                      MixHrtfParams *newparams, HrtfState *hrtfstate,
-                      ALsizei BufferSize);
-void MixDirectHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                       const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
-                       const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
-                       ALsizei BufferSize);
-void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
-             ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
-             ALsizei BufferSize);
-void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains,
-                const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
-                ALsizei InPos, ALsizei BufferSize);
-
-/* SSE resamplers */
-inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALint *restrict pos_arr, ALsizei size)
-{
-    ALsizei i;
-
-    pos_arr[0] = 0;
-    frac_arr[0] = frac;
-    for(i = 1;i < size;i++)
-    {
-        ALint frac_tmp = frac_arr[i-1] + increment;
-        pos_arr[i] = pos_arr[i-1] + (frac_tmp>>FRACTIONBITS);
-        frac_arr[i] = frac_tmp&FRACTIONMASK;
-    }
-}
-
-const ALfloat *Resample_lerp32_SSE2(const InterpState *state, const ALfloat *restrict src,
-                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                    ALsizei numsamples);
-const ALfloat *Resample_lerp32_SSE41(const InterpState *state, const ALfloat *restrict src,
-                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                     ALsizei numsamples);
-
-const ALfloat *Resample_fir4_32_SSE3(const InterpState *state, const ALfloat *restrict src,
-                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                     ALsizei numsamples);
-const ALfloat *Resample_fir4_32_SSE41(const InterpState *state, const ALfloat *restrict src,
-                                      ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                      ALsizei numsamples);
-
-const ALfloat *Resample_bsinc32_SSE(const InterpState *state, const ALfloat *restrict src,
-                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                    ALsizei dstlen);
-
-/* Neon mixers */
-void MixHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                  const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                  const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
-                  struct HrtfState *hrtfstate, ALsizei BufferSize);
-void MixHrtfBlend_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                       const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                       const ALsizei IrSize, const HrtfParams *oldparams,
-                       MixHrtfParams *newparams, HrtfState *hrtfstate,
-                       ALsizei BufferSize);
-void MixDirectHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                        const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
-                        const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
-                        ALsizei BufferSize);
-void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
-              ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
-              ALsizei BufferSize);
-void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains,
-                 const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
-                 ALsizei InPos, ALsizei BufferSize);
-
-/* Neon resamplers */
-const ALfloat *Resample_lerp32_Neon(const InterpState *state, const ALfloat *restrict src,
-                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                    ALsizei numsamples);
-const ALfloat *Resample_fir4_32_Neon(const InterpState *state, const ALfloat *restrict src,
-                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                     ALsizei numsamples);
-const ALfloat *Resample_bsinc32_Neon(const InterpState *state, const ALfloat *restrict src,
-                                     ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                     ALsizei dstlen);
-
-#endif /* MIXER_DEFS_H */

Alc/mixer_inc.c

@@ -1,114 +0,0 @@
-#include "config.h"
-
-#include "alMain.h"
-#include "alSource.h"
-
-#include "hrtf.h"
-#include "mixer_defs.h"
-#include "align.h"
-#include "alu.h"
-
-
-static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
-                               const ALsizei irSize,
-                               const ALfloat (*restrict Coeffs)[2],
-                               ALfloat left, ALfloat right);
-
-
-void MixHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-             const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-             const ALsizei IrSize, MixHrtfParams *hrtfparams, HrtfState *hrtfstate,
-             ALsizei BufferSize)
-{
-    const ALfloat (*Coeffs)[2] = ASSUME_ALIGNED(hrtfparams->Coeffs, 16);
-    const ALsizei Delay[2] = { hrtfparams->Delay[0], hrtfparams->Delay[1] };
-    ALfloat gainstep = hrtfparams->GainStep;
-    ALfloat gain = hrtfparams->Gain;
-    ALfloat left, right;
-    ALsizei i;
-
-    LeftOut  += OutPos;
-    RightOut += OutPos;
-    for(i = 0;i < BufferSize;i++)
-    {
-        hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++);
-        left = hrtfstate->History[(Offset-Delay[0])&HRTF_HISTORY_MASK]*gain;
-        right = hrtfstate->History[(Offset-Delay[1])&HRTF_HISTORY_MASK]*gain;
-
-        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f;
-        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f;
-
-        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, Coeffs, left, right);
-        *(LeftOut++)  += hrtfstate->Values[Offset&HRIR_MASK][0];
-        *(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1];
-
-        gain += gainstep;
-        Offset++;
-    }
-    hrtfparams->Gain = gain;
-}
-
-void MixHrtfBlend(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                  const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                  const ALsizei IrSize, const HrtfParams *oldparams,
-                  MixHrtfParams *newparams, HrtfState *hrtfstate,
-                  ALsizei BufferSize)
-{
-    const ALfloat (*OldCoeffs)[2] = ASSUME_ALIGNED(oldparams->Coeffs, 16);
-    const ALsizei OldDelay[2] = { oldparams->Delay[0], oldparams->Delay[1] };
-    ALfloat oldGain = oldparams->Gain;
-    ALfloat oldGainStep = -oldGain / (ALfloat)BufferSize;
-    const ALfloat (*NewCoeffs)[2] = ASSUME_ALIGNED(newparams->Coeffs, 16);
-    const ALsizei NewDelay[2] = { newparams->Delay[0], newparams->Delay[1] };
-    ALfloat newGain = newparams->Gain;
-    ALfloat newGainStep = newparams->GainStep;
-    ALfloat left, right;
-    ALsizei i;
-
-    LeftOut  += OutPos;
-    RightOut += OutPos;
-    for(i = 0;i < BufferSize;i++)
-    {
-        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f;
-        hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f;
-
-        hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++);
-
-        left = hrtfstate->History[(Offset-OldDelay[0])&HRTF_HISTORY_MASK]*oldGain;
-        right = hrtfstate->History[(Offset-OldDelay[1])&HRTF_HISTORY_MASK]*oldGain;
-        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, OldCoeffs, left, right);
-
-        left = hrtfstate->History[(Offset-NewDelay[0])&HRTF_HISTORY_MASK]*newGain;
-        right = hrtfstate->History[(Offset-NewDelay[1])&HRTF_HISTORY_MASK]*newGain;
-        ApplyCoeffs(Offset, hrtfstate->Values, IrSize, NewCoeffs, left, right);
-
-        *(LeftOut++)  += hrtfstate->Values[Offset&HRIR_MASK][0];
-        *(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1];
-
-        oldGain += oldGainStep;
-        newGain += newGainStep;
-        Offset++;
-    }
-    newparams->Gain = newGain;
-}
-
-void MixDirectHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                   const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
-                   const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
-                   ALsizei BufferSize)
-{
-    ALfloat insample;
-    ALsizei i;
-
-    for(i = 0;i < BufferSize;i++)
-    {
-        Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f;
-        Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f;
-        Offset++;
-
-        insample = *(data++);
-        ApplyCoeffs(Offset, Values, IrSize, Coeffs, insample, insample);
-        *(LeftOut++)  += Values[Offset&HRIR_MASK][0];
-        *(RightOut++) += Values[Offset&HRIR_MASK][1];
-    }
-}

Alc/mixer_neon.c

@@ -1,331 +0,0 @@
-#include "config.h"
-
-#include <arm_neon.h>
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alu.h"
-#include "hrtf.h"
-#include "mixer_defs.h"
-
-
-const ALfloat *Resample_lerp32_Neon(const InterpState* UNUSED(state),
-  const ALfloat *restrict src, ALsizei frac, ALint increment,
-  ALfloat *restrict dst, ALsizei numsamples)
-{
-    const int32x4_t increment4 = vdupq_n_s32(increment*4);
-    const float32x4_t fracOne4 = vdupq_n_f32(1.0f/FRACTIONONE);
-    const int32x4_t fracMask4 = vdupq_n_s32(FRACTIONMASK);
-    alignas(16) ALint pos_[4];
-    alignas(16) ALsizei frac_[4];
-    int32x4_t pos4;
-    int32x4_t frac4;
-    ALsizei i;
-
-    InitiatePositionArrays(frac, increment, frac_, pos_, 4);
-
-    frac4 = vld1q_s32(frac_);
-    pos4 = vld1q_s32(pos_);
-
-    for(i = 0;numsamples-i > 3;i += 4)
-    {
-        const float32x4_t val1 = (float32x4_t){src[pos_[0]], src[pos_[1]], src[pos_[2]], src[pos_[3]]};
-        const float32x4_t val2 = (float32x4_t){src[pos_[0]+1], src[pos_[1]+1], src[pos_[2]+1], src[pos_[3]+1]};
-
-        /* val1 + (val2-val1)*mu */
-        const float32x4_t r0 = vsubq_f32(val2, val1);
-        const float32x4_t mu = vmulq_f32(vcvtq_f32_s32(frac4), fracOne4);
-        const float32x4_t out = vmlaq_f32(val1, mu, r0);
-
-        vst1q_f32(&dst[i], out);
-
-        frac4 = vaddq_s32(frac4, increment4);
-        pos4 = vaddq_s32(pos4, vshrq_n_s32(frac4, FRACTIONBITS));
-        frac4 = vandq_s32(frac4, fracMask4);
-
-        vst1q_s32(pos_, pos4);
-    }
-
-    if(i < numsamples)
-    {
-        /* NOTE: These four elements represent the position *after* the last
-         * four samples, so the lowest element is the next position to
-         * resample.
-         */
-        ALint pos = pos_[0];
-        frac = vgetq_lane_s32(frac4, 0);
-        do {
-            dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
-
-            frac += increment;
-            pos  += frac>>FRACTIONBITS;
-            frac &= FRACTIONMASK;
-        } while(++i < numsamples);
-    }
-    return dst;
-}
-
-const ALfloat *Resample_fir4_32_Neon(const InterpState* UNUSED(state),
-  const ALfloat *restrict src, ALsizei frac, ALint increment,
-  ALfloat *restrict dst, ALsizei numsamples)
-{
-    const int32x4_t increment4 = vdupq_n_s32(increment*4);
-    const int32x4_t fracMask4 = vdupq_n_s32(FRACTIONMASK);
-    alignas(16) ALint pos_[4];
-    alignas(16) ALsizei frac_[4];
-    int32x4_t pos4;
-    int32x4_t frac4;
-    ALsizei i;
-
-    InitiatePositionArrays(frac, increment, frac_, pos_, 4);
-
-    frac4 = vld1q_s32(frac_);
-    pos4 = vld1q_s32(pos_);
-
-    --src;
-    for(i = 0;numsamples-i > 3;i += 4)
-    {
-        const float32x4_t val0 = vld1q_f32(&src[pos_[0]]);
-        const float32x4_t val1 = vld1q_f32(&src[pos_[1]]);
-        const float32x4_t val2 = vld1q_f32(&src[pos_[2]]);
-        const float32x4_t val3 = vld1q_f32(&src[pos_[3]]);
-        float32x4_t k0 = vld1q_f32(sinc4Tab[frac_[0]]);
-        float32x4_t k1 = vld1q_f32(sinc4Tab[frac_[1]]);
-        float32x4_t k2 = vld1q_f32(sinc4Tab[frac_[2]]);
-        float32x4_t k3 = vld1q_f32(sinc4Tab[frac_[3]]);
-        float32x4_t out;
-
-        k0 = vmulq_f32(k0, val0);
-        k1 = vmulq_f32(k1, val1);
-        k2 = vmulq_f32(k2, val2);
-        k3 = vmulq_f32(k3, val3);
-        k0 = vcombine_f32(vpadd_f32(vget_low_f32(k0), vget_high_f32(k0)),
-                          vpadd_f32(vget_low_f32(k1), vget_high_f32(k1)));
-        k2 = vcombine_f32(vpadd_f32(vget_low_f32(k2), vget_high_f32(k2)),
-                          vpadd_f32(vget_low_f32(k3), vget_high_f32(k3)));
-        out = vcombine_f32(vpadd_f32(vget_low_f32(k0), vget_high_f32(k0)),
-                           vpadd_f32(vget_low_f32(k2), vget_high_f32(k2)));
-
-        vst1q_f32(&dst[i], out);
-
-        frac4 = vaddq_s32(frac4, increment4);
-        pos4 = vaddq_s32(pos4, vshrq_n_s32(frac4, FRACTIONBITS));
-        frac4 = vandq_s32(frac4, fracMask4);
-
-        vst1q_s32(pos_, pos4);
-        vst1q_s32(frac_, frac4);
-    }
-
-    if(i < numsamples)
-    {
-        /* NOTE: These four elements represent the position *after* the last
-         * four samples, so the lowest element is the next position to
-         * resample.
-         */
-        ALint pos = pos_[0];
-        frac = frac_[0];
-        do {
-            dst[i] = resample_fir4(src[pos], src[pos+1], src[pos+2], src[pos+3], frac);
-
-            frac += increment;
-            pos  += frac>>FRACTIONBITS;
-            frac &= FRACTIONMASK;
-        } while(++i < numsamples);
-    }
-    return dst;
-}
-
-const ALfloat *Resample_bsinc32_Neon(const InterpState *state,
-  const ALfloat *restrict src, ALsizei frac, ALint increment,
-  ALfloat *restrict dst, ALsizei dstlen)
-{
-    const float32x4_t sf4 = vdupq_n_f32(state->bsinc.sf);
-    const ALsizei m = state->bsinc.m;
-    const ALfloat *fil, *scd, *phd, *spd;
-    ALsizei pi, i, j;
-    float32x4_t r4;
-    ALfloat pf;
-
-    src += state->bsinc.l;
-    for(i = 0;i < dstlen;i++)
-    {
-        // Calculate the phase index and factor.
-#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
-        pi = frac >> FRAC_PHASE_BITDIFF;
-        pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
-#undef FRAC_PHASE_BITDIFF
-
-        fil = ASSUME_ALIGNED(state->bsinc.coeffs[pi].filter, 16);
-        scd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].scDelta, 16);
-        phd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].phDelta, 16);
-        spd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].spDelta, 16);
-
-        // Apply the scale and phase interpolated filter.
-        r4 = vdupq_n_f32(0.0f);
-        {
-            const float32x4_t pf4 = vdupq_n_f32(pf);
-            for(j = 0;j < m;j+=4)
-            {
-                /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
-                const float32x4_t f4 = vmlaq_f32(vmlaq_f32(vld1q_f32(&fil[j]),
-                                                           sf4, vld1q_f32(&scd[j])),
-                    pf4, vmlaq_f32(vld1q_f32(&phd[j]),
-                        sf4, vld1q_f32(&spd[j])
-                    )
-                );
-                /* r += f*src */
-                r4 = vmlaq_f32(r4, f4, vld1q_f32(&src[j]));
-            }
-        }
-        r4 = vaddq_f32(r4, vcombine_f32(vrev64_f32(vget_high_f32(r4)),
-                                        vrev64_f32(vget_low_f32(r4))));
-        dst[i] = vget_lane_f32(vadd_f32(vget_low_f32(r4), vget_high_f32(r4)), 0);
-
-        frac += increment;
-        src  += frac>>FRACTIONBITS;
-        frac &= FRACTIONMASK;
-    }
-    return dst;
-}
-
-
-static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
-                               const ALsizei IrSize,
-                               const ALfloat (*restrict Coeffs)[2],
-                               ALfloat left, ALfloat right)
-{
-    ALsizei c;
-    float32x4_t leftright4;
-    {
-        float32x2_t leftright2 = vdup_n_f32(0.0);
-        leftright2 = vset_lane_f32(left, leftright2, 0);
-        leftright2 = vset_lane_f32(right, leftright2, 1);
-        leftright4 = vcombine_f32(leftright2, leftright2);
-    }
-    Values = ASSUME_ALIGNED(Values, 16);
-    Coeffs = ASSUME_ALIGNED(Coeffs, 16);
-    for(c = 0;c < IrSize;c += 2)
-    {
-        const ALsizei o0 = (Offset+c)&HRIR_MASK;
-        const ALsizei o1 = (o0+1)&HRIR_MASK;
-        float32x4_t vals = vcombine_f32(vld1_f32((float32_t*)&Values[o0][0]),
-                                        vld1_f32((float32_t*)&Values[o1][0]));
-        float32x4_t coefs = vld1q_f32((float32_t*)&Coeffs[c][0]);
-
-        vals = vmlaq_f32(vals, coefs, leftright4);
-
-        vst1_f32((float32_t*)&Values[o0][0], vget_low_f32(vals));
-        vst1_f32((float32_t*)&Values[o1][0], vget_high_f32(vals));
-    }
-}
-
-#define MixHrtf MixHrtf_Neon
-#define MixHrtfBlend MixHrtfBlend_Neon
-#define MixDirectHrtf MixDirectHrtf_Neon
-#include "mixer_inc.c"
-#undef MixHrtf
-
-
-void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
-              ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
-              ALsizei BufferSize)
-{
-    ALfloat gain, delta, step;
-    float32x4_t gain4;
-    ALsizei c;
-
-    data = ASSUME_ALIGNED(data, 16);
-    OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
-
-    delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
-
-    for(c = 0;c < OutChans;c++)
-    {
-        ALsizei pos = 0;
-        gain = CurrentGains[c];
-        step = (TargetGains[c] - gain) * delta;
-        if(fabsf(step) > FLT_EPSILON)
-        {
-            ALsizei minsize = mini(BufferSize, Counter);
-            /* Mix with applying gain steps in aligned multiples of 4. */
-            if(minsize-pos > 3)
-            {
-                float32x4_t step4;
-                gain4 = vsetq_lane_f32(gain, gain4, 0);
-                gain4 = vsetq_lane_f32(gain + step, gain4, 1);
-                gain4 = vsetq_lane_f32(gain + step + step, gain4, 2);
-                gain4 = vsetq_lane_f32(gain + step + step + step, gain4, 3);
-                step4 = vdupq_n_f32(step + step + step + step);
-                do {
-                    const float32x4_t val4 = vld1q_f32(&data[pos]);
-                    float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]);
-                    dry4 = vmlaq_f32(dry4, val4, gain4);
-                    gain4 = vaddq_f32(gain4, step4);
-                    vst1q_f32(&OutBuffer[c][OutPos+pos], dry4);
-                    pos += 4;
-                } while(minsize-pos > 3);
-                /* NOTE: gain4 now represents the next four gains after the
-                 * last four mixed samples, so the lowest element represents
-                 * the next gain to apply.
-                 */
-                gain = vgetq_lane_f32(gain4, 0);
-            }
-            /* Mix with applying left over gain steps that aren't aligned multiples of 4. */
-            for(;pos < minsize;pos++)
-            {
-                OutBuffer[c][OutPos+pos] += data[pos]*gain;
-                gain += step;
-            }
-            if(pos == Counter)
-                gain = TargetGains[c];
-            CurrentGains[c] = gain;
-
-            /* Mix until pos is aligned with 4 or the mix is done. */
-            minsize = mini(BufferSize, (pos+3)&~3);
-            for(;pos < minsize;pos++)
-                OutBuffer[c][OutPos+pos] += data[pos]*gain;
-        }
-
-        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-            continue;
-        gain4 = vdupq_n_f32(gain);
-        for(;BufferSize-pos > 3;pos += 4)
-        {
-            const float32x4_t val4 = vld1q_f32(&data[pos]);
-            float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]);
-            dry4 = vmlaq_f32(dry4, val4, gain4);
-            vst1q_f32(&OutBuffer[c][OutPos+pos], dry4);
-        }
-        for(;pos < BufferSize;pos++)
-            OutBuffer[c][OutPos+pos] += data[pos]*gain;
-    }
-}
-
-void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
-{
-    float32x4_t gain4;
-    ALsizei c;
-
-    data = ASSUME_ALIGNED(data, 16);
-    OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
-
-    for(c = 0;c < InChans;c++)
-    {
-        ALsizei pos = 0;
-        ALfloat gain = Gains[c];
-        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-            continue;
-
-        gain4 = vdupq_n_f32(gain);
-        for(;BufferSize-pos > 3;pos += 4)
-        {
-            const float32x4_t val4 = vld1q_f32(&data[c][InPos+pos]);
-            float32x4_t dry4 = vld1q_f32(&OutBuffer[pos]);
-            dry4 = vmlaq_f32(dry4, val4, gain4);
-            vst1q_f32(&OutBuffer[pos], dry4);
-        }
-        for(;pos < BufferSize;pos++)
-            OutBuffer[pos] += data[c][InPos+pos]*gain;
-    }
-}

Alc/mixer_sse.c

@@ -1,230 +0,0 @@
-#include "config.h"
-
-#include <xmmintrin.h>
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alu.h"
-
-#include "alSource.h"
-#include "alAuxEffectSlot.h"
-#include "mixer_defs.h"
-
-
-const ALfloat *Resample_bsinc32_SSE(const InterpState *state, const ALfloat *restrict src,
-                                    ALsizei frac, ALint increment, ALfloat *restrict dst,
-                                    ALsizei dstlen)
-{
-    const __m128 sf4 = _mm_set1_ps(state->bsinc.sf);
-    const ALsizei m = state->bsinc.m;
-    const ALfloat *fil, *scd, *phd, *spd;
-    ALsizei pi, i, j;
-    ALfloat pf;
-    __m128 r4;
-
-    src += state->bsinc.l;
-    for(i = 0;i < dstlen;i++)
-    {
-        // Calculate the phase index and factor.
-#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
-        pi = frac >> FRAC_PHASE_BITDIFF;
-        pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
-#undef FRAC_PHASE_BITDIFF
-
-        fil = ASSUME_ALIGNED(state->bsinc.coeffs[pi].filter, 16);
-        scd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].scDelta, 16);
-        phd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].phDelta, 16);
-        spd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].spDelta, 16);
-
-        // Apply the scale and phase interpolated filter.
-        r4 = _mm_setzero_ps();
-        {
-            const __m128 pf4 = _mm_set1_ps(pf);
-#define LD4(x) _mm_load_ps(x)
-#define ULD4(x) _mm_loadu_ps(x)
-#define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z))
-            for(j = 0;j < m;j+=4)
-            {
-                /* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
-                const __m128 f4 = MLA4(MLA4(LD4(&fil[j]), sf4, LD4(&scd[j])),
-                    pf4, MLA4(LD4(&phd[j]), sf4, LD4(&spd[j]))
-                );
-                /* r += f*src */
-                r4 = MLA4(r4, f4, ULD4(&src[j]));
-            }
-#undef MLA4
-#undef ULD4
-#undef LD4
-        }
-        r4 = _mm_add_ps(r4, _mm_shuffle_ps(r4, r4, _MM_SHUFFLE(0, 1, 2, 3)));
-        r4 = _mm_add_ps(r4, _mm_movehl_ps(r4, r4));
-        dst[i] = _mm_cvtss_f32(r4);
-
-        frac += increment;
-        src  += frac>>FRACTIONBITS;
-        frac &= FRACTIONMASK;
-    }
-    return dst;
-}
-
-
-static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
-                               const ALsizei IrSize,
-                               const ALfloat (*restrict Coeffs)[2],
-                               ALfloat left, ALfloat right)
-{
-    const __m128 lrlr = _mm_setr_ps(left, right, left, right);
-    __m128 vals = _mm_setzero_ps();
-    __m128 coeffs;
-    ALsizei i;
-
-    Values = ASSUME_ALIGNED(Values, 16);
-    Coeffs = ASSUME_ALIGNED(Coeffs, 16);
-    if((Offset&1))
-    {
-        const ALsizei o0 = Offset&HRIR_MASK;
-        const ALsizei o1 = (Offset+IrSize-1)&HRIR_MASK;
-        __m128 imp0, imp1;
-
-        coeffs = _mm_load_ps(&Coeffs[0][0]);
-        vals = _mm_loadl_pi(vals, (__m64*)&Values[o0][0]);
-        imp0 = _mm_mul_ps(lrlr, coeffs);
-        vals = _mm_add_ps(imp0, vals);
-        _mm_storel_pi((__m64*)&Values[o0][0], vals);
-        for(i = 1;i < IrSize-1;i += 2)
-        {
-            const ALsizei o2 = (Offset+i)&HRIR_MASK;
-
-            coeffs = _mm_load_ps(&Coeffs[i+1][0]);
-            vals = _mm_load_ps(&Values[o2][0]);
-            imp1 = _mm_mul_ps(lrlr, coeffs);
-            imp0 = _mm_shuffle_ps(imp0, imp1, _MM_SHUFFLE(1, 0, 3, 2));
-            vals = _mm_add_ps(imp0, vals);
-            _mm_store_ps(&Values[o2][0], vals);
-            imp0 = imp1;
-        }
-        vals = _mm_loadl_pi(vals, (__m64*)&Values[o1][0]);
-        imp0 = _mm_movehl_ps(imp0, imp0);
-        vals = _mm_add_ps(imp0, vals);
-        _mm_storel_pi((__m64*)&Values[o1][0], vals);
-    }
-    else
-    {
-        for(i = 0;i < IrSize;i += 2)
-        {
-            const ALsizei o = (Offset + i)&HRIR_MASK;
-
-            coeffs = _mm_load_ps(&Coeffs[i][0]);
-            vals = _mm_load_ps(&Values[o][0]);
-            vals = _mm_add_ps(vals, _mm_mul_ps(lrlr, coeffs));
-            _mm_store_ps(&Values[o][0], vals);
-        }
-    }
-}
-
-#define MixHrtf MixHrtf_SSE
-#define MixHrtfBlend MixHrtfBlend_SSE
-#define MixDirectHrtf MixDirectHrtf_SSE
-#include "mixer_inc.c"
-#undef MixHrtf
-
-
-void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
-             ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
-             ALsizei BufferSize)
-{
-    ALfloat gain, delta, step;
-    __m128 gain4;
-    ALsizei c;
-
-    delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
-
-    for(c = 0;c < OutChans;c++)
-    {
-        ALsizei pos = 0;
-        gain = CurrentGains[c];
-        step = (TargetGains[c] - gain) * delta;
-        if(fabsf(step) > FLT_EPSILON)
-        {
-            ALsizei minsize = mini(BufferSize, Counter);
-            /* Mix with applying gain steps in aligned multiples of 4. */
-            if(minsize-pos > 3)
-            {
-                __m128 step4;
-                gain4 = _mm_setr_ps(
-                    gain,
-                    gain + step,
-                    gain + step + step,
-                    gain + step + step + step
-                );
-                step4 = _mm_set1_ps(step + step + step + step);
-                do {
-                    const __m128 val4 = _mm_load_ps(&data[pos]);
-                    __m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]);
-                    dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
-                    gain4 = _mm_add_ps(gain4, step4);
-                    _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
-                    pos += 4;
-                } while(minsize-pos > 3);
-                /* NOTE: gain4 now represents the next four gains after the
-                 * last four mixed samples, so the lowest element represents
-                 * the next gain to apply.
-                 */
-                gain = _mm_cvtss_f32(gain4);
-            }
-            /* Mix with applying left over gain steps that aren't aligned multiples of 4. */
-            for(;pos < minsize;pos++)
-            {
-                OutBuffer[c][OutPos+pos] += data[pos]*gain;
-                gain += step;
-            }
-            if(pos == Counter)
-                gain = TargetGains[c];
-            CurrentGains[c] = gain;
-
-            /* Mix until pos is aligned with 4 or the mix is done. */
-            minsize = mini(BufferSize, (pos+3)&~3);
-            for(;pos < minsize;pos++)
-                OutBuffer[c][OutPos+pos] += data[pos]*gain;
-        }
-
-        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-            continue;
-        gain4 = _mm_set1_ps(gain);
-        for(;BufferSize-pos > 3;pos += 4)
-        {
-            const __m128 val4 = _mm_load_ps(&data[pos]);
-            __m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]);
-            dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
-            _mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
-        }
-        for(;pos < BufferSize;pos++)
-            OutBuffer[c][OutPos+pos] += data[pos]*gain;
-    }
-}
-
-void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
-{
-    __m128 gain4;
-    ALsizei c;
-
-    for(c = 0;c < InChans;c++)
-    {
-        ALsizei pos = 0;
-        ALfloat gain = Gains[c];
-        if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
-            continue;
-
-        gain4 = _mm_set1_ps(gain);
-        for(;BufferSize-pos > 3;pos += 4)
-        {
-            const __m128 val4 = _mm_load_ps(&data[c][InPos+pos]);
-            __m128 dry4 = _mm_load_ps(&OutBuffer[pos]);
-            dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
-            _mm_store_ps(&OutBuffer[pos], dry4);
-        }
-        for(;pos < BufferSize;pos++)
-            OutBuffer[pos] += data[c][InPos+pos]*gain;
-    }
-}

Alc/mixer_sse2.c

@@ -1,82 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <xmmintrin.h>
-#include <emmintrin.h>
-
-#include "alu.h"
-#include "mixer_defs.h"
-
-
-const ALfloat *Resample_lerp32_SSE2(const InterpState* UNUSED(state),
-  const ALfloat *restrict src, ALsizei frac, ALint increment,
-  ALfloat *restrict dst, ALsizei numsamples)
-{
-    const __m128i increment4 = _mm_set1_epi32(increment*4);
-    const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE);
-    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
-    union { alignas(16) ALint i[4]; float f[4]; } pos_;
-    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
-    __m128i frac4, pos4;
-    ALint pos;
-    ALsizei i;
-
-    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
-
-    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
-    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
-
-    for(i = 0;numsamples-i > 3;i += 4)
-    {
-        const __m128 val1 = _mm_setr_ps(src[pos_.i[0]], src[pos_.i[1]], src[pos_.i[2]], src[pos_.i[3]]);
-        const __m128 val2 = _mm_setr_ps(src[pos_.i[0]+1], src[pos_.i[1]+1], src[pos_.i[2]+1], src[pos_.i[3]+1]);
-
-        /* val1 + (val2-val1)*mu */
-        const __m128 r0 = _mm_sub_ps(val2, val1);
-        const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4);
-        const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0));
-
-        _mm_store_ps(&dst[i], out);
-
-        frac4 = _mm_add_epi32(frac4, increment4);
-        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
-        frac4 = _mm_and_si128(frac4, fracMask4);
-
-        _mm_store_ps(pos_.f, _mm_castsi128_ps(pos4));
-    }
-
-    /* NOTE: These four elements represent the position *after* the last four
-     * samples, so the lowest element is the next position to resample.
-     */
-    pos = pos_.i[0];
-    frac = _mm_cvtsi128_si32(frac4);
-
-    for(;i < numsamples;i++)
-    {
-        dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
-
-        frac += increment;
-        pos  += frac>>FRACTIONBITS;
-        frac &= FRACTIONMASK;
-    }
-    return dst;
-}

Alc/mixer_sse3.c

@@ -1,97 +0,0 @@
-/**
- * OpenAL cross platform audio library, SSE3 mixer functions
- *
- * Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
- * Copyright (C) 2015 by Chris Robinson <chris.kcat@gmail.com>.
- *
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <xmmintrin.h>
-#include <emmintrin.h>
-#include <pmmintrin.h>
-
-#include "alu.h"
-#include "mixer_defs.h"
-
-
-const ALfloat *Resample_fir4_32_SSE3(const InterpState* UNUSED(state),
-  const ALfloat *restrict src, ALsizei frac, ALint increment,
-  ALfloat *restrict dst, ALsizei numsamples)
-{
-    const __m128i increment4 = _mm_set1_epi32(increment*4);
-    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
-    union { alignas(16) ALint i[4]; float f[4]; } pos_;
-    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
-    __m128i frac4, pos4;
-    ALint pos;
-    ALsizei i;
-
-    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
-
-    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
-    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
-
-    --src;
-    for(i = 0;numsamples-i > 3;i += 4)
-    {
-        const __m128 val0 = _mm_loadu_ps(&src[pos_.i[0]]);
-        const __m128 val1 = _mm_loadu_ps(&src[pos_.i[1]]);
-        const __m128 val2 = _mm_loadu_ps(&src[pos_.i[2]]);
-        const __m128 val3 = _mm_loadu_ps(&src[pos_.i[3]]);
-        __m128 k0 = _mm_load_ps(sinc4Tab[frac_.i[0]]);
-        __m128 k1 = _mm_load_ps(sinc4Tab[frac_.i[1]]);
-        __m128 k2 = _mm_load_ps(sinc4Tab[frac_.i[2]]);
-        __m128 k3 = _mm_load_ps(sinc4Tab[frac_.i[3]]);
-        __m128 out;
-
-        k0 = _mm_mul_ps(k0, val0);
-        k1 = _mm_mul_ps(k1, val1);
-        k2 = _mm_mul_ps(k2, val2);
-        k3 = _mm_mul_ps(k3, val3);
-        k0 = _mm_hadd_ps(k0, k1);
-        k2 = _mm_hadd_ps(k2, k3);
-        out = _mm_hadd_ps(k0, k2);
-
-        _mm_store_ps(&dst[i], out);
-
-        frac4 = _mm_add_epi32(frac4, increment4);
-        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
-        frac4 = _mm_and_si128(frac4, fracMask4);
-
-        _mm_store_ps(pos_.f, _mm_castsi128_ps(pos4));
-        _mm_store_ps(frac_.f, _mm_castsi128_ps(frac4));
-    }
-
-    /* NOTE: These four elements represent the position *after* the last four
-     * samples, so the lowest element is the next position to resample.
-     */
-    pos = pos_.i[0];
-    frac = frac_.i[0];
-
-    for(;i < numsamples;i++)
-    {
-        dst[i] = resample_fir4(src[pos], src[pos+1], src[pos+2], src[pos+3], frac);
-
-        frac += increment;
-        pos  += frac>>FRACTIONBITS;
-        frac &= FRACTIONMASK;
-    }
-    return dst;
-}

Alc/mixer_sse41.c

@@ -1,154 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <xmmintrin.h>
-#include <emmintrin.h>
-#include <smmintrin.h>
-
-#include "alu.h"
-#include "mixer_defs.h"
-
-
-const ALfloat *Resample_lerp32_SSE41(const InterpState* UNUSED(state),
-  const ALfloat *restrict src, ALsizei frac, ALint increment,
-  ALfloat *restrict dst, ALsizei numsamples)
-{
-    const __m128i increment4 = _mm_set1_epi32(increment*4);
-    const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE);
-    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
-    union { alignas(16) ALint i[4]; float f[4]; } pos_;
-    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
-    __m128i frac4, pos4;
-    ALint pos;
-    ALsizei i;
-
-    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
-
-    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
-    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
-
-    for(i = 0;numsamples-i > 3;i += 4)
-    {
-        const __m128 val1 = _mm_setr_ps(src[pos_.i[0]], src[pos_.i[1]], src[pos_.i[2]], src[pos_.i[3]]);
-        const __m128 val2 = _mm_setr_ps(src[pos_.i[0]+1], src[pos_.i[1]+1], src[pos_.i[2]+1], src[pos_.i[3]+1]);
-
-        /* val1 + (val2-val1)*mu */
-        const __m128 r0 = _mm_sub_ps(val2, val1);
-        const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4);
-        const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0));
-
-        _mm_store_ps(&dst[i], out);
-
-        frac4 = _mm_add_epi32(frac4, increment4);
-        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
-        frac4 = _mm_and_si128(frac4, fracMask4);
-
-        pos_.i[0] = _mm_extract_epi32(pos4, 0);
-        pos_.i[1] = _mm_extract_epi32(pos4, 1);
-        pos_.i[2] = _mm_extract_epi32(pos4, 2);
-        pos_.i[3] = _mm_extract_epi32(pos4, 3);
-    }
-
-    /* NOTE: These four elements represent the position *after* the last four
-     * samples, so the lowest element is the next position to resample.
-     */
-    pos = pos_.i[0];
-    frac = _mm_cvtsi128_si32(frac4);
-
-    for(;i < numsamples;i++)
-    {
-        dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
-
-        frac += increment;
-        pos  += frac>>FRACTIONBITS;
-        frac &= FRACTIONMASK;
-    }
-    return dst;
-}
-
-const ALfloat *Resample_fir4_32_SSE41(const InterpState* UNUSED(state),
-  const ALfloat *restrict src, ALsizei frac, ALint increment,
-  ALfloat *restrict dst, ALsizei numsamples)
-{
-    const __m128i increment4 = _mm_set1_epi32(increment*4);
-    const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
-    union { alignas(16) ALint i[4]; float f[4]; } pos_;
-    union { alignas(16) ALsizei i[4]; float f[4]; } frac_;
-    __m128i frac4, pos4;
-    ALint pos;
-    ALsizei i;
-
-    InitiatePositionArrays(frac, increment, frac_.i, pos_.i, 4);
-
-    frac4 = _mm_castps_si128(_mm_load_ps(frac_.f));
-    pos4 = _mm_castps_si128(_mm_load_ps(pos_.f));
-
-    --src;
-    for(i = 0;numsamples-i > 3;i += 4)
-    {
-        const __m128 val0 = _mm_loadu_ps(&src[pos_.i[0]]);
-        const __m128 val1 = _mm_loadu_ps(&src[pos_.i[1]]);
-        const __m128 val2 = _mm_loadu_ps(&src[pos_.i[2]]);
-        const __m128 val3 = _mm_loadu_ps(&src[pos_.i[3]]);
-        __m128 k0 = _mm_load_ps(sinc4Tab[frac_.i[0]]);
-        __m128 k1 = _mm_load_ps(sinc4Tab[frac_.i[1]]);
-        __m128 k2 = _mm_load_ps(sinc4Tab[frac_.i[2]]);
-        __m128 k3 = _mm_load_ps(sinc4Tab[frac_.i[3]]);
-        __m128 out;
-
-        k0 = _mm_mul_ps(k0, val0);
-        k1 = _mm_mul_ps(k1, val1);
-        k2 = _mm_mul_ps(k2, val2);
-        k3 = _mm_mul_ps(k3, val3);
-        k0 = _mm_hadd_ps(k0, k1);
-        k2 = _mm_hadd_ps(k2, k3);
-        out = _mm_hadd_ps(k0, k2);
-
-        _mm_store_ps(&dst[i], out);
-
-        frac4 = _mm_add_epi32(frac4, increment4);
-        pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
-        frac4 = _mm_and_si128(frac4, fracMask4);
-
-        pos_.i[0] = _mm_extract_epi32(pos4, 0);
-        pos_.i[1] = _mm_extract_epi32(pos4, 1);
-        pos_.i[2] = _mm_extract_epi32(pos4, 2);
-        pos_.i[3] = _mm_extract_epi32(pos4, 3);
-        frac_.i[0] = _mm_extract_epi32(frac4, 0);
-        frac_.i[1] = _mm_extract_epi32(frac4, 1);
-        frac_.i[2] = _mm_extract_epi32(frac4, 2);
-        frac_.i[3] = _mm_extract_epi32(frac4, 3);
-    }
-
-    pos = pos_.i[0];
-    frac = frac_.i[0];
-
-    for(;i < numsamples;i++)
-    {
-        dst[i] = resample_fir4(src[pos], src[pos+1], src[pos+2], src[pos+3], frac);
-
-        frac += increment;
-        pos  += frac>>FRACTIONBITS;
-        frac &= FRACTIONMASK;
-    }
-    return dst;
-}

Alc/nfcfilter.c

@@ -1,418 +0,0 @@
-
-#include "config.h"
-
-#include "nfcfilter.h"
-
-#include "alu.h"
-
-
-/* Near-field control filters are the basis for handling the near-field effect.
- * The near-field effect is a bass-boost present in the directional components
- * of a recorded signal, created as a result of the wavefront curvature (itself
- * a function of sound distance). Proper reproduction dictates this be
- * compensated for using a bass-cut given the playback speaker distance, to
- * avoid excessive bass in the playback.
- *
- * For real-time rendered audio, emulating the near-field effect based on the
- * sound source's distance, and subsequently compensating for it at output
- * based on the speaker distances, can create a more realistic perception of
- * sound distance beyond a simple 1/r attenuation.
- *
- * These filters do just that. Each one applies a low-shelf filter, created as
- * the combination of a bass-boost for a given sound source distance (near-
- * field emulation) along with a bass-cut for a given control/speaker distance
- * (near-field compensation).
- *
- * Note that it is necessary to apply a cut along with the boost, since the
- * boost alone is unstable in higher-order ambisonics as it causes an infinite
- * DC gain (even first-order ambisonics requires there to be no DC offset for
- * the boost to work). Consequently, ambisonics requires a control parameter to
- * be used to avoid an unstable boost-only filter. NFC-HOA defines this control
- * as a reference delay, calculated with:
- *
- * reference_delay = control_distance / speed_of_sound
- *
- * This means w0 (for input) or w1 (for output) should be set to:
- *
- * wN = 1 / (reference_delay * sample_rate)
- *
- * when dealing with NFC-HOA content. For FOA input content, which does not
- * specify a reference_delay variable, w0 should be set to 0 to apply only
- * near-field compensation for output. It's important that w1 be a finite,
- * positive, non-0 value or else the bass-boost will become unstable again.
- * Also, w0 should not be too large compared to w1, to avoid excessively loud
- * low frequencies.
- */
-
-static const float B[4][3] = {
-    {    0.0f                             },
-    {    1.0f                             },
-    {    3.0f,     3.0f                   },
-    { 3.6778f,  6.4595f, 2.3222f          },
-  /*{ 4.2076f, 11.4877f, 5.7924f, 9.1401f }*/
-};
-
-void NfcFilterCreate1(NfcFilter *nfc, const float w0, const float w1)
-{
-    float b_00, g_0;
-    float r;
-
-    memset(nfc, 0, sizeof(*nfc));
-
-    nfc->g = 1.0f;
-    nfc->coeffs[0] = 1.0f;
-
-    /* Calculate bass-boost coefficients. */
-    r = 0.5f * w0;
-    b_00 = B[1][0] * r;
-    g_0 = 1.0f + b_00;
-
-    nfc->coeffs[0] *= g_0;
-    nfc->coeffs[1] = (2.0f * b_00) / g_0;
-
-    /* Calculate bass-cut coefficients. */
-    r = 0.5f * w1;
-    b_00 = B[1][0] * r;
-    g_0 = 1.0f + b_00;
-
-    nfc->g /= g_0;
-    nfc->coeffs[0] /= g_0;
-    nfc->coeffs[1+1] = (2.0f * b_00) / g_0;
-}
-
-void NfcFilterAdjust1(NfcFilter *nfc, const float w0)
-{
-    float b_00, g_0;
-    float r;
-
-    r = 0.5f * w0;
-    b_00 = B[1][0] * r;
-    g_0 = 1.0f + b_00;
-
-    nfc->coeffs[0] = nfc->g * g_0;
-    nfc->coeffs[1] = (2.0f * b_00) / g_0;
-}
-
-void NfcFilterUpdate1(NfcFilter *nfc, ALfloat *restrict dst, const float *restrict src, const int count)
-{
-    const float b0 = nfc->coeffs[0];
-    const float a0 = nfc->coeffs[1];
-    const float a1 = nfc->coeffs[2];
-    float z1 = nfc->history[0];
-    int i;
-
-    for(i = 0;i < count;i++)
-    {
-        float out = src[i] * b0;
-        float y;
-
-        y = out - (a1*z1);
-        out = y + (a0*z1);
-        z1 += y;
-
-        dst[i] = out;
-    }
-    nfc->history[0] = z1;
-}
-
-
-void NfcFilterCreate2(NfcFilter *nfc, const float w0, const float w1)
-{
-    float b_10, b_11, g_1;
-    float r;
-
-    memset(nfc, 0, sizeof(*nfc));
-
-    nfc->g = 1.0f;
-    nfc->coeffs[0] = 1.0f;
-
-    /* Calculate bass-boost coefficients. */
-    r = 0.5f * w0;
-    b_10 = B[2][0] * r;
-    b_11 = B[2][1] * r * r;
-    g_1 = 1.0f + b_10 + b_11;
-
-    nfc->coeffs[0] *= g_1;
-    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-    nfc->coeffs[2] = (4.0f * b_11) / g_1;
-
-    /* Calculate bass-cut coefficients. */
-    r = 0.5f * w1;
-    b_10 = B[2][0] * r;
-    b_11 = B[2][1] * r * r;
-    g_1 = 1.0f + b_10 + b_11;
-
-    nfc->g /= g_1;
-    nfc->coeffs[0] /= g_1;
-    nfc->coeffs[2+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-    nfc->coeffs[2+2] = (4.0f * b_11) / g_1;
-}
-
-void NfcFilterAdjust2(NfcFilter *nfc, const float w0)
-{
-    float b_10, b_11, g_1;
-    float r;
-
-    r = 0.5f * w0;
-    b_10 = B[2][0] * r;
-    b_11 = B[2][1] * r * r;
-    g_1 = 1.0f + b_10 + b_11;
-
-    nfc->coeffs[0] = nfc->g * g_1;
-    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-    nfc->coeffs[2] = (4.0f * b_11) / g_1;
-}
-
-void NfcFilterUpdate2(NfcFilter *nfc, ALfloat *restrict dst, const float *restrict src, const int count)
-{
-    const float b0 = nfc->coeffs[0];
-    const float a00 = nfc->coeffs[1];
-    const float a01 = nfc->coeffs[2];
-    const float a10 = nfc->coeffs[3];
-    const float a11 = nfc->coeffs[4];
-    float z1 = nfc->history[0];
-    float z2 = nfc->history[1];
-    int i;
-
-    for(i = 0;i < count;i++)
-    {
-        float out = src[i] * b0;
-        float y;
-
-        y = out - (a10*z1) - (a11*z2);
-        out = y + (a00*z1) + (a01*z2);
-        z2 += z1;
-        z1 += y;
-
-        dst[i] = out;
-    }
-    nfc->history[0] = z1;
-    nfc->history[1] = z2;
-}
-
-
-void NfcFilterCreate3(NfcFilter *nfc, const float w0, const float w1)
-{
-    float b_10, b_11, g_1;
-    float b_00, g_0;
-    float r;
-
-    memset(nfc, 0, sizeof(*nfc));
-
-    nfc->g = 1.0f;
-    nfc->coeffs[0] = 1.0f;
-
-    /* Calculate bass-boost coefficients. */
-    r = 0.5f * w0;
-    b_10 = B[3][0] * r;
-    b_11 = B[3][1] * r * r;
-    g_1 = 1.0f + b_10 + b_11;
-
-    nfc->coeffs[0] *= g_1;
-    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-    nfc->coeffs[2] = (4.0f * b_11) / g_1;
-
-    b_00 = B[3][2] * r;
-    g_0 = 1.0f + b_00;
-
-    nfc->coeffs[0] *= g_0;
-    nfc->coeffs[2+1] = (2.0f * b_00) / g_0;
-
-    /* Calculate bass-cut coefficients. */
-    r = 0.5f * w1;
-    b_10 = B[3][0] * r;
-    b_11 = B[3][1] * r * r;
-    g_1 = 1.0f + b_10 + b_11;
-
-    nfc->g /= g_1;
-    nfc->coeffs[0] /= g_1;
-    nfc->coeffs[3+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-    nfc->coeffs[3+2] = (4.0f * b_11) / g_1;
-    
-    b_00 = B[3][2] * r;
-    g_0 = 1.0f + b_00;
-
-    nfc->g /= g_0;
-    nfc->coeffs[0] /= g_0;
-    nfc->coeffs[3+2+1] = (2.0f * b_00) / g_0;
-}
-
-void NfcFilterAdjust3(NfcFilter *nfc, const float w0)
-{
-    float b_10, b_11, g_1;
-    float b_00, g_0;
-    float r;
-
-    r = 0.5f * w0;
-    b_10 = B[3][0] * r;
-    b_11 = B[3][1] * r * r;
-    g_1 = 1.0f + b_10 + b_11;
-
-    nfc->coeffs[0] = nfc->g * g_1;
-    nfc->coeffs[1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-    nfc->coeffs[2] = (4.0f * b_11) / g_1;
-
-    b_00 = B[3][2] * r;
-    g_0 = 1.0f + b_00;
-
-    nfc->coeffs[0] *= g_0;
-    nfc->coeffs[2+1] = (2.0f * b_00) / g_0;
-}
-
-void NfcFilterUpdate3(NfcFilter *nfc, ALfloat *restrict dst, const float *restrict src, const int count)
-{
-    const float b0 = nfc->coeffs[0];
-    const float a00 = nfc->coeffs[1];
-    const float a01 = nfc->coeffs[2];
-    const float a02 = nfc->coeffs[3];
-    const float a10 = nfc->coeffs[4];
-    const float a11 = nfc->coeffs[5];
-    const float a12 = nfc->coeffs[6];
-    float z1 = nfc->history[0];
-    float z2 = nfc->history[1];
-    float z3 = nfc->history[2];
-    int i;
-
-    for(i = 0;i < count;i++)
-    {
-        float out = src[i] * b0;
-        float y;
-
-        y = out - (a10*z1) - (a11*z2);
-        out = y + (a00*z1) + (a01*z2);
-        z2 += z1;
-        z1 += y;
-
-        y = out - (a12*z3);
-        out = y + (a02*z3);
-        z3 += y;
-
-        dst[i] = out;
-    }
-    nfc->history[0] = z1;
-    nfc->history[1] = z2;
-    nfc->history[2] = z3;
-}
-
-
-#if 0 /* Original methods the above are derived from. */
-static void NfcFilterCreate(NfcFilter *nfc, const ALsizei order, const float src_dist, const float ctl_dist, const float rate)
-{
-    static const float B[4][5] = {
-        {                                     },
-        {    1.0f                             },
-        {    3.0f,     3.0f                   },
-        { 3.6778f,  6.4595f, 2.3222f          },
-        { 4.2076f, 11.4877f, 5.7924f, 9.1401f }
-    };
-    float w0 = SPEEDOFSOUNDMETRESPERSEC / (src_dist * rate);
-    float w1 = SPEEDOFSOUNDMETRESPERSEC / (ctl_dist * rate);
-    ALsizei i;
-    float r;
-
-    nfc->g = 1.0f;
-    nfc->coeffs[0] = 1.0f;
-
-    /* NOTE: Slight adjustment from the literature to raise the center
-     * frequency a bit (0.5 -> 1.0).
-     */
-    r = 1.0f * w0;
-    for(i = 0; i < (order-1);i += 2)
-    {
-        float b_10 = B[order][i  ] * r;
-        float b_11 = B[order][i+1] * r * r;
-        float g_1 = 1.0f + b_10 + b_11;
-
-        nfc->b[i] = b_10;
-        nfc->b[i + 1] = b_11;
-        nfc->coeffs[0] *= g_1;
-        nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-        nfc->coeffs[i+2] = (4.0f * b_11) / g_1;
-    }
-    if(i < order)
-    {
-        float b_00 = B[order][i] * r;
-        float g_0 = 1.0f + b_00;
-
-        nfc->b[i] = b_00;
-        nfc->coeffs[0] *= g_0;
-        nfc->coeffs[i+1] = (2.0f * b_00) / g_0;
-    }
-
-    r = 1.0f * w1;
-    for(i = 0;i < (order-1);i += 2)
-    {
-        float b_10 = B[order][i  ] * r;
-        float b_11 = B[order][i+1] * r * r;
-        float g_1 = 1.0f + b_10 + b_11;
-
-        nfc->g /= g_1;
-        nfc->coeffs[0] /= g_1;
-        nfc->coeffs[order+i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-        nfc->coeffs[order+i+2] = (4.0f * b_11) / g_1;
-    }
-    if(i < order)
-    {
-        float b_00 = B[order][i] * r;
-        float g_0 = 1.0f + b_00;
-
-        nfc->g /= g_0;
-        nfc->coeffs[0] /= g_0;
-        nfc->coeffs[order+i+1] = (2.0f * b_00) / g_0;
-    }
-
-    for(i = 0; i < MAX_AMBI_ORDER; i++)
-        nfc->history[i] = 0.0f;
-}
-
-static void NfcFilterAdjust(NfcFilter *nfc, const float distance)
-{
-    int i;
-
-    nfc->coeffs[0] = nfc->g;
-
-    for(i = 0;i < (nfc->order-1);i += 2)
-    {
-        float b_10 = nfc->b[i] / distance;
-        float b_11 = nfc->b[i+1] / (distance * distance);
-        float g_1 = 1.0f + b_10 + b_11;
-
-        nfc->coeffs[0] *= g_1;
-        nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
-        nfc->coeffs[i+2] = (4.0f * b_11) / g_1;
-    }
-    if(i < nfc->order)
-    {
-        float b_00 = nfc->b[i] / distance;
-        float g_0 = 1.0f + b_00;
-
-        nfc->coeffs[0] *= g_0;
-        nfc->coeffs[i+1] = (2.0f * b_00) / g_0;
-    }
-}
-
-static float NfcFilterUpdate(const float in, NfcFilter *nfc)
-{
-    int i;
-    float out = in * nfc->coeffs[0];
-
-    for(i = 0;i < (nfc->order-1);i += 2)
-    {
-        float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) -
-                        (nfc->coeffs[nfc->order+i+2] * nfc->history[i+1]) + 1.0e-30f;
-        out = y + (nfc->coeffs[i+1]*nfc->history[i]) + (nfc->coeffs[i+2]*nfc->history[i+1]);
-
-        nfc->history[i+1] += nfc->history[i];
-        nfc->history[i] += y;
-    }
-    if(i < nfc->order)
-    {
-        float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) + 1.0e-30f;
-
-        out = y + (nfc->coeffs[i+1] * nfc->history[i]);
-        nfc->history[i] += y;
-    }
-
-    return out;
-}
-#endif

Alc/nfcfilter.h

@@ -1,37 +0,0 @@
-#ifndef NFCFILTER_H
-#define NFCFILTER_H
-
-#include "alMain.h"
-
-typedef struct NfcFilter {
-    float g;
-    float coeffs[MAX_AMBI_ORDER*2 + 1];
-    float history[MAX_AMBI_ORDER];
-} NfcFilter;
-
-/* NOTE:
- * w0 = speed_of_sound / (source_distance * sample_rate);
- * w1 = speed_of_sound / (control_distance * sample_rate);
- *
- * Generally speaking, the control distance should be approximately the average
- * speaker distance, or based on the reference delay if outputing NFC-HOA. It
- * must not be negative, 0, or infinite. The source distance should not be too
- * small relative to the control distance.
- */
-
-/* Near-field control filter for first-order ambisonic channels (1-3). */
-void NfcFilterCreate1(NfcFilter *nfc, const float w0, const float w1);
-void NfcFilterAdjust1(NfcFilter *nfc, const float w0);
-void NfcFilterUpdate1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
-
-/* Near-field control filter for second-order ambisonic channels (4-8). */
-void NfcFilterCreate2(NfcFilter *nfc, const float w0, const float w1);
-void NfcFilterAdjust2(NfcFilter *nfc, const float w0);
-void NfcFilterUpdate2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
-
-/* Near-field control filter for third-order ambisonic channels (9-15). */
-void NfcFilterCreate3(NfcFilter *nfc, const float w0, const float w1);
-void NfcFilterAdjust3(NfcFilter *nfc, const float w0);
-void NfcFilterUpdate3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
-
-#endif /* NFCFILTER_H */

Alc/uhjfilter.c

@@ -1,134 +0,0 @@
-
-#include "config.h"
-
-#include "alu.h"
-#include "uhjfilter.h"
-
-/* This is the maximum number of samples processed for each inner loop
- * iteration. */
-#define MAX_UPDATE_SAMPLES  128
-
-
-static const ALfloat Filter1Coeff[4] = {
-    0.6923878f, 0.9360654322959f, 0.9882295226860f, 0.9987488452737f
-};
-static const ALfloat Filter2Coeff[4] = {
-    0.4021921162426f, 0.8561710882420f, 0.9722909545651f, 0.9952884791278f
-};
-
-static void allpass_process(AllPassState *state, ALfloat *restrict dst, const ALfloat *restrict src, const ALfloat aa, ALsizei todo)
-{
-    ALsizei i;
-
-    if(todo > 1)
-    {
-        dst[0] = aa*(src[0] + state->y[1]) - state->x[1];
-        dst[1] = aa*(src[1] + state->y[0]) - state->x[0];
-        for(i = 2;i < todo;i++)
-            dst[i] = aa*(src[i] + dst[i-2]) - src[i-2];
-        state->x[1] = src[i-2];
-        state->x[0] = src[i-1];
-        state->y[1] = dst[i-2];
-        state->y[0] = dst[i-1];
-    }
-    else if(todo == 1)
-    {
-        dst[0] = aa*(src[0] + state->y[1]) - state->x[1];
-        state->x[1] = state->x[0];
-        state->x[0] = src[0];
-        state->y[1] = state->y[0];
-        state->y[0] = dst[0];
-    }
-}
-
-
-/* NOTE: There seems to be a bit of an inconsistency in how this encoding is
- * supposed to work. Some references, such as
- *
- * http://members.tripod.com/martin_leese/Ambisonic/UHJ_file_format.html
- *
- * specify a pre-scaling of sqrt(2) on the W channel input, while other
- * references, such as
- *
- * https://en.wikipedia.org/wiki/Ambisonic_UHJ_format#Encoding.5B1.5D
- * and
- * https://wiki.xiph.org/Ambisonics#UHJ_format
- *
- * do not. The sqrt(2) scaling is in line with B-Format decoder coefficients
- * which include such a scaling for the W channel input, however the original
- * source for this equation is a 1985 paper by Michael Gerzon, which does not
- * apparently include the scaling. Applying the extra scaling creates a louder
- * result with a narrower stereo image compared to not scaling, and I don't
- * know which is the intended result.
- */
-
-void EncodeUhj2(Uhj2Encoder *enc, ALfloat *restrict LeftOut, ALfloat *restrict RightOut, ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
-{
-    ALfloat D[MAX_UPDATE_SAMPLES], S[MAX_UPDATE_SAMPLES];
-    ALfloat temp[2][MAX_UPDATE_SAMPLES];
-    ALsizei base, i;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        ALsizei todo = mini(SamplesToDo - base, MAX_UPDATE_SAMPLES);
-
-        /* D = 0.6554516*Y */
-        for(i = 0;i < todo;i++)
-            temp[0][i] = 0.6554516f*InSamples[2][base+i];
-        allpass_process(&enc->Filter1_Y[0], temp[1], temp[0],
-                        Filter1Coeff[0]*Filter1Coeff[0], todo);
-        allpass_process(&enc->Filter1_Y[1], temp[0], temp[1],
-                        Filter1Coeff[1]*Filter1Coeff[1], todo);
-        allpass_process(&enc->Filter1_Y[2], temp[1], temp[0],
-                        Filter1Coeff[2]*Filter1Coeff[2], todo);
-        /* NOTE: Filter1 requires a 1 sample delay for the final output, so
-         * take the last processed sample from the previous run as the first
-         * output sample.
-         */
-        D[0] = enc->Filter1_Y[3].y[0];
-        allpass_process(&enc->Filter1_Y[3], temp[0], temp[1],
-                        Filter1Coeff[3]*Filter1Coeff[3], todo);
-        for(i = 1;i < todo;i++)
-            D[i] = temp[0][i-1];
-
-        /* D += j(-0.3420201*W + 0.5098604*X) */
-        for(i = 0;i < todo;i++)
-            temp[0][i] = -0.3420201f*InSamples[0][base+i] +
-                          0.5098604f*InSamples[1][base+i];
-        allpass_process(&enc->Filter2_WX[0], temp[1], temp[0],
-                        Filter2Coeff[0]*Filter2Coeff[0], todo);
-        allpass_process(&enc->Filter2_WX[1], temp[0], temp[1],
-                        Filter2Coeff[1]*Filter2Coeff[1], todo);
-        allpass_process(&enc->Filter2_WX[2], temp[1], temp[0],
-                        Filter2Coeff[2]*Filter2Coeff[2], todo);
-        allpass_process(&enc->Filter2_WX[3], temp[0], temp[1],
-                        Filter2Coeff[3]*Filter2Coeff[3], todo);
-        for(i = 0;i < todo;i++)
-            D[i] += temp[0][i];
-
-        /* S = 0.9396926*W + 0.1855740*X */
-        for(i = 0;i < todo;i++)
-            temp[0][i] = 0.9396926f*InSamples[0][base+i] +
-                         0.1855740f*InSamples[1][base+i];
-        allpass_process(&enc->Filter1_WX[0], temp[1], temp[0],
-                        Filter1Coeff[0]*Filter1Coeff[0], todo);
-        allpass_process(&enc->Filter1_WX[1], temp[0], temp[1],
-                        Filter1Coeff[1]*Filter1Coeff[1], todo);
-        allpass_process(&enc->Filter1_WX[2], temp[1], temp[0],
-                        Filter1Coeff[2]*Filter1Coeff[2], todo);
-        S[0] = enc->Filter1_WX[3].y[0];
-        allpass_process(&enc->Filter1_WX[3], temp[0], temp[1],
-                        Filter1Coeff[3]*Filter1Coeff[3], todo);
-        for(i = 1;i < todo;i++)
-            S[i] = temp[0][i-1];
-
-        /* Left = (S + D)/2.0 */
-        for(i = 0;i < todo;i++)
-            *(LeftOut++) += (S[i] + D[i]) * 0.5f;
-        /* Right = (S - D)/2.0 */
-        for(i = 0;i < todo;i++)
-            *(RightOut++) += (S[i] - D[i]) * 0.5f;
-
-        base += todo;
-    }
-}

Alc/uhjfilter.h

@@ -1,49 +0,0 @@
-#ifndef UHJFILTER_H
-#define UHJFILTER_H
-
-#include "AL/al.h"
-
-#include "alMain.h"
-
-typedef struct AllPassState {
-    ALfloat x[2]; /* Last two input samples */
-    ALfloat y[2]; /* Last two output samples */
-} AllPassState;
-
-/* Encoding 2-channel UHJ from B-Format is done as:
- *
- * S = 0.9396926*W + 0.1855740*X
- * D = j(-0.3420201*W + 0.5098604*X) + 0.6554516*Y
- *
- * Left = (S + D)/2.0
- * Right = (S - D)/2.0
- *
- * where j is a wide-band +90 degree phase shift.
- *
- * The phase shift is done using a Hilbert transform, described here:
- * https://web.archive.org/web/20060708031958/http://www.biochem.oulu.fi/~oniemita/dsp/hilbert/
- * It works using 2 sets of 4 chained filters. The first filter chain produces
- * a phase shift of varying magnitude over a wide range of frequencies, while
- * the second filter chain produces a phase shift 90 degrees ahead of the
- * first over the same range.
- *
- * Combining these two stages requires the use of three filter chains. S-
- * channel output uses a Filter1 chain on the W and X channel mix, while the D-
- * channel output uses a Filter1 chain on the Y channel plus a Filter2 chain on
- * the W and X channel mix. This results in the W and X input mix on the D-
- * channel output having the required +90 degree phase shift relative to the
- * other inputs.
- */
-
-typedef struct Uhj2Encoder {
-    AllPassState Filter1_WX[4];
-    AllPassState Filter1_Y[4];
-    AllPassState Filter2_WX[4];
-} Uhj2Encoder;
-
-/* Encodes a 2-channel UHJ (stereo-compatible) signal from a B-Format input
- * signal. The input must use FuMa channel ordering and scaling.
- */
-void EncodeUhj2(Uhj2Encoder *enc, ALfloat *restrict LeftOut, ALfloat *restrict RightOut, ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
-
-#endif /* UHJFILTER_H */

Alc/vector.h

@@ -1,93 +0,0 @@
-#ifndef AL_VECTOR_H
-#define AL_VECTOR_H
-
-#include <stdlib.h>
-
-#include <AL/al.h>
-
-#include "almalloc.h"
-
-
-#define TYPEDEF_VECTOR(T, N) typedef struct {                                 \
-    size_t Capacity;                                                          \
-    size_t Size;                                                              \
-    T Data[];                                                                 \
-} _##N;                                                                       \
-typedef _##N* N;                                                              \
-typedef const _##N* const_##N;
-
-#define VECTOR(T) struct {                                                    \
-    size_t Capacity;                                                          \
-    size_t Size;                                                              \
-    T Data[];                                                                 \
-}*
-
-#define VECTOR_INIT(_x)       do { (_x) = NULL; } while(0)
-#define VECTOR_INIT_STATIC()  NULL
-#define VECTOR_DEINIT(_x)     do { al_free((_x)); (_x) = NULL; } while(0)
-
-#define VECTOR_RESIZE(_x, _s, _c) do {                                        \
-    size_t _size = (_s);                                                      \
-    size_t _cap = (_c);                                                       \
-    if(_size > _cap)                                                          \
-        _cap = _size;                                                         \
-                                                                              \
-    if(!(_x) && _cap == 0)                                                    \
-        break;                                                                \
-                                                                              \
-    if(((_x) ? (_x)->Capacity : 0) < _cap)                                    \
-    {                                                                         \
-        ptrdiff_t data_offset = (char*)((_x)->Data) - (char*)(_x);            \
-        size_t old_size = ((_x) ? (_x)->Size : 0);                            \
-        void *temp;                                                           \
-                                                                              \
-        temp = al_calloc(16, data_offset + sizeof((_x)->Data[0])*_cap);       \
-        assert(temp != NULL);                                                 \
-        if((_x))                                                              \
-            memcpy(((char*)temp)+data_offset, (_x)->Data,                     \
-                   sizeof((_x)->Data[0])*old_size);                           \
-                                                                              \
-        al_free((_x));                                                        \
-        (_x) = temp;                                                          \
-        (_x)->Capacity = _cap;                                                \
-    }                                                                         \
-    (_x)->Size = _size;                                                       \
-} while(0)                                                                    \
-
-#define VECTOR_CAPACITY(_x) ((_x) ? (_x)->Capacity : 0)
-#define VECTOR_SIZE(_x)     ((_x) ? (_x)->Size : 0)
-
-#define VECTOR_BEGIN(_x) ((_x) ? (_x)->Data + 0 : NULL)
-#define VECTOR_END(_x)   ((_x) ? (_x)->Data + (_x)->Size : NULL)
-
-#define VECTOR_PUSH_BACK(_x, _obj) do {      \
-    size_t _pbsize = VECTOR_SIZE(_x)+1;      \
-    VECTOR_RESIZE(_x, _pbsize, _pbsize);     \
-    (_x)->Data[(_x)->Size-1] = (_obj);       \
-} while(0)
-#define VECTOR_POP_BACK(_x) ((void)((_x)->Size--))
-
-#define VECTOR_BACK(_x)  ((_x)->Data[(_x)->Size-1])
-#define VECTOR_FRONT(_x) ((_x)->Data[0])
-
-#define VECTOR_ELEM(_x, _o) ((_x)->Data[(_o)])
-
-#define VECTOR_FOR_EACH(_t, _x, _f)  do {                                     \
-    _t *_iter = VECTOR_BEGIN((_x));                                           \
-    _t *_end = VECTOR_END((_x));                                              \
-    for(;_iter != _end;++_iter)                                               \
-        _f(_iter);                                                            \
-} while(0)
-
-#define VECTOR_FIND_IF(_i, _t, _x, _f)  do {                                  \
-    _t *_iter = VECTOR_BEGIN((_x));                                           \
-    _t *_end = VECTOR_END((_x));                                              \
-    for(;_iter != _end;++_iter)                                               \
-    {                                                                         \
-        if(_f(_iter))                                                         \
-            break;                                                            \
-    }                                                                         \
-    (_i) = _iter;                                                             \
-} while(0)
-
-#endif /* AL_VECTOR_H */

COPYING

@@ -1,481 +0,0 @@
-                  GNU LIBRARY GENERAL PUBLIC LICENSE
-                       Version 2, June 1991
-
- Copyright (C) 1991 Free Software Foundation, Inc.
- 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
- Everyone is permitted to copy and distribute verbatim copies
- of this license document, but changing it is not allowed.
-
-[This is the first released version of the library GPL.  It is
- numbered 2 because it goes with version 2 of the ordinary GPL.]
-
-                            Preamble
-
-  The licenses for most software are designed to take away your
-freedom to share and change it.  By contrast, the GNU General Public
-Licenses are intended to guarantee your freedom to share and change
-free software--to make sure the software is free for all its users.
-
-  This license, the Library General Public License, applies to some
-specially designated Free Software Foundation software, and to any
-other libraries whose authors decide to use it.  You can use it for
-your libraries, too.
-
-  When we speak of free software, we are referring to freedom, not
-price.  Our General Public Licenses are designed to make sure that you
-have the freedom to distribute copies of free software (and charge for
-this service if you wish), that you receive source code or can get it
-if you want it, that you can change the software or use pieces of it
-in new free programs; and that you know you can do these things.
-
-  To protect your rights, we need to make restrictions that forbid
-anyone to deny you these rights or to ask you to surrender the rights.
-These restrictions translate to certain responsibilities for you if
-you distribute copies of the library, or if you modify it.
-
-  For example, if you distribute copies of the library, whether gratis
-or for a fee, you must give the recipients all the rights that we gave
-you.  You must make sure that they, too, receive or can get the source
-code.  If you link a program with the library, you must provide
-complete object files to the recipients so that they can relink them
-with the library, after making changes to the library and recompiling
-it.  And you must show them these terms so they know their rights.
-
-  Our method of protecting your rights has two steps: (1) copyright
-the library, and (2) offer you this license which gives you legal
-permission to copy, distribute and/or modify the library.
-
-  Also, for each distributor's protection, we want to make certain
-that everyone understands that there is no warranty for this free
-library.  If the library is modified by someone else and passed on, we
-want its recipients to know that what they have is not the original
-version, so that any problems introduced by others will not reflect on
-the original authors' reputations.
-
-  Finally, any free program is threatened constantly by software
-patents.  We wish to avoid the danger that companies distributing free
-software will individually obtain patent licenses, thus in effect
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-we have made it clear that any patent must be licensed for everyone's
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-
-  Most GNU software, including some libraries, is covered by the ordinary
-GNU General Public License, which was designed for utility programs.  This
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-  The reason we have a separate public license for some libraries is that
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-treats it as such.
-
-  Because of this blurred distinction, using the ordinary General
-Public License for libraries did not effectively promote software
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-  However, unrestricted linking of non-free programs would deprive the
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-preserving your freedom as a user of such programs to change the free
-libraries that are incorporated in them.  (We have not seen how to achieve
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-
-  The precise terms and conditions for copying, distribution and
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-
-  Note that it is possible for a library to be covered by the ordinary
-General Public License rather than by this special one.
-
-                  GNU LIBRARY GENERAL PUBLIC LICENSE
-   TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
-
-  0. This License Agreement applies to any software library which
-contains a notice placed by the copyright holder or other authorized
-party saying it may be distributed under the terms of this Library
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ChangeLog

@@ -1,290 +0,0 @@
-openal-soft-1.18.1:
-
-    Fixed an issue where resuming a source might not restart playing it.
-
-    Fixed PulseAudio playback when the configured stream length is much less
-    than the requested length.
-
-    Fixed MMDevAPI capture with sample rates not matching the backing device.
-
-    Fixed int32 output for the Wave Writer.
-
-    Fixed enumeration of OSS devices that are missing device files.
-
-    Added correct retrieval of the executable's path on FreeBSD.
-
-    Added a config option to specify the dithering depth.
-
-    Added a 5.1 decoder preset that excludes front-center output.
-
-openal-soft-1.18.0:
-
-    Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
-
-    Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
-    AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
-
-    Implemented 3D processing for some effects. Currently implemented for
-    Reverb, Compressor, Equalizer, and Ring Modulator.
-
-    Implemented 2-channel UHJ output encoding. This needs to be enabled with a
-    config option to be used.
-
-    Implemented dual-band processing for high-quality ambisonic decoding.
-
-    Implemented distance-compensation for surround sound output.
-
-    Implemented near-field emulation and compensation with ambisonic rendering.
-    Currently only applies when using the high-quality ambisonic decoder or
-    ambisonic output, with appropriate config options.
-
-    Implemented an output limiter to reduce the amount of distortion from
-    clipping.
-
-    Implemented dithering for 8-bit and 16-bit output.
-
-    Implemented a config option to select a preferred HRTF.
-
-    Implemented a run-time check for NEON extensions using /proc/cpuinfo.
-
-    Implemented experimental capture support for the OpenSL backend.
-
-    Fixed building on compilers with NEON support but don't default to having
-    NEON enabled.
-
-    Fixed support for JACK on Windows.
-
-    Fixed starting a source while alcSuspendContext is in effect.
-
-    Fixed detection of headsets as headphones, with MMDevAPI.
-
-    Added support for AmbDec config files, for custom ambisonic decoder
-    configurations. Version 3 files only.
-
-    Added backend-specific options to alsoft-config.
-
-    Added first-, second-, and third-order ambisonic output formats. Currently
-    only works with backends that don't rely on channel labels, like JACK,
-    ALSA, and OSS.
-
-    Added a build option to embed the default HRTFs into the lib.
-
-    Added AmbDec presets to enable high-quality ambisonic decoding.
-
-    Added an AmbDec preset for 3D7.1 speaker setups.
-
-    Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
-    the provided ambdec presets.
-
-    Added the ability for MMDevAPI to open devices given a Device ID or GUID
-    string.
-
-    Added an option to the example apps to open a specific device.
-
-    Increased the maximum auxiliary send limit to 16 (up from 4). Requires
-    requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
-    attribute.
-
-    Increased the default auxiliary effect slot count to 64 (up from 4).
-
-    Reduced the default period count to 3 (down from 4).
-
-    Slightly improved automatic naming for enumerated HRTFs.
-
-    Improved B-Format decoding with HRTF output.
-
-    Improved internal property handling for better batching behavior.
-
-    Improved performance of certain filter uses.
-
-    Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
-    extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
-
-openal-soft-1.17.2:
-
-    Implemented device enumeration for OSSv4.
-
-    Fixed building on OSX.
-
-    Fixed building on non-Windows systems without POSIX-2008.
-
-    Fixed Dedicated Dialog and Dedicated LFE effect output.
-
-    Added a build option to override the share install dir.
-
-    Added a build option to static-link libgcc for MinGW.
-
-openal-soft-1.17.1:
-
-    Fixed building with JACK and without PulseAudio.
-
-    Fixed building on FreeBSD.
-
-    Fixed the ALSA backend's allow-resampler option.
-
-    Fixed handling of inexact ALSA period counts.
-
-    Altered device naming scheme on Windows backends to better match other
-    drivers.
-
-    Updated the CoreAudio backend to use the AudioComponent API. This clears up
-    deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
-
-openal-soft-1.17.0:
-
-    Implemented a JACK playback backend.
-
-    Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
-
-    Implemented the ALC_SOFT_HRTF extension.
-
-    Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
-
-    Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
-    24-point Sinc resampling, and performs anti-aliasing.
-
-    Implemented B-Format output support for the wave file writer. This creates
-    FuMa-style first-order Ambisonics wave files (AMB format).
-
-    Implemented a stereo-mode config option for treating stereo modes as either
-    speakers or headphones.
-
-    Implemented per-device configuration options.
-
-    Fixed handling of PulseAudio and MMDevAPI devices that have identical
-    descriptions.
-
-    Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
-
-    Fixed logging of Unicode characters on Windows.
-
-    Fixed 5.1 surround sound channels. By default it will now use the side
-    channels for the surround output. A configuration using rear channels is
-    still available.
-
-    Fixed the QSA backend potentially altering the capture format.
-
-    Fixed detecting MMDevAPI's default device.
-
-    Fixed returning the default capture device name.
-
-    Fixed mixing property calculations when deferring context updates.
-
-    Altered the behavior of alcSuspendContext and alcProcessContext to better
-    match certain Windows drivers.
-
-    Altered the panning algorithm, utilizing Ambisonics for better side and
-    back positioning cues with surround sound output.
-
-    Improved support for certain older Windows apps.
-
-    Improved the alffplay example to support surround sound streams.
-
-    Improved support for building as a sub-project.
-
-    Added an HRTF playback example.
-
-    Added a tone generator output test.
-
-    Added a toolchain to help with cross-compiling to Android.
-
-openal-soft-1.16.0:
-
-    Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
-    effects.
-
-    Implemented high-pass and band-pass EFX filters.
-
-    Implemented the high-pass filter for the EAXReverb effect.
-
-    Implemented SSE2 and SSE4.1 linear resamplers.
-
-    Implemented Neon-enhanced non-HRTF mixers.
-
-    Implemented a QSA backend, for QNX.
-
-    Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
-    AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
-    extensions.
-
-    Fixed resetting mmdevapi backend devices.
-
-    Fixed clamping when converting 32-bit float samples to integer.
-
-    Fixed modulation range in the Modulator effect.
-
-    Several fixes for the OpenSL playback backend.
-
-    Fixed device specifier names that have Unicode characters on Windows.
-
-    Added support for filenames and paths with Unicode (UTF-8) characters on
-    Windows.
-
-    Added support for alsoft.conf config files found in XDG Base Directory
-    Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
-    defaults) on non-Windows systems.
-
-    Added a GUI configuration utility (requires Qt 4.8).
-
-    Added support for environment variable expansion in config options (not
-    keys or section names).
-
-    Added an example that uses SDL2 and ffmpeg.
-
-    Modified examples to use SDL_sound.
-
-    Modified CMake config option names for better sorting.
-
-    HRTF data sets specified in the hrtf_tables config option may now be
-    relative or absolute filenames.
-
-    Made the default HRTF data set an external file, and added a data set for
-    48khz playback in addition to 44.1khz.
-
-    Added support for C11 atomic methods.
-
-    Improved support for some non-GNU build systems.
-
-openal-soft-1.15.1:
-
-    Fixed a regression with retrieving the source's AL_GAIN property.
-
-openal-soft-1.15:
-
-    Fixed device enumeration with the OSS backend.
-
-    Reorganized internal mixing logic, so unneeded steps can potentially be
-    skipped for better performance.
-
-    Removed the lookup table for calculating the mixing pans. The panning is
-    now calculated directly for better precision.
-
-    Improved the panning of stereo source channels when using stereo output.
-
-    Improved source filter quality on send paths.
-
-    Added a config option to allow PulseAudio to move streams between devices.
-
-    The PulseAudio backend will now attempt to spawn a server by default.
-
-    Added a workaround for a DirectSound bug relating to float32 output.
-
-    Added SSE-based mixers, for HRTF and non-HRTF mixing.
-
-    Added support for the new AL_SOFT_source_latency extension.
-
-    Improved ALSA capture by avoiding an extra buffer when using sizes
-    supported by the underlying device.
-
-    Improved the makehrtf utility to support new options and input formats.
-
-    Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
-    the header includes can optionally be omitted.
-
-    Added a couple example code programs to show how to apply reverb, and
-    retrieve latency.
-
-    The configuration sample is now installed into the share/openal/ directory
-    instead of /etc/openal.
-
-    The configuration sample now gets installed by default.

OpenAL32/Include/alAuxEffectSlot.h

@@ -1,190 +0,0 @@
-#ifndef _AL_AUXEFFECTSLOT_H_
-#define _AL_AUXEFFECTSLOT_H_
-
-#include "alMain.h"
-#include "alEffect.h"
-
-#include "atomic.h"
-#include "align.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-struct ALeffectStateVtable;
-struct ALeffectslot;
-
-typedef struct ALeffectState {
-    RefCount Ref;
-    const struct ALeffectStateVtable *vtbl;
-
-    ALfloat (*OutBuffer)[BUFFERSIZE];
-    ALsizei OutChannels;
-} ALeffectState;
-
-void ALeffectState_Construct(ALeffectState *state);
-void ALeffectState_Destruct(ALeffectState *state);
-
-struct ALeffectStateVtable {
-    void (*const Destruct)(ALeffectState *state);
-
-    ALboolean (*const deviceUpdate)(ALeffectState *state, ALCdevice *device);
-    void (*const update)(ALeffectState *state, const ALCdevice *device, const struct ALeffectslot *slot, const union ALeffectProps *props);
-    void (*const process)(ALeffectState *state, ALsizei samplesToDo, const ALfloat (*restrict samplesIn)[BUFFERSIZE], ALfloat (*restrict samplesOut)[BUFFERSIZE], ALsizei numChannels);
-
-    void (*const Delete)(void *ptr);
-};
-
-#define DEFINE_ALEFFECTSTATE_VTABLE(T)                                        \
-DECLARE_THUNK(T, ALeffectState, void, Destruct)                               \
-DECLARE_THUNK1(T, ALeffectState, ALboolean, deviceUpdate, ALCdevice*)         \
-DECLARE_THUNK3(T, ALeffectState, void, update, const ALCdevice*, const ALeffectslot*, const ALeffectProps*) \
-DECLARE_THUNK4(T, ALeffectState, void, process, ALsizei, const ALfloatBUFFERSIZE*restrict, ALfloatBUFFERSIZE*restrict, ALsizei) \
-static void T##_ALeffectState_Delete(void *ptr)                               \
-{ return T##_Delete(STATIC_UPCAST(T, ALeffectState, (ALeffectState*)ptr)); }  \
-                                                                              \
-static const struct ALeffectStateVtable T##_ALeffectState_vtable = {          \
-    T##_ALeffectState_Destruct,                                               \
-                                                                              \
-    T##_ALeffectState_deviceUpdate,                                           \
-    T##_ALeffectState_update,                                                 \
-    T##_ALeffectState_process,                                                \
-                                                                              \
-    T##_ALeffectState_Delete,                                                 \
-}
-
-
-struct ALeffectStateFactoryVtable;
-
-typedef struct ALeffectStateFactory {
-    const struct ALeffectStateFactoryVtable *vtbl;
-} ALeffectStateFactory;
-
-struct ALeffectStateFactoryVtable {
-    ALeffectState *(*const create)(ALeffectStateFactory *factory);
-};
-
-#define DEFINE_ALEFFECTSTATEFACTORY_VTABLE(T)                                 \
-DECLARE_THUNK(T, ALeffectStateFactory, ALeffectState*, create)                \
-                                                                              \
-static const struct ALeffectStateFactoryVtable T##_ALeffectStateFactory_vtable = { \
-    T##_ALeffectStateFactory_create,                                          \
-}
-
-
-#define MAX_EFFECT_CHANNELS (4)
-
-
-struct ALeffectslotArray {
-    ALsizei count;
-    struct ALeffectslot *slot[];
-};
-
-
-struct ALeffectslotProps {
-    ALfloat   Gain;
-    ALboolean AuxSendAuto;
-
-    ALenum Type;
-    ALeffectProps Props;
-
-    ALeffectState *State;
-
-    ATOMIC(struct ALeffectslotProps*) next;
-};
-
-
-typedef struct ALeffectslot {
-    ALfloat   Gain;
-    ALboolean AuxSendAuto;
-
-    struct {
-        ALenum Type;
-        ALeffectProps Props;
-
-        ALeffectState *State;
-    } Effect;
-
-    ATOMIC_FLAG PropsClean;
-
-    RefCount ref;
-
-    ATOMIC(struct ALeffectslotProps*) Update;
-    ATOMIC(struct ALeffectslotProps*) FreeList;
-
-    struct {
-        ALfloat   Gain;
-        ALboolean AuxSendAuto;
-
-        ALenum EffectType;
-        ALeffectState *EffectState;
-
-        ALfloat RoomRolloff; /* Added to the source's room rolloff, not multiplied. */
-        ALfloat DecayTime;
-        ALfloat DecayHFRatio;
-        ALboolean DecayHFLimit;
-        ALfloat AirAbsorptionGainHF;
-    } Params;
-
-    /* Self ID */
-    ALuint id;
-
-    ALsizei NumChannels;
-    BFChannelConfig ChanMap[MAX_EFFECT_CHANNELS];
-    /* Wet buffer configuration is ACN channel order with N3D scaling:
-     * * Channel 0 is the unattenuated mono signal.
-     * * Channel 1 is OpenAL -X
-     * * Channel 2 is OpenAL Y
-     * * Channel 3 is OpenAL -Z
-     * Consequently, effects that only want to work with mono input can use
-     * channel 0 by itself. Effects that want multichannel can process the
-     * ambisonics signal and make a B-Format pan (ComputeFirstOrderGains) for
-     * first-order device output (FOAOut).
-     */
-    alignas(16) ALfloat WetBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE];
-} ALeffectslot;
-
-inline void LockEffectSlotsRead(ALCcontext *context)
-{ LockUIntMapRead(&context->EffectSlotMap); }
-inline void UnlockEffectSlotsRead(ALCcontext *context)
-{ UnlockUIntMapRead(&context->EffectSlotMap); }
-inline void LockEffectSlotsWrite(ALCcontext *context)
-{ LockUIntMapWrite(&context->EffectSlotMap); }
-inline void UnlockEffectSlotsWrite(ALCcontext *context)
-{ UnlockUIntMapWrite(&context->EffectSlotMap); }
-
-inline struct ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id)
-{ return (struct ALeffectslot*)LookupUIntMapKeyNoLock(&context->EffectSlotMap, id); }
-inline struct ALeffectslot *RemoveEffectSlot(ALCcontext *context, ALuint id)
-{ return (struct ALeffectslot*)RemoveUIntMapKeyNoLock(&context->EffectSlotMap, id); }
-
-ALenum InitEffectSlot(ALeffectslot *slot);
-void DeinitEffectSlot(ALeffectslot *slot);
-void UpdateEffectSlotProps(ALeffectslot *slot);
-void UpdateAllEffectSlotProps(ALCcontext *context);
-ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context);
-
-
-ALeffectStateFactory *ALnullStateFactory_getFactory(void);
-ALeffectStateFactory *ALreverbStateFactory_getFactory(void);
-ALeffectStateFactory *ALchorusStateFactory_getFactory(void);
-ALeffectStateFactory *ALcompressorStateFactory_getFactory(void);
-ALeffectStateFactory *ALdistortionStateFactory_getFactory(void);
-ALeffectStateFactory *ALechoStateFactory_getFactory(void);
-ALeffectStateFactory *ALequalizerStateFactory_getFactory(void);
-ALeffectStateFactory *ALflangerStateFactory_getFactory(void);
-ALeffectStateFactory *ALmodulatorStateFactory_getFactory(void);
-
-ALeffectStateFactory *ALdedicatedStateFactory_getFactory(void);
-
-
-ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect);
-
-void InitEffectFactoryMap(void);
-void DeinitEffectFactoryMap(void);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alBuffer.h

@@ -1,128 +0,0 @@
-#ifndef _AL_BUFFER_H_
-#define _AL_BUFFER_H_
-
-#include "alMain.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/* User formats */
-enum UserFmtType {
-    UserFmtByte   = AL_BYTE_SOFT,
-    UserFmtUByte  = AL_UNSIGNED_BYTE_SOFT,
-    UserFmtShort  = AL_SHORT_SOFT,
-    UserFmtUShort = AL_UNSIGNED_SHORT_SOFT,
-    UserFmtInt    = AL_INT_SOFT,
-    UserFmtUInt   = AL_UNSIGNED_INT_SOFT,
-    UserFmtFloat  = AL_FLOAT_SOFT,
-    UserFmtDouble = AL_DOUBLE_SOFT,
-    UserFmtMulaw  = AL_MULAW_SOFT,
-    UserFmtAlaw   = 0x10000000,
-    UserFmtIMA4,
-    UserFmtMSADPCM,
-};
-enum UserFmtChannels {
-    UserFmtMono      = AL_MONO_SOFT,
-    UserFmtStereo    = AL_STEREO_SOFT,
-    UserFmtRear      = AL_REAR_SOFT,
-    UserFmtQuad      = AL_QUAD_SOFT,
-    UserFmtX51       = AL_5POINT1_SOFT, /* (WFX order) */
-    UserFmtX61       = AL_6POINT1_SOFT, /* (WFX order) */
-    UserFmtX71       = AL_7POINT1_SOFT, /* (WFX order) */
-    UserFmtBFormat2D = AL_BFORMAT2D_SOFT, /* WXY */
-    UserFmtBFormat3D = AL_BFORMAT3D_SOFT, /* WXYZ */
-};
-
-ALsizei BytesFromUserFmt(enum UserFmtType type);
-ALsizei ChannelsFromUserFmt(enum UserFmtChannels chans);
-inline ALsizei FrameSizeFromUserFmt(enum UserFmtChannels chans, enum UserFmtType type)
-{
-    return ChannelsFromUserFmt(chans) * BytesFromUserFmt(type);
-}
-
-
-/* Storable formats */
-enum FmtType {
-    FmtByte  = UserFmtByte,
-    FmtShort = UserFmtShort,
-    FmtFloat = UserFmtFloat,
-};
-enum FmtChannels {
-    FmtMono   = UserFmtMono,
-    FmtStereo = UserFmtStereo,
-    FmtRear   = UserFmtRear,
-    FmtQuad   = UserFmtQuad,
-    FmtX51    = UserFmtX51,
-    FmtX61    = UserFmtX61,
-    FmtX71    = UserFmtX71,
-    FmtBFormat2D = UserFmtBFormat2D,
-    FmtBFormat3D = UserFmtBFormat3D,
-};
-#define MAX_INPUT_CHANNELS  (8)
-
-ALsizei BytesFromFmt(enum FmtType type);
-ALsizei ChannelsFromFmt(enum FmtChannels chans);
-inline ALsizei FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type)
-{
-    return ChannelsFromFmt(chans) * BytesFromFmt(type);
-}
-
-
-typedef struct ALbuffer {
-    ALvoid  *data;
-
-    ALsizei  Frequency;
-    ALenum   Format;
-    ALsizei  SampleLen;
-
-    enum FmtChannels FmtChannels;
-    enum FmtType     FmtType;
-    ALuint BytesAlloc;
-
-    enum UserFmtChannels OriginalChannels;
-    enum UserFmtType     OriginalType;
-    ALsizei              OriginalSize;
-    ALsizei              OriginalAlign;
-
-    ALsizei LoopStart;
-    ALsizei LoopEnd;
-
-    ATOMIC(ALsizei) UnpackAlign;
-    ATOMIC(ALsizei) PackAlign;
-
-    /* Number of times buffer was attached to a source (deletion can only occur when 0) */
-    RefCount ref;
-
-    RWLock lock;
-
-    /* Self ID */
-    ALuint id;
-} ALbuffer;
-
-ALbuffer *NewBuffer(ALCcontext *context);
-void DeleteBuffer(ALCdevice *device, ALbuffer *buffer);
-
-ALenum LoadData(ALbuffer *buffer, ALuint freq, ALenum NewFormat, ALsizei frames, enum UserFmtChannels SrcChannels, enum UserFmtType SrcType, const ALvoid *data, ALsizei align, ALboolean storesrc);
-
-inline void LockBuffersRead(ALCdevice *device)
-{ LockUIntMapRead(&device->BufferMap); }
-inline void UnlockBuffersRead(ALCdevice *device)
-{ UnlockUIntMapRead(&device->BufferMap); }
-inline void LockBuffersWrite(ALCdevice *device)
-{ LockUIntMapWrite(&device->BufferMap); }
-inline void UnlockBuffersWrite(ALCdevice *device)
-{ UnlockUIntMapWrite(&device->BufferMap); }
-
-inline struct ALbuffer *LookupBuffer(ALCdevice *device, ALuint id)
-{ return (struct ALbuffer*)LookupUIntMapKeyNoLock(&device->BufferMap, id); }
-inline struct ALbuffer *RemoveBuffer(ALCdevice *device, ALuint id)
-{ return (struct ALbuffer*)RemoveUIntMapKeyNoLock(&device->BufferMap, id); }
-
-ALvoid ReleaseALBuffers(ALCdevice *device);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alEffect.h

@@ -1,196 +0,0 @@
-#ifndef _AL_EFFECT_H_
-#define _AL_EFFECT_H_
-
-#include "alMain.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-struct ALeffect;
-
-enum {
-    AL__EAXREVERB = 0,
-    AL__REVERB,
-    AL__CHORUS,
-    AL__COMPRESSOR,
-    AL__DISTORTION,
-    AL__ECHO,
-    AL__EQUALIZER,
-    AL__FLANGER,
-    AL__MODULATOR,
-    AL__DEDICATED,
-
-    MAX_EFFECTS
-};
-extern ALboolean DisabledEffects[MAX_EFFECTS];
-
-extern ALfloat ReverbBoost;
-extern ALboolean EmulateEAXReverb;
-
-struct ALeffectVtable {
-    void (*const setParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
-    void (*const setParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
-    void (*const setParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
-    void (*const setParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals);
-
-    void (*const getParami)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *val);
-    void (*const getParamiv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals);
-    void (*const getParamf)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
-    void (*const getParamfv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
-};
-
-#define DEFINE_ALEFFECT_VTABLE(T)           \
-const struct ALeffectVtable T##_vtable = {  \
-    T##_setParami, T##_setParamiv,          \
-    T##_setParamf, T##_setParamfv,          \
-    T##_getParami, T##_getParamiv,          \
-    T##_getParamf, T##_getParamfv,          \
-}
-
-extern const struct ALeffectVtable ALeaxreverb_vtable;
-extern const struct ALeffectVtable ALreverb_vtable;
-extern const struct ALeffectVtable ALchorus_vtable;
-extern const struct ALeffectVtable ALcompressor_vtable;
-extern const struct ALeffectVtable ALdistortion_vtable;
-extern const struct ALeffectVtable ALecho_vtable;
-extern const struct ALeffectVtable ALequalizer_vtable;
-extern const struct ALeffectVtable ALflanger_vtable;
-extern const struct ALeffectVtable ALmodulator_vtable;
-extern const struct ALeffectVtable ALnull_vtable;
-extern const struct ALeffectVtable ALdedicated_vtable;
-
-
-typedef union ALeffectProps {
-    struct {
-        // Shared Reverb Properties
-        ALfloat Density;
-        ALfloat Diffusion;
-        ALfloat Gain;
-        ALfloat GainHF;
-        ALfloat DecayTime;
-        ALfloat DecayHFRatio;
-        ALfloat ReflectionsGain;
-        ALfloat ReflectionsDelay;
-        ALfloat LateReverbGain;
-        ALfloat LateReverbDelay;
-        ALfloat AirAbsorptionGainHF;
-        ALfloat RoomRolloffFactor;
-        ALboolean DecayHFLimit;
-
-        // Additional EAX Reverb Properties
-        ALfloat GainLF;
-        ALfloat DecayLFRatio;
-        ALfloat ReflectionsPan[3];
-        ALfloat LateReverbPan[3];
-        ALfloat EchoTime;
-        ALfloat EchoDepth;
-        ALfloat ModulationTime;
-        ALfloat ModulationDepth;
-        ALfloat HFReference;
-        ALfloat LFReference;
-    } Reverb;
-
-    struct {
-        ALint Waveform;
-        ALint Phase;
-        ALfloat Rate;
-        ALfloat Depth;
-        ALfloat Feedback;
-        ALfloat Delay;
-    } Chorus;
-
-    struct {
-        ALboolean OnOff;
-    } Compressor;
-
-    struct {
-        ALfloat Edge;
-        ALfloat Gain;
-        ALfloat LowpassCutoff;
-        ALfloat EQCenter;
-        ALfloat EQBandwidth;
-    } Distortion;
-
-    struct {
-        ALfloat Delay;
-        ALfloat LRDelay;
-
-        ALfloat Damping;
-        ALfloat Feedback;
-
-        ALfloat Spread;
-    } Echo;
-
-    struct {
-        ALfloat LowCutoff;
-        ALfloat LowGain;
-        ALfloat Mid1Center;
-        ALfloat Mid1Gain;
-        ALfloat Mid1Width;
-        ALfloat Mid2Center;
-        ALfloat Mid2Gain;
-        ALfloat Mid2Width;
-        ALfloat HighCutoff;
-        ALfloat HighGain;
-    } Equalizer;
-
-    struct {
-        ALint Waveform;
-        ALint Phase;
-        ALfloat Rate;
-        ALfloat Depth;
-        ALfloat Feedback;
-        ALfloat Delay;
-    } Flanger;
-
-    struct {
-        ALfloat Frequency;
-        ALfloat HighPassCutoff;
-        ALint Waveform;
-    } Modulator;
-
-    struct {
-        ALfloat Gain;
-    } Dedicated;
-} ALeffectProps;
-
-typedef struct ALeffect {
-    // Effect type (AL_EFFECT_NULL, ...)
-    ALenum type;
-
-    ALeffectProps Props;
-
-    const struct ALeffectVtable *vtbl;
-
-    /* Self ID */
-    ALuint id;
-} ALeffect;
-
-inline void LockEffectsRead(ALCdevice *device)
-{ LockUIntMapRead(&device->EffectMap); }
-inline void UnlockEffectsRead(ALCdevice *device)
-{ UnlockUIntMapRead(&device->EffectMap); }
-inline void LockEffectsWrite(ALCdevice *device)
-{ LockUIntMapWrite(&device->EffectMap); }
-inline void UnlockEffectsWrite(ALCdevice *device)
-{ UnlockUIntMapWrite(&device->EffectMap); }
-
-inline struct ALeffect *LookupEffect(ALCdevice *device, ALuint id)
-{ return (struct ALeffect*)LookupUIntMapKeyNoLock(&device->EffectMap, id); }
-inline struct ALeffect *RemoveEffect(ALCdevice *device, ALuint id)
-{ return (struct ALeffect*)RemoveUIntMapKeyNoLock(&device->EffectMap, id); }
-
-inline ALboolean IsReverbEffect(ALenum type)
-{ return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; }
-
-ALenum InitEffect(ALeffect *effect);
-ALvoid ReleaseALEffects(ALCdevice *device);
-
-ALvoid LoadReverbPreset(const char *name, ALeffect *effect);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alError.h

@@ -1,33 +0,0 @@
-#ifndef _AL_ERROR_H_
-#define _AL_ERROR_H_
-
-#include "alMain.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-extern ALboolean TrapALError;
-
-ALvoid alSetError(ALCcontext *Context, ALenum errorCode);
-
-#define SET_ERROR_AND_RETURN(ctx, err) do {                                    \
-    alSetError((ctx), (err));                                                  \
-    return;                                                                    \
-} while(0)
-
-#define SET_ERROR_AND_RETURN_VALUE(ctx, err, val) do {                         \
-    alSetError((ctx), (err));                                                  \
-    return (val);                                                              \
-} while(0)
-
-#define SET_ERROR_AND_GOTO(ctx, err, lbl) do {                                 \
-    alSetError((ctx), (err));                                                  \
-    goto lbl;                                                                  \
-} while(0)
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alFilter.h

@@ -1,164 +0,0 @@
-#ifndef _AL_FILTER_H_
-#define _AL_FILTER_H_
-
-#include "alMain.h"
-
-#include "math_defs.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#define LOWPASSFREQREF  (5000.0f)
-#define HIGHPASSFREQREF  (250.0f)
-
-
-/* Filters implementation is based on the "Cookbook formulae for audio
- * EQ biquad filter coefficients" by Robert Bristow-Johnson
- * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
- */
-/* Implementation note: For the shelf filters, the specified gain is for the
- * reference frequency, which is the centerpoint of the transition band. This
- * better matches EFX filter design. To set the gain for the shelf itself, use
- * the square root of the desired linear gain (or halve the dB gain).
- */
-
-typedef enum ALfilterType {
-    /** EFX-style low-pass filter, specifying a gain and reference frequency. */
-    ALfilterType_HighShelf,
-    /** EFX-style high-pass filter, specifying a gain and reference frequency. */
-    ALfilterType_LowShelf,
-    /** Peaking filter, specifying a gain and reference frequency. */
-    ALfilterType_Peaking,
-
-    /** Low-pass cut-off filter, specifying a cut-off frequency. */
-    ALfilterType_LowPass,
-    /** High-pass cut-off filter, specifying a cut-off frequency. */
-    ALfilterType_HighPass,
-    /** Band-pass filter, specifying a center frequency. */
-    ALfilterType_BandPass,
-} ALfilterType;
-
-typedef struct ALfilterState {
-    ALfloat x[2]; /* History of two last input samples  */
-    ALfloat y[2]; /* History of two last output samples */
-    ALfloat b0, b1, b2; /* Transfer function coefficients "b" */
-    ALfloat a1, a2; /* Transfer function coefficients "a" (a0 is pre-applied) */
-} ALfilterState;
-/* Currently only a C-based filter process method is implemented. */
-#define ALfilterState_process ALfilterState_processC
-
-/* Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
- * reference gain and shelf slope parameter.
- * 0 < gain
- * 0 < slope <= 1
- */
-inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope)
-{
-    return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f);
-}
-/* Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the frequency
- * multiple (i.e. ref_freq / sampling_freq) and bandwidth.
- * 0 < freq_mult < 0.5.
- */
-inline ALfloat calc_rcpQ_from_bandwidth(ALfloat freq_mult, ALfloat bandwidth)
-{
-    ALfloat w0 = F_TAU * freq_mult;
-    return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
-}
-
-inline void ALfilterState_clear(ALfilterState *filter)
-{
-    filter->x[0] = 0.0f;
-    filter->x[1] = 0.0f;
-    filter->y[0] = 0.0f;
-    filter->y[1] = 0.0f;
-}
-
-void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat rcpQ);
-
-inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src)
-{
-    dst->b0 = src->b0;
-    dst->b1 = src->b1;
-    dst->b2 = src->b2;
-    dst->a1 = src->a1;
-    dst->a2 = src->a2;
-}
-
-void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples);
-
-inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples)
-{
-    if(numsamples >= 2)
-    {
-        filter->x[1] = src[numsamples-2];
-        filter->x[0] = src[numsamples-1];
-        filter->y[1] = src[numsamples-2];
-        filter->y[0] = src[numsamples-1];
-    }
-    else if(numsamples == 1)
-    {
-        filter->x[1] = filter->x[0];
-        filter->x[0] = src[0];
-        filter->y[1] = filter->y[0];
-        filter->y[0] = src[0];
-    }
-}
-
-
-typedef struct ALfilter {
-    // Filter type (AL_FILTER_NULL, ...)
-    ALenum type;
-
-    ALfloat Gain;
-    ALfloat GainHF;
-    ALfloat HFReference;
-    ALfloat GainLF;
-    ALfloat LFReference;
-
-    void (*SetParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint val);
-    void (*SetParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals);
-    void (*SetParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val);
-    void (*SetParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals);
-
-    void (*GetParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *val);
-    void (*GetParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals);
-    void (*GetParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val);
-    void (*GetParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals);
-
-    /* Self ID */
-    ALuint id;
-} ALfilter;
-
-#define ALfilter_SetParami(x, c, p, v)  ((x)->SetParami((x),(c),(p),(v)))
-#define ALfilter_SetParamiv(x, c, p, v) ((x)->SetParamiv((x),(c),(p),(v)))
-#define ALfilter_SetParamf(x, c, p, v)  ((x)->SetParamf((x),(c),(p),(v)))
-#define ALfilter_SetParamfv(x, c, p, v) ((x)->SetParamfv((x),(c),(p),(v)))
-
-#define ALfilter_GetParami(x, c, p, v)  ((x)->GetParami((x),(c),(p),(v)))
-#define ALfilter_GetParamiv(x, c, p, v) ((x)->GetParamiv((x),(c),(p),(v)))
-#define ALfilter_GetParamf(x, c, p, v)  ((x)->GetParamf((x),(c),(p),(v)))
-#define ALfilter_GetParamfv(x, c, p, v) ((x)->GetParamfv((x),(c),(p),(v)))
-
-inline void LockFiltersRead(ALCdevice *device)
-{ LockUIntMapRead(&device->FilterMap); }
-inline void UnlockFiltersRead(ALCdevice *device)
-{ UnlockUIntMapRead(&device->FilterMap); }
-inline void LockFiltersWrite(ALCdevice *device)
-{ LockUIntMapWrite(&device->FilterMap); }
-inline void UnlockFiltersWrite(ALCdevice *device)
-{ UnlockUIntMapWrite(&device->FilterMap); }
-
-inline struct ALfilter *LookupFilter(ALCdevice *device, ALuint id)
-{ return (struct ALfilter*)LookupUIntMapKeyNoLock(&device->FilterMap, id); }
-inline struct ALfilter *RemoveFilter(ALCdevice *device, ALuint id)
-{ return (struct ALfilter*)RemoveUIntMapKeyNoLock(&device->FilterMap, id); }
-
-ALvoid ReleaseALFilters(ALCdevice *device);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alListener.h

@@ -1,66 +0,0 @@
-#ifndef _AL_LISTENER_H_
-#define _AL_LISTENER_H_
-
-#include "alMain.h"
-#include "alu.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-struct ALlistenerProps {
-    ALfloat Position[3];
-    ALfloat Velocity[3];
-    ALfloat Forward[3];
-    ALfloat Up[3];
-    ALfloat Gain;
-    ALfloat MetersPerUnit;
-
-    ALfloat DopplerFactor;
-    ALfloat DopplerVelocity;
-    ALfloat SpeedOfSound;
-    ALboolean SourceDistanceModel;
-    enum DistanceModel DistanceModel;
-
-    ATOMIC(struct ALlistenerProps*) next;
-};
-
-typedef struct ALlistener {
-    alignas(16) ALfloat Position[3];
-    ALfloat Velocity[3];
-    ALfloat Forward[3];
-    ALfloat Up[3];
-    ALfloat Gain;
-    ALfloat MetersPerUnit;
-
-    /* Pointer to the most recent property values that are awaiting an update.
-     */
-    ATOMIC(struct ALlistenerProps*) Update;
-
-    /* A linked list of unused property containers, free to use for future
-     * updates.
-     */
-    ATOMIC(struct ALlistenerProps*) FreeList;
-
-    struct {
-        aluMatrixf Matrix;
-        aluVector  Velocity;
-
-        ALfloat Gain;
-        ALfloat MetersPerUnit;
-
-        ALfloat DopplerFactor;
-        ALfloat SpeedOfSound;
-
-        ALboolean SourceDistanceModel;
-        enum DistanceModel DistanceModel;
-    } Params;
-} ALlistener;
-
-void UpdateListenerProps(ALCcontext *context);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alSource.h

@@ -1,128 +0,0 @@
-#ifndef _AL_SOURCE_H_
-#define _AL_SOURCE_H_
-
-#include "bool.h"
-#include "alMain.h"
-#include "alu.h"
-#include "hrtf.h"
-#include "atomic.h"
-
-#define MAX_SENDS      16
-#define DEFAULT_SENDS  2
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-struct ALbuffer;
-struct ALsource;
-
-
-typedef struct ALbufferlistitem {
-    struct ALbuffer *buffer;
-    ATOMIC(struct ALbufferlistitem*) next;
-} ALbufferlistitem;
-
-
-typedef struct ALsource {
-    /** Source properties. */
-    ALfloat   Pitch;
-    ALfloat   Gain;
-    ALfloat   OuterGain;
-    ALfloat   MinGain;
-    ALfloat   MaxGain;
-    ALfloat   InnerAngle;
-    ALfloat   OuterAngle;
-    ALfloat   RefDistance;
-    ALfloat   MaxDistance;
-    ALfloat   RolloffFactor;
-    ALfloat   Position[3];
-    ALfloat   Velocity[3];
-    ALfloat   Direction[3];
-    ALfloat   Orientation[2][3];
-    ALboolean HeadRelative;
-    ALboolean Looping;
-    enum DistanceModel DistanceModel;
-    enum Resampler Resampler;
-    ALboolean DirectChannels;
-    enum SpatializeMode Spatialize;
-
-    ALboolean DryGainHFAuto;
-    ALboolean WetGainAuto;
-    ALboolean WetGainHFAuto;
-    ALfloat   OuterGainHF;
-
-    ALfloat AirAbsorptionFactor;
-    ALfloat RoomRolloffFactor;
-    ALfloat DopplerFactor;
-
-    /* NOTE: Stereo pan angles are specified in radians, counter-clockwise
-     * rather than clockwise.
-     */
-    ALfloat StereoPan[2];
-
-    ALfloat Radius;
-
-    /** Direct filter and auxiliary send info. */
-    struct {
-        ALfloat Gain;
-        ALfloat GainHF;
-        ALfloat HFReference;
-        ALfloat GainLF;
-        ALfloat LFReference;
-    } Direct;
-    struct {
-        struct ALeffectslot *Slot;
-        ALfloat Gain;
-        ALfloat GainHF;
-        ALfloat HFReference;
-        ALfloat GainLF;
-        ALfloat LFReference;
-    } *Send;
-
-    /**
-     * Last user-specified offset, and the offset type (bytes, samples, or
-     * seconds).
-     */
-    ALdouble Offset;
-    ALenum   OffsetType;
-
-    /** Source type (static, streaming, or undetermined) */
-    ALint SourceType;
-
-    /** Source state (initial, playing, paused, or stopped) */
-    ATOMIC(ALenum) state;
-
-    /** Source Buffer Queue head. */
-    RWLock queue_lock;
-    ALbufferlistitem *queue;
-
-    ATOMIC_FLAG PropsClean;
-
-    /** Self ID */
-    ALuint id;
-} ALsource;
-
-inline void LockSourcesRead(ALCcontext *context)
-{ LockUIntMapRead(&context->SourceMap); }
-inline void UnlockSourcesRead(ALCcontext *context)
-{ UnlockUIntMapRead(&context->SourceMap); }
-inline void LockSourcesWrite(ALCcontext *context)
-{ LockUIntMapWrite(&context->SourceMap); }
-inline void UnlockSourcesWrite(ALCcontext *context)
-{ UnlockUIntMapWrite(&context->SourceMap); }
-
-inline struct ALsource *LookupSource(ALCcontext *context, ALuint id)
-{ return (struct ALsource*)LookupUIntMapKeyNoLock(&context->SourceMap, id); }
-inline struct ALsource *RemoveSource(ALCcontext *context, ALuint id)
-{ return (struct ALsource*)RemoveUIntMapKeyNoLock(&context->SourceMap, id); }
-
-void UpdateAllSourceProps(ALCcontext *context);
-
-ALvoid ReleaseALSources(ALCcontext *Context);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/alThunk.h

@@ -1,20 +0,0 @@
-#ifndef ALTHUNK_H
-#define ALTHUNK_H
-
-#include "alMain.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-void ThunkInit(void);
-void ThunkExit(void);
-ALenum NewThunkEntry(ALuint *index);
-void FreeThunkEntry(ALuint index);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif //ALTHUNK_H
-

OpenAL32/Include/alu.h

@@ -1,516 +0,0 @@
-#ifndef _ALU_H_
-#define _ALU_H_
-
-#include <limits.h>
-#include <math.h>
-#ifdef HAVE_FLOAT_H
-#include <float.h>
-#endif
-#ifdef HAVE_IEEEFP_H
-#include <ieeefp.h>
-#endif
-
-#include "alMain.h"
-#include "alBuffer.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-
-#include "hrtf.h"
-#include "align.h"
-#include "nfcfilter.h"
-#include "math_defs.h"
-
-
-#define MAX_PITCH  (255)
-
-/* Maximum number of buffer samples before the current pos needed for resampling. */
-#define MAX_PRE_SAMPLES 12
-
-/* Maximum number of buffer samples after the current pos needed for resampling. */
-#define MAX_POST_SAMPLES 12
-
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-struct ALsource;
-struct ALbufferlistitem;
-struct ALvoice;
-struct ALeffectslot;
-
-
-#define DITHER_RNG_SEED 22222
-
-
-enum SpatializeMode {
-    SpatializeOff = AL_FALSE,
-    SpatializeOn = AL_TRUE,
-    SpatializeAuto = AL_AUTO_SOFT
-};
-
-enum Resampler {
-    PointResampler,
-    LinearResampler,
-    FIR4Resampler,
-    BSincResampler,
-
-    ResamplerMax = BSincResampler
-};
-extern enum Resampler ResamplerDefault;
-
-/* The number of distinct scale and phase intervals within the filter table. */
-#define BSINC_SCALE_BITS  4
-#define BSINC_SCALE_COUNT (1<<BSINC_SCALE_BITS)
-#define BSINC_PHASE_BITS  4
-#define BSINC_PHASE_COUNT (1<<BSINC_PHASE_BITS)
-
-/* Interpolator state.  Kind of a misnomer since the interpolator itself is
- * stateless.  This just keeps it from having to recompute scale-related
- * mappings for every sample.
- */
-typedef struct BsincState {
-    ALfloat sf; /* Scale interpolation factor. */
-    ALuint m;   /* Coefficient count. */
-    ALint l;    /* Left coefficient offset. */
-    struct {
-        const ALfloat *filter;   /* Filter coefficients. */
-        const ALfloat *scDelta;  /* Scale deltas. */
-        const ALfloat *phDelta;  /* Phase deltas. */
-        const ALfloat *spDelta;  /* Scale-phase deltas. */
-    } coeffs[BSINC_PHASE_COUNT];
-} BsincState;
-
-typedef union InterpState {
-    BsincState bsinc;
-} InterpState;
-
-ALboolean BsincPrepare(const ALuint increment, BsincState *state);
-
-typedef const ALfloat* (*ResamplerFunc)(const InterpState *state,
-    const ALfloat *restrict src, ALsizei frac, ALint increment,
-    ALfloat *restrict dst, ALsizei dstlen
-);
-
-
-typedef union aluVector {
-    alignas(16) ALfloat v[4];
-} aluVector;
-
-inline void aluVectorSet(aluVector *vector, ALfloat x, ALfloat y, ALfloat z, ALfloat w)
-{
-    vector->v[0] = x;
-    vector->v[1] = y;
-    vector->v[2] = z;
-    vector->v[3] = w;
-}
-
-
-typedef union aluMatrixf {
-    alignas(16) ALfloat m[4][4];
-} aluMatrixf;
-extern const aluMatrixf IdentityMatrixf;
-
-inline void aluMatrixfSetRow(aluMatrixf *matrix, ALuint row,
-                             ALfloat m0, ALfloat m1, ALfloat m2, ALfloat m3)
-{
-    matrix->m[row][0] = m0;
-    matrix->m[row][1] = m1;
-    matrix->m[row][2] = m2;
-    matrix->m[row][3] = m3;
-}
-
-inline void aluMatrixfSet(aluMatrixf *matrix, ALfloat m00, ALfloat m01, ALfloat m02, ALfloat m03,
-                                              ALfloat m10, ALfloat m11, ALfloat m12, ALfloat m13,
-                                              ALfloat m20, ALfloat m21, ALfloat m22, ALfloat m23,
-                                              ALfloat m30, ALfloat m31, ALfloat m32, ALfloat m33)
-{
-    aluMatrixfSetRow(matrix, 0, m00, m01, m02, m03);
-    aluMatrixfSetRow(matrix, 1, m10, m11, m12, m13);
-    aluMatrixfSetRow(matrix, 2, m20, m21, m22, m23);
-    aluMatrixfSetRow(matrix, 3, m30, m31, m32, m33);
-}
-
-
-enum ActiveFilters {
-    AF_None = 0,
-    AF_LowPass = 1,
-    AF_HighPass = 2,
-    AF_BandPass = AF_LowPass | AF_HighPass
-};
-
-
-typedef struct MixHrtfParams {
-    const ALfloat (*Coeffs)[2];
-    ALsizei Delay[2];
-    ALfloat Gain;
-    ALfloat GainStep;
-} MixHrtfParams;
-
-
-typedef struct DirectParams {
-    ALfilterState LowPass;
-    ALfilterState HighPass;
-
-    NfcFilter NFCtrlFilter[MAX_AMBI_ORDER];
-
-    struct {
-        HrtfParams Old;
-        HrtfParams Target;
-        HrtfState State;
-    } Hrtf;
-
-    struct {
-        ALfloat Current[MAX_OUTPUT_CHANNELS];
-        ALfloat Target[MAX_OUTPUT_CHANNELS];
-    } Gains;
-} DirectParams;
-
-typedef struct SendParams {
-    ALfilterState LowPass;
-    ALfilterState HighPass;
-
-    struct {
-        ALfloat Current[MAX_OUTPUT_CHANNELS];
-        ALfloat Target[MAX_OUTPUT_CHANNELS];
-    } Gains;
-} SendParams;
-
-
-struct ALvoiceProps {
-    ATOMIC(struct ALvoiceProps*) next;
-
-    ALfloat Pitch;
-    ALfloat Gain;
-    ALfloat OuterGain;
-    ALfloat MinGain;
-    ALfloat MaxGain;
-    ALfloat InnerAngle;
-    ALfloat OuterAngle;
-    ALfloat RefDistance;
-    ALfloat MaxDistance;
-    ALfloat RolloffFactor;
-    ALfloat Position[3];
-    ALfloat Velocity[3];
-    ALfloat Direction[3];
-    ALfloat Orientation[2][3];
-    ALboolean HeadRelative;
-    enum DistanceModel DistanceModel;
-    enum Resampler Resampler;
-    ALboolean DirectChannels;
-    enum SpatializeMode SpatializeMode;
-
-    ALboolean DryGainHFAuto;
-    ALboolean WetGainAuto;
-    ALboolean WetGainHFAuto;
-    ALfloat   OuterGainHF;
-
-    ALfloat AirAbsorptionFactor;
-    ALfloat RoomRolloffFactor;
-    ALfloat DopplerFactor;
-
-    ALfloat StereoPan[2];
-
-    ALfloat Radius;
-
-    /** Direct filter and auxiliary send info. */
-    struct {
-        ALfloat Gain;
-        ALfloat GainHF;
-        ALfloat HFReference;
-        ALfloat GainLF;
-        ALfloat LFReference;
-    } Direct;
-    struct {
-        struct ALeffectslot *Slot;
-        ALfloat Gain;
-        ALfloat GainHF;
-        ALfloat HFReference;
-        ALfloat GainLF;
-        ALfloat LFReference;
-    } Send[];
-};
-
-/* If not 'fading', gain targets are used directly without fading. */
-#define VOICE_IS_FADING (1<<0)
-#define VOICE_HAS_HRTF  (1<<1)
-#define VOICE_HAS_NFC   (1<<2)
-
-typedef struct ALvoice {
-    struct ALvoiceProps *Props;
-
-    ATOMIC(struct ALvoiceProps*) Update;
-    ATOMIC(struct ALvoiceProps*) FreeList;
-
-    ATOMIC(struct ALsource*) Source;
-    ATOMIC(bool) Playing;
-
-    /**
-     * Source offset in samples, relative to the currently playing buffer, NOT
-     * the whole queue, and the fractional (fixed-point) offset to the next
-     * sample.
-     */
-    ATOMIC(ALuint) position;
-    ATOMIC(ALsizei) position_fraction;
-
-    /* Current buffer queue item being played. */
-    ATOMIC(struct ALbufferlistitem*) current_buffer;
-
-    /* Buffer queue item to loop to at end of queue (will be NULL for non-
-     * looping voices).
-     */
-    ATOMIC(struct ALbufferlistitem*) loop_buffer;
-
-    /**
-     * Number of channels and bytes-per-sample for the attached source's
-     * buffer(s).
-     */
-    ALsizei NumChannels;
-    ALsizei SampleSize;
-
-    /** Current target parameters used for mixing. */
-    ALint Step;
-
-    ResamplerFunc Resampler;
-
-    ALuint Flags;
-
-    ALuint Offset; /* Number of output samples mixed since starting. */
-
-    alignas(16) ALfloat PrevSamples[MAX_INPUT_CHANNELS][MAX_PRE_SAMPLES];
-
-    InterpState ResampleState;
-
-    struct {
-        enum ActiveFilters FilterType;
-        DirectParams Params[MAX_INPUT_CHANNELS];
-
-        ALfloat (*Buffer)[BUFFERSIZE];
-        ALsizei Channels;
-        ALsizei ChannelsPerOrder[MAX_AMBI_ORDER+1];
-    } Direct;
-
-    struct {
-        enum ActiveFilters FilterType;
-        SendParams Params[MAX_INPUT_CHANNELS];
-
-        ALfloat (*Buffer)[BUFFERSIZE];
-        ALsizei Channels;
-    } Send[];
-} ALvoice;
-
-void DeinitVoice(ALvoice *voice);
-
-
-typedef void (*MixerFunc)(const ALfloat *data, ALsizei OutChans,
-                          ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains,
-                          const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
-                          ALsizei BufferSize);
-typedef void (*RowMixerFunc)(ALfloat *OutBuffer, const ALfloat *gains,
-                             const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
-                             ALsizei InPos, ALsizei BufferSize);
-typedef void (*HrtfMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                              const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                              const ALsizei IrSize, MixHrtfParams *hrtfparams,
-                              HrtfState *hrtfstate, ALsizei BufferSize);
-typedef void (*HrtfMixerBlendFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                                   const ALfloat *data, ALsizei Offset, ALsizei OutPos,
-                                   const ALsizei IrSize, const HrtfParams *oldparams,
-                                   MixHrtfParams *newparams, HrtfState *hrtfstate,
-                                   ALsizei BufferSize);
-typedef void (*HrtfDirectMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
-                                    const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
-                                    const ALfloat (*restrict Coeffs)[2],
-                                    ALfloat (*restrict Values)[2], ALsizei BufferSize);
-
-
-#define GAIN_MIX_MAX  (16.0f) /* +24dB */
-
-#define GAIN_SILENCE_THRESHOLD  (0.00001f) /* -100dB */
-
-#define SPEEDOFSOUNDMETRESPERSEC  (343.3f)
-#define AIRABSORBGAINHF           (0.99426f) /* -0.05dB */
-
-/* Target gain for the reverb decay feedback reaching the decay time. */
-#define REVERB_DECAY_GAIN  (0.001f) /* -60 dB */
-
-#define FRACTIONBITS (12)
-#define FRACTIONONE  (1<<FRACTIONBITS)
-#define FRACTIONMASK (FRACTIONONE-1)
-
-
-inline ALfloat minf(ALfloat a, ALfloat b)
-{ return ((a > b) ? b : a); }
-inline ALfloat maxf(ALfloat a, ALfloat b)
-{ return ((a > b) ? a : b); }
-inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max)
-{ return minf(max, maxf(min, val)); }
-
-inline ALdouble mind(ALdouble a, ALdouble b)
-{ return ((a > b) ? b : a); }
-inline ALdouble maxd(ALdouble a, ALdouble b)
-{ return ((a > b) ? a : b); }
-inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max)
-{ return mind(max, maxd(min, val)); }
-
-inline ALuint minu(ALuint a, ALuint b)
-{ return ((a > b) ? b : a); }
-inline ALuint maxu(ALuint a, ALuint b)
-{ return ((a > b) ? a : b); }
-inline ALuint clampu(ALuint val, ALuint min, ALuint max)
-{ return minu(max, maxu(min, val)); }
-
-inline ALint mini(ALint a, ALint b)
-{ return ((a > b) ? b : a); }
-inline ALint maxi(ALint a, ALint b)
-{ return ((a > b) ? a : b); }
-inline ALint clampi(ALint val, ALint min, ALint max)
-{ return mini(max, maxi(min, val)); }
-
-inline ALint64 mini64(ALint64 a, ALint64 b)
-{ return ((a > b) ? b : a); }
-inline ALint64 maxi64(ALint64 a, ALint64 b)
-{ return ((a > b) ? a : b); }
-inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max)
-{ return mini64(max, maxi64(min, val)); }
-
-inline ALuint64 minu64(ALuint64 a, ALuint64 b)
-{ return ((a > b) ? b : a); }
-inline ALuint64 maxu64(ALuint64 a, ALuint64 b)
-{ return ((a > b) ? a : b); }
-inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max)
-{ return minu64(max, maxu64(min, val)); }
-
-
-extern alignas(16) const ALfloat bsincTab[18840];
-extern alignas(16) const ALfloat sinc4Tab[FRACTIONONE][4];
-
-
-inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu)
-{
-    return val1 + (val2-val1)*mu;
-}
-inline ALfloat resample_fir4(ALfloat val0, ALfloat val1, ALfloat val2, ALfloat val3, ALsizei frac)
-{
-    return sinc4Tab[frac][0]*val0 + sinc4Tab[frac][1]*val1 +
-           sinc4Tab[frac][2]*val2 + sinc4Tab[frac][3]*val3;
-}
-
-
-enum HrtfRequestMode {
-    Hrtf_Default = 0,
-    Hrtf_Enable = 1,
-    Hrtf_Disable = 2,
-};
-
-void aluInitMixer(void);
-
-MixerFunc SelectMixer(void);
-RowMixerFunc SelectRowMixer(void);
-ResamplerFunc SelectResampler(enum Resampler resampler);
-
-/* aluInitRenderer
- *
- * Set up the appropriate panning method and mixing method given the device
- * properties.
- */
-void aluInitRenderer(ALCdevice *device, ALint hrtf_id, enum HrtfRequestMode hrtf_appreq, enum HrtfRequestMode hrtf_userreq);
-
-void aluInitEffectPanning(struct ALeffectslot *slot);
-
-/**
- * CalcDirectionCoeffs
- *
- * Calculates ambisonic coefficients based on a direction vector. The vector
- * must be normalized (unit length), and the spread is the angular width of the
- * sound (0...tau).
- */
-void CalcDirectionCoeffs(const ALfloat dir[3], ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]);
-
-/**
- * CalcAngleCoeffs
- *
- * Calculates ambisonic coefficients based on azimuth and elevation. The
- * azimuth and elevation parameters are in radians, going right and up
- * respectively.
- */
-inline void CalcAngleCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS])
-{
-    ALfloat dir[3] = {
-        sinf(azimuth) * cosf(elevation),
-        sinf(elevation),
-        -cosf(azimuth) * cosf(elevation)
-    };
-    CalcDirectionCoeffs(dir, spread, coeffs);
-}
-
-/**
- * CalcAnglePairwiseCoeffs
- *
- * Calculates ambisonic coefficients based on azimuth and elevation. The
- * azimuth and elevation parameters are in radians, going right and up
- * respectively. This pairwise variant warps the result such that +30 azimuth
- * is full right, and -30 azimuth is full left.
- */
-void CalcAnglePairwiseCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS]);
-
-/**
- * ComputeAmbientGains
- *
- * Computes channel gains for ambient, omni-directional sounds.
- */
-#define ComputeAmbientGains(b, g, o) do {                                     \
-    if((b).CoeffCount > 0)                                                    \
-        ComputeAmbientGainsMC((b).Ambi.Coeffs, (b).NumChannels, g, o);        \
-    else                                                                      \
-        ComputeAmbientGainsBF((b).Ambi.Map, (b).NumChannels, g, o);           \
-} while (0)
-void ComputeAmbientGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
-void ComputeAmbientGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
-
-/**
- * ComputePanningGains
- *
- * Computes panning gains using the given channel decoder coefficients and the
- * pre-calculated direction or angle coefficients.
- */
-#define ComputePanningGains(b, c, g, o) do {                                  \
-    if((b).CoeffCount > 0)                                                    \
-        ComputePanningGainsMC((b).Ambi.Coeffs, (b).NumChannels, (b).CoeffCount, c, g, o);\
-    else                                                                      \
-        ComputePanningGainsBF((b).Ambi.Map, (b).NumChannels, c, g, o);        \
-} while (0)
-void ComputePanningGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALsizei numcoeffs, const ALfloat coeffs[MAX_AMBI_COEFFS], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
-void ComputePanningGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat coeffs[MAX_AMBI_COEFFS], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
-
-/**
- * ComputeFirstOrderGains
- *
- * Sets channel gains for a first-order ambisonics input channel. The matrix is
- * a 1x4 'slice' of a transform matrix for the input channel, used to scale and
- * orient the sound samples.
- */
-#define ComputeFirstOrderGains(b, m, g, o) do {                               \
-    if((b).CoeffCount > 0)                                                    \
-        ComputeFirstOrderGainsMC((b).Ambi.Coeffs, (b).NumChannels, m, g, o);  \
-    else                                                                      \
-        ComputeFirstOrderGainsBF((b).Ambi.Map, (b).NumChannels, m, g, o);     \
-} while (0)
-void ComputeFirstOrderGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
-void ComputeFirstOrderGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat mtx[4], ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
-
-
-ALboolean MixSource(struct ALvoice *voice, struct ALsource *Source, ALCdevice *Device, ALsizei SamplesToDo);
-
-void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples);
-/* Caller must lock the device. */
-void aluHandleDisconnect(ALCdevice *device);
-
-extern ALfloat ConeScale;
-extern ALfloat ZScale;
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif

OpenAL32/Include/bs2b.h

@@ -1,94 +0,0 @@
-/*-
- * Copyright (c) 2005 Boris Mikhaylov
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files (the
- * "Software"), to deal in the Software without restriction, including
- * without limitation the rights to use, copy, modify, merge, publish,
- * distribute, sublicense, and/or sell copies of the Software, and to
- * permit persons to whom the Software is furnished to do so, subject to
- * the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
- * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
- * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
- * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-#ifndef BS2B_H
-#define BS2B_H
-
-/* Number of crossfeed levels */
-#define BS2B_CLEVELS           3
-
-/* Normal crossfeed levels */
-#define BS2B_HIGH_CLEVEL       3
-#define BS2B_MIDDLE_CLEVEL     2
-#define BS2B_LOW_CLEVEL        1
-
-/* Easy crossfeed levels */
-#define BS2B_HIGH_ECLEVEL      BS2B_HIGH_CLEVEL    + BS2B_CLEVELS
-#define BS2B_MIDDLE_ECLEVEL    BS2B_MIDDLE_CLEVEL  + BS2B_CLEVELS
-#define BS2B_LOW_ECLEVEL       BS2B_LOW_CLEVEL     + BS2B_CLEVELS
-
-/* Default crossfeed levels */
-#define BS2B_DEFAULT_CLEVEL    BS2B_HIGH_ECLEVEL
-/* Default sample rate (Hz) */
-#define BS2B_DEFAULT_SRATE     44100
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-struct bs2b {
-    int level;  /* Crossfeed level */
-    int srate;   /* Sample rate (Hz) */
-
-    /* Lowpass IIR filter coefficients */
-    float a0_lo;
-    float b1_lo;
-
-    /* Highboost IIR filter coefficients */
-    float a0_hi;
-    float a1_hi;
-    float b1_hi;
-
-    /* Buffer of last filtered sample.
-     * [0] - first channel, [1] - second channel
-     */
-    struct t_last_sample {
-        float asis;
-        float lo;
-        float hi;
-    } last_sample[2];
-};
-
-/* Clear buffers and set new coefficients with new crossfeed level and sample
- * rate values.
- * level - crossfeed level of *LEVEL values.
- * srate - sample rate by Hz.
- */
-void bs2b_set_params(struct bs2b *bs2b, int level, int srate);
-
-/* Return current crossfeed level value */
-int bs2b_get_level(struct bs2b *bs2b);
-
-/* Return current sample rate value */
-int bs2b_get_srate(struct bs2b *bs2b);
-
-/* Clear buffer */
-void bs2b_clear(struct bs2b *bs2b);
-
-void bs2b_cross_feed(struct bs2b *bs2b, float *restrict Left, float *restrict Right, int SamplesToDo);
-
-#ifdef __cplusplus
-}    /* extern "C" */
-#endif /* __cplusplus */
-
-#endif /* BS2B_H */

OpenAL32/Include/sample_cvt.h

@@ -1,9 +0,0 @@
-#ifndef SAMPLE_CVT_H
-#define SAMPLE_CVT_H
-
-#include "AL/al.h"
-#include "alBuffer.h"
-
-void ConvertData(ALvoid *dst, enum UserFmtType dstType, const ALvoid *src, enum UserFmtType srcType, ALsizei numchans, ALsizei len, ALsizei align);
-
-#endif /* SAMPLE_CVT_H */

OpenAL32/alAuxEffectSlot.c

@@ -1,727 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <math.h>
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alAuxEffectSlot.h"
-#include "alThunk.h"
-#include "alError.h"
-#include "alListener.h"
-#include "alSource.h"
-
-#include "almalloc.h"
-
-
-extern inline void LockEffectSlotsRead(ALCcontext *context);
-extern inline void UnlockEffectSlotsRead(ALCcontext *context);
-extern inline void LockEffectSlotsWrite(ALCcontext *context);
-extern inline void UnlockEffectSlotsWrite(ALCcontext *context);
-extern inline struct ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id);
-extern inline struct ALeffectslot *RemoveEffectSlot(ALCcontext *context, ALuint id);
-
-static UIntMap EffectStateFactoryMap;
-static inline ALeffectStateFactory *getFactoryByType(ALenum type)
-{
-    ALeffectStateFactory* (*getFactory)(void) = LookupUIntMapKey(&EffectStateFactoryMap, type);
-    if(getFactory != NULL)
-        return getFactory();
-    return NULL;
-}
-
-static void ALeffectState_IncRef(ALeffectState *state);
-static void ALeffectState_DecRef(ALeffectState *state);
-
-#define DO_UPDATEPROPS() do {                                                 \
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))          \
-        UpdateEffectSlotProps(slot);                                          \
-    else                                                                      \
-        ATOMIC_FLAG_CLEAR(&slot->PropsClean, almemory_order_release);         \
-} while(0)
-
-
-AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots)
-{
-    ALCcontext *context;
-    ALeffectslot **tmpslots = NULL;
-    ALsizei cur;
-    ALenum err;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(n >= 0))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    tmpslots = al_malloc(DEF_ALIGN, sizeof(ALeffectslot*)*n);
-
-    LockEffectSlotsWrite(context);
-    for(cur = 0;cur < n;cur++)
-    {
-        ALeffectslot *slot = al_calloc(16, sizeof(ALeffectslot));
-        err = AL_OUT_OF_MEMORY;
-        if(!slot || (err=InitEffectSlot(slot)) != AL_NO_ERROR)
-        {
-            al_free(slot);
-            UnlockEffectSlotsWrite(context);
-
-            alDeleteAuxiliaryEffectSlots(cur, effectslots);
-            SET_ERROR_AND_GOTO(context, err, done);
-        }
-
-        err = NewThunkEntry(&slot->id);
-        if(err == AL_NO_ERROR)
-            err = InsertUIntMapEntryNoLock(&context->EffectSlotMap, slot->id, slot);
-        if(err != AL_NO_ERROR)
-        {
-            FreeThunkEntry(slot->id);
-            ALeffectState_DecRef(slot->Effect.State);
-            if(slot->Params.EffectState)
-                ALeffectState_DecRef(slot->Params.EffectState);
-            al_free(slot);
-            UnlockEffectSlotsWrite(context);
-
-            alDeleteAuxiliaryEffectSlots(cur, effectslots);
-            SET_ERROR_AND_GOTO(context, err, done);
-        }
-
-        aluInitEffectPanning(slot);
-
-        tmpslots[cur] = slot;
-        effectslots[cur] = slot->id;
-    }
-    if(n > 0)
-    {
-        struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire);
-        struct ALeffectslotArray *newarray = NULL;
-        ALsizei newcount = curarray->count + n;
-        ALCdevice *device;
-
-        newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
-        newarray->count = newcount;
-        memcpy(newarray->slot, tmpslots, sizeof(ALeffectslot*)*n);
-        if(curarray)
-            memcpy(newarray->slot+n, curarray->slot, sizeof(ALeffectslot*)*curarray->count);
-
-        newarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray,
-                                       almemory_order_acq_rel);
-        device = context->Device;
-        while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
-            althrd_yield();
-        al_free(newarray);
-    }
-    UnlockEffectSlotsWrite(context);
-
-done:
-    al_free(tmpslots);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots)
-{
-    ALCcontext *context;
-    ALeffectslot *slot;
-    ALsizei i;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    LockEffectSlotsWrite(context);
-    if(!(n >= 0))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    for(i = 0;i < n;i++)
-    {
-        if((slot=LookupEffectSlot(context, effectslots[i])) == NULL)
-            SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-        if(ReadRef(&slot->ref) != 0)
-            SET_ERROR_AND_GOTO(context, AL_INVALID_OPERATION, done);
-    }
-
-    // All effectslots are valid
-    if(n > 0)
-    {
-        struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire);
-        struct ALeffectslotArray *newarray = NULL;
-        ALsizei newcount = curarray->count - n;
-        ALCdevice *device;
-        ALsizei j, k;
-
-        assert(newcount >= 0);
-        newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
-        newarray->count = newcount;
-        for(i = j = 0;i < newarray->count;)
-        {
-            slot = curarray->slot[j++];
-            for(k = 0;k < n;k++)
-            {
-                if(slot->id == effectslots[k])
-                    break;
-            }
-            if(k == n)
-                newarray->slot[i++] = slot;
-        }
-
-        newarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray,
-                                       almemory_order_acq_rel);
-        device = context->Device;
-        while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
-            althrd_yield();
-        al_free(newarray);
-    }
-
-    for(i = 0;i < n;i++)
-    {
-        if((slot=RemoveEffectSlot(context, effectslots[i])) == NULL)
-            continue;
-        FreeThunkEntry(slot->id);
-
-        DeinitEffectSlot(slot);
-
-        memset(slot, 0, sizeof(*slot));
-        al_free(slot);
-    }
-
-done:
-    UnlockEffectSlotsWrite(context);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot)
-{
-    ALCcontext *context;
-    ALboolean  ret;
-
-    context = GetContextRef();
-    if(!context) return AL_FALSE;
-
-    LockEffectSlotsRead(context);
-    ret = (LookupEffectSlot(context, effectslot) ? AL_TRUE : AL_FALSE);
-    UnlockEffectSlotsRead(context);
-
-    ALCcontext_DecRef(context);
-
-    return ret;
-}
-
-AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint value)
-{
-    ALCdevice *device;
-    ALCcontext *context;
-    ALeffectslot *slot;
-    ALeffect *effect = NULL;
-    ALenum err;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    LockEffectSlotsRead(context);
-    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    case AL_EFFECTSLOT_EFFECT:
-        device = context->Device;
-
-        LockEffectsRead(device);
-        effect = (value ? LookupEffect(device, value) : NULL);
-        if(!(value == 0 || effect != NULL))
-        {
-            UnlockEffectsRead(device);
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        }
-        err = InitializeEffect(device, slot, effect);
-        UnlockEffectsRead(device);
-
-        if(err != AL_NO_ERROR)
-            SET_ERROR_AND_GOTO(context, err, done);
-        break;
-
-    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
-        if(!(value == AL_TRUE || value == AL_FALSE))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        slot->AuxSendAuto = value;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    DO_UPDATEPROPS();
-
-done:
-    UnlockEffectSlotsRead(context);
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *values)
-{
-    ALCcontext *context;
-
-    switch(param)
-    {
-    case AL_EFFECTSLOT_EFFECT:
-    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
-        alAuxiliaryEffectSloti(effectslot, param, values[0]);
-        return;
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    LockEffectSlotsRead(context);
-    if(LookupEffectSlot(context, effectslot) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    UnlockEffectSlotsRead(context);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat value)
-{
-    ALCcontext *context;
-    ALeffectslot *slot;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    LockEffectSlotsRead(context);
-    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    case AL_EFFECTSLOT_GAIN:
-        if(!(value >= 0.0f && value <= 1.0f))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        slot->Gain = value;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    DO_UPDATEPROPS();
-
-done:
-    UnlockEffectSlotsRead(context);
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *values)
-{
-    ALCcontext *context;
-
-    switch(param)
-    {
-    case AL_EFFECTSLOT_GAIN:
-        alAuxiliaryEffectSlotf(effectslot, param, values[0]);
-        return;
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    LockEffectSlotsRead(context);
-    if(LookupEffectSlot(context, effectslot) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    UnlockEffectSlotsRead(context);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *value)
-{
-    ALCcontext *context;
-    ALeffectslot *slot;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    LockEffectSlotsRead(context);
-    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
-        *value = slot->AuxSendAuto;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    UnlockEffectSlotsRead(context);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *values)
-{
-    ALCcontext *context;
-
-    switch(param)
-    {
-    case AL_EFFECTSLOT_EFFECT:
-    case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
-        alGetAuxiliaryEffectSloti(effectslot, param, values);
-        return;
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    LockEffectSlotsRead(context);
-    if(LookupEffectSlot(context, effectslot) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    UnlockEffectSlotsRead(context);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *value)
-{
-    ALCcontext *context;
-    ALeffectslot *slot;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    LockEffectSlotsRead(context);
-    if((slot=LookupEffectSlot(context, effectslot)) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    case AL_EFFECTSLOT_GAIN:
-        *value = slot->Gain;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    UnlockEffectSlotsRead(context);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *values)
-{
-    ALCcontext *context;
-
-    switch(param)
-    {
-    case AL_EFFECTSLOT_GAIN:
-        alGetAuxiliaryEffectSlotf(effectslot, param, values);
-        return;
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    LockEffectSlotsRead(context);
-    if(LookupEffectSlot(context, effectslot) == NULL)
-        SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    UnlockEffectSlotsRead(context);
-    ALCcontext_DecRef(context);
-}
-
-
-void InitEffectFactoryMap(void)
-{
-    InitUIntMap(&EffectStateFactoryMap, INT_MAX);
-
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_NULL, ALnullStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_EAXREVERB, ALreverbStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_REVERB, ALreverbStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_CHORUS, ALchorusStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_COMPRESSOR, ALcompressorStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_DISTORTION, ALdistortionStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_ECHO, ALechoStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_EQUALIZER, ALequalizerStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_FLANGER, ALflangerStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_RING_MODULATOR, ALmodulatorStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_DEDICATED_DIALOGUE, ALdedicatedStateFactory_getFactory);
-    InsertUIntMapEntry(&EffectStateFactoryMap, AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT, ALdedicatedStateFactory_getFactory);
-}
-
-void DeinitEffectFactoryMap(void)
-{
-    ResetUIntMap(&EffectStateFactoryMap);
-}
-
-
-ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect)
-{
-    ALenum newtype = (effect ? effect->type : AL_EFFECT_NULL);
-    struct ALeffectslotProps *props;
-    ALeffectState *State;
-
-    if(newtype != EffectSlot->Effect.Type)
-    {
-        ALeffectStateFactory *factory;
-
-        factory = getFactoryByType(newtype);
-        if(!factory)
-        {
-            ERR("Failed to find factory for effect type 0x%04x\n", newtype);
-            return AL_INVALID_ENUM;
-        }
-        State = V0(factory,create)();
-        if(!State) return AL_OUT_OF_MEMORY;
-
-        START_MIXER_MODE();
-        almtx_lock(&Device->BackendLock);
-        State->OutBuffer = Device->Dry.Buffer;
-        State->OutChannels = Device->Dry.NumChannels;
-        if(V(State,deviceUpdate)(Device) == AL_FALSE)
-        {
-            almtx_unlock(&Device->BackendLock);
-            LEAVE_MIXER_MODE();
-            ALeffectState_DecRef(State);
-            return AL_OUT_OF_MEMORY;
-        }
-        almtx_unlock(&Device->BackendLock);
-        END_MIXER_MODE();
-
-        if(!effect)
-        {
-            EffectSlot->Effect.Type = AL_EFFECT_NULL;
-            memset(&EffectSlot->Effect.Props, 0, sizeof(EffectSlot->Effect.Props));
-        }
-        else
-        {
-            EffectSlot->Effect.Type = effect->type;
-            EffectSlot->Effect.Props = effect->Props;
-        }
-
-        ALeffectState_DecRef(EffectSlot->Effect.State);
-        EffectSlot->Effect.State = State;
-    }
-    else if(effect)
-        EffectSlot->Effect.Props = effect->Props;
-
-    /* Remove state references from old effect slot property updates. */
-    props = ATOMIC_LOAD_SEQ(&EffectSlot->FreeList);
-    while(props)
-    {
-        if(props->State)
-            ALeffectState_DecRef(props->State);
-        props->State = NULL;
-        props = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
-    }
-
-    return AL_NO_ERROR;
-}
-
-
-static void ALeffectState_IncRef(ALeffectState *state)
-{
-    uint ref;
-    ref = IncrementRef(&state->Ref);
-    TRACEREF("%p increasing refcount to %u\n", state, ref);
-}
-
-static void ALeffectState_DecRef(ALeffectState *state)
-{
-    uint ref;
-    ref = DecrementRef(&state->Ref);
-    TRACEREF("%p decreasing refcount to %u\n", state, ref);
-    if(ref == 0) DELETE_OBJ(state);
-}
-
-
-void ALeffectState_Construct(ALeffectState *state)
-{
-    InitRef(&state->Ref, 1);
-
-    state->OutBuffer = NULL;
-    state->OutChannels = 0;
-}
-
-void ALeffectState_Destruct(ALeffectState *UNUSED(state))
-{
-}
-
-
-ALenum InitEffectSlot(ALeffectslot *slot)
-{
-    ALeffectStateFactory *factory;
-
-    slot->Effect.Type = AL_EFFECT_NULL;
-
-    factory = getFactoryByType(AL_EFFECT_NULL);
-    if(!(slot->Effect.State=V0(factory,create)()))
-        return AL_OUT_OF_MEMORY;
-
-    slot->Gain = 1.0;
-    slot->AuxSendAuto = AL_TRUE;
-    ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_relaxed);
-    InitRef(&slot->ref, 0);
-
-    ATOMIC_INIT(&slot->Update, NULL);
-    ATOMIC_INIT(&slot->FreeList, NULL);
-
-    slot->Params.Gain = 1.0f;
-    slot->Params.AuxSendAuto = AL_TRUE;
-    ALeffectState_IncRef(slot->Effect.State);
-    slot->Params.EffectState = slot->Effect.State;
-    slot->Params.RoomRolloff = 0.0f;
-    slot->Params.DecayTime = 0.0f;
-    slot->Params.DecayHFRatio = 0.0f;
-    slot->Params.DecayHFLimit = AL_FALSE;
-    slot->Params.AirAbsorptionGainHF = 1.0f;
-
-    return AL_NO_ERROR;
-}
-
-void DeinitEffectSlot(ALeffectslot *slot)
-{
-    struct ALeffectslotProps *props;
-    size_t count = 0;
-
-    props = ATOMIC_LOAD_SEQ(&slot->Update);
-    if(props)
-    {
-        if(props->State) ALeffectState_DecRef(props->State);
-        TRACE("Freed unapplied AuxiliaryEffectSlot update %p\n", props);
-        al_free(props);
-    }
-    props = ATOMIC_LOAD(&slot->FreeList, almemory_order_relaxed);
-    while(props)
-    {
-        struct ALeffectslotProps *next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
-        if(props->State) ALeffectState_DecRef(props->State);
-        al_free(props);
-        props = next;
-        ++count;
-    }
-    TRACE("Freed "SZFMT" AuxiliaryEffectSlot property object%s\n", count, (count==1)?"":"s");
-
-    ALeffectState_DecRef(slot->Effect.State);
-    if(slot->Params.EffectState)
-        ALeffectState_DecRef(slot->Params.EffectState);
-}
-
-void UpdateEffectSlotProps(ALeffectslot *slot)
-{
-    struct ALeffectslotProps *props;
-    ALeffectState *oldstate;
-
-    /* Get an unused property container, or allocate a new one as needed. */
-    props = ATOMIC_LOAD(&slot->FreeList, almemory_order_relaxed);
-    if(!props)
-        props = al_calloc(16, sizeof(*props));
-    else
-    {
-        struct ALeffectslotProps *next;
-        do {
-            next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
-        } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&slot->FreeList, &props, next,
-                almemory_order_seq_cst, almemory_order_acquire) == 0);
-    }
-
-    /* Copy in current property values. */
-    props->Gain = slot->Gain;
-    props->AuxSendAuto = slot->AuxSendAuto;
-
-    props->Type = slot->Effect.Type;
-    props->Props = slot->Effect.Props;
-    /* Swap out any stale effect state object there may be in the container, to
-     * delete it.
-     */
-    ALeffectState_IncRef(slot->Effect.State);
-    oldstate = props->State;
-    props->State = slot->Effect.State;
-
-    /* Set the new container for updating internal parameters. */
-    props = ATOMIC_EXCHANGE_PTR(&slot->Update, props, almemory_order_acq_rel);
-    if(props)
-    {
-        /* If there was an unused update container, put it back in the
-         * freelist.
-         */
-        ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &slot->FreeList, props);
-    }
-
-    if(oldstate)
-        ALeffectState_DecRef(oldstate);
-}
-
-void UpdateAllEffectSlotProps(ALCcontext *context)
-{
-    struct ALeffectslotArray *auxslots;
-    ALsizei i;
-
-    LockEffectSlotsRead(context);
-    auxslots = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire);
-    for(i = 0;i < auxslots->count;i++)
-    {
-        ALeffectslot *slot = auxslots->slot[i];
-        if(!ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_acq_rel))
-            UpdateEffectSlotProps(slot);
-    }
-    UnlockEffectSlotsRead(context);
-}
-
-ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context)
-{
-    ALsizei pos;
-    for(pos = 0;pos < Context->EffectSlotMap.size;pos++)
-    {
-        ALeffectslot *temp = Context->EffectSlotMap.values[pos];
-        Context->EffectSlotMap.values[pos] = NULL;
-
-        DeinitEffectSlot(temp);
-
-        FreeThunkEntry(temp->id);
-        memset(temp, 0, sizeof(ALeffectslot));
-        al_free(temp);
-    }
-}

OpenAL32/alEffect.c

@@ -1,716 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <math.h>
-#include <float.h>
-
-#include "AL/al.h"
-#include "AL/alc.h"
-#include "alMain.h"
-#include "alEffect.h"
-#include "alThunk.h"
-#include "alError.h"
-
-
-ALboolean DisabledEffects[MAX_EFFECTS];
-
-extern inline void LockEffectsRead(ALCdevice *device);
-extern inline void UnlockEffectsRead(ALCdevice *device);
-extern inline void LockEffectsWrite(ALCdevice *device);
-extern inline void UnlockEffectsWrite(ALCdevice *device);
-extern inline struct ALeffect *LookupEffect(ALCdevice *device, ALuint id);
-extern inline struct ALeffect *RemoveEffect(ALCdevice *device, ALuint id);
-extern inline ALboolean IsReverbEffect(ALenum type);
-
-static void InitEffectParams(ALeffect *effect, ALenum type);
-
-
-AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects)
-{
-    ALCdevice *device;
-    ALCcontext *context;
-    ALsizei cur;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(n >= 0))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-
-    device = context->Device;
-    for(cur = 0;cur < n;cur++)
-    {
-        ALeffect *effect = al_calloc(16, sizeof(ALeffect));
-        ALenum err = AL_OUT_OF_MEMORY;
-        if(!effect || (err=InitEffect(effect)) != AL_NO_ERROR)
-        {
-            al_free(effect);
-            alDeleteEffects(cur, effects);
-            SET_ERROR_AND_GOTO(context, err, done);
-        }
-
-        err = NewThunkEntry(&effect->id);
-        if(err == AL_NO_ERROR)
-            err = InsertUIntMapEntry(&device->EffectMap, effect->id, effect);
-        if(err != AL_NO_ERROR)
-        {
-            FreeThunkEntry(effect->id);
-            memset(effect, 0, sizeof(ALeffect));
-            al_free(effect);
-
-            alDeleteEffects(cur, effects);
-            SET_ERROR_AND_GOTO(context, err, done);
-        }
-
-        effects[cur] = effect->id;
-    }
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects)
-{
-    ALCdevice *device;
-    ALCcontext *context;
-    ALeffect *effect;
-    ALsizei i;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    device = context->Device;
-    LockEffectsWrite(device);
-    if(!(n >= 0))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    for(i = 0;i < n;i++)
-    {
-        if(effects[i] && LookupEffect(device, effects[i]) == NULL)
-            SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    }
-    for(i = 0;i < n;i++)
-    {
-        if((effect=RemoveEffect(device, effects[i])) == NULL)
-            continue;
-        FreeThunkEntry(effect->id);
-
-        memset(effect, 0, sizeof(*effect));
-        al_free(effect);
-    }
-
-done:
-    UnlockEffectsWrite(device);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect)
-{
-    ALCcontext *Context;
-    ALboolean  result;
-
-    Context = GetContextRef();
-    if(!Context) return AL_FALSE;
-
-    LockEffectsRead(Context->Device);
-    result = ((!effect || LookupEffect(Context->Device, effect)) ?
-              AL_TRUE : AL_FALSE);
-    UnlockEffectsRead(Context->Device);
-
-    ALCcontext_DecRef(Context);
-
-    return result;
-}
-
-AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsWrite(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        if(param == AL_EFFECT_TYPE)
-        {
-            ALboolean isOk = (value == AL_EFFECT_NULL);
-            ALint i;
-            for(i = 0;!isOk && EffectList[i].val;i++)
-            {
-                if(value == EffectList[i].val &&
-                   !DisabledEffects[EffectList[i].type])
-                    isOk = AL_TRUE;
-            }
-
-            if(isOk)
-                InitEffectParams(ALEffect, value);
-            else
-                alSetError(Context, AL_INVALID_VALUE);
-        }
-        else
-        {
-            /* Call the appropriate handler */
-            V(ALEffect,setParami)(Context, param, value);
-        }
-    }
-    UnlockEffectsWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    switch(param)
-    {
-        case AL_EFFECT_TYPE:
-            alEffecti(effect, param, values[0]);
-            return;
-    }
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsWrite(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        V(ALEffect,setParamiv)(Context, param, values);
-    }
-    UnlockEffectsWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsWrite(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        V(ALEffect,setParamf)(Context, param, value);
-    }
-    UnlockEffectsWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsWrite(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        V(ALEffect,setParamfv)(Context, param, values);
-    }
-    UnlockEffectsWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsRead(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        if(param == AL_EFFECT_TYPE)
-            *value = ALEffect->type;
-        else
-        {
-            /* Call the appropriate handler */
-            V(ALEffect,getParami)(Context, param, value);
-        }
-    }
-    UnlockEffectsRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    switch(param)
-    {
-        case AL_EFFECT_TYPE:
-            alGetEffecti(effect, param, values);
-            return;
-    }
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsRead(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        V(ALEffect,getParamiv)(Context, param, values);
-    }
-    UnlockEffectsRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsRead(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        V(ALEffect,getParamf)(Context, param, value);
-    }
-    UnlockEffectsRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALeffect   *ALEffect;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockEffectsRead(Device);
-    if((ALEffect=LookupEffect(Device, effect)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        V(ALEffect,getParamfv)(Context, param, values);
-    }
-    UnlockEffectsRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-
-ALenum InitEffect(ALeffect *effect)
-{
-    InitEffectParams(effect, AL_EFFECT_NULL);
-    return AL_NO_ERROR;
-}
-
-ALvoid ReleaseALEffects(ALCdevice *device)
-{
-    ALsizei i;
-    for(i = 0;i < device->EffectMap.size;i++)
-    {
-        ALeffect *temp = device->EffectMap.values[i];
-        device->EffectMap.values[i] = NULL;
-
-        // Release effect structure
-        FreeThunkEntry(temp->id);
-        memset(temp, 0, sizeof(ALeffect));
-        al_free(temp);
-    }
-}
-
-
-static void InitEffectParams(ALeffect *effect, ALenum type)
-{
-    switch(type)
-    {
-    case AL_EFFECT_EAXREVERB:
-        effect->Props.Reverb.Density   = AL_EAXREVERB_DEFAULT_DENSITY;
-        effect->Props.Reverb.Diffusion = AL_EAXREVERB_DEFAULT_DIFFUSION;
-        effect->Props.Reverb.Gain   = AL_EAXREVERB_DEFAULT_GAIN;
-        effect->Props.Reverb.GainHF = AL_EAXREVERB_DEFAULT_GAINHF;
-        effect->Props.Reverb.GainLF = AL_EAXREVERB_DEFAULT_GAINLF;
-        effect->Props.Reverb.DecayTime    = AL_EAXREVERB_DEFAULT_DECAY_TIME;
-        effect->Props.Reverb.DecayHFRatio = AL_EAXREVERB_DEFAULT_DECAY_HFRATIO;
-        effect->Props.Reverb.DecayLFRatio = AL_EAXREVERB_DEFAULT_DECAY_LFRATIO;
-        effect->Props.Reverb.ReflectionsGain   = AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN;
-        effect->Props.Reverb.ReflectionsDelay  = AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY;
-        effect->Props.Reverb.ReflectionsPan[0] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
-        effect->Props.Reverb.ReflectionsPan[1] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
-        effect->Props.Reverb.ReflectionsPan[2] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
-        effect->Props.Reverb.LateReverbGain   = AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN;
-        effect->Props.Reverb.LateReverbDelay  = AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY;
-        effect->Props.Reverb.LateReverbPan[0] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
-        effect->Props.Reverb.LateReverbPan[1] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
-        effect->Props.Reverb.LateReverbPan[2] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
-        effect->Props.Reverb.EchoTime  = AL_EAXREVERB_DEFAULT_ECHO_TIME;
-        effect->Props.Reverb.EchoDepth = AL_EAXREVERB_DEFAULT_ECHO_DEPTH;
-        effect->Props.Reverb.ModulationTime  = AL_EAXREVERB_DEFAULT_MODULATION_TIME;
-        effect->Props.Reverb.ModulationDepth = AL_EAXREVERB_DEFAULT_MODULATION_DEPTH;
-        effect->Props.Reverb.AirAbsorptionGainHF = AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF;
-        effect->Props.Reverb.HFReference = AL_EAXREVERB_DEFAULT_HFREFERENCE;
-        effect->Props.Reverb.LFReference = AL_EAXREVERB_DEFAULT_LFREFERENCE;
-        effect->Props.Reverb.RoomRolloffFactor = AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR;
-        effect->Props.Reverb.DecayHFLimit = AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT;
-        SET_VTABLE1(ALeaxreverb, effect);
-        break;
-    case AL_EFFECT_REVERB:
-        effect->Props.Reverb.Density   = AL_REVERB_DEFAULT_DENSITY;
-        effect->Props.Reverb.Diffusion = AL_REVERB_DEFAULT_DIFFUSION;
-        effect->Props.Reverb.Gain   = AL_REVERB_DEFAULT_GAIN;
-        effect->Props.Reverb.GainHF = AL_REVERB_DEFAULT_GAINHF;
-        effect->Props.Reverb.GainLF = 1.0f;
-        effect->Props.Reverb.DecayTime    = AL_REVERB_DEFAULT_DECAY_TIME;
-        effect->Props.Reverb.DecayHFRatio = AL_REVERB_DEFAULT_DECAY_HFRATIO;
-        effect->Props.Reverb.DecayLFRatio = 1.0f;
-        effect->Props.Reverb.ReflectionsGain   = AL_REVERB_DEFAULT_REFLECTIONS_GAIN;
-        effect->Props.Reverb.ReflectionsDelay  = AL_REVERB_DEFAULT_REFLECTIONS_DELAY;
-        effect->Props.Reverb.ReflectionsPan[0] = 0.0f;
-        effect->Props.Reverb.ReflectionsPan[1] = 0.0f;
-        effect->Props.Reverb.ReflectionsPan[2] = 0.0f;
-        effect->Props.Reverb.LateReverbGain   = AL_REVERB_DEFAULT_LATE_REVERB_GAIN;
-        effect->Props.Reverb.LateReverbDelay  = AL_REVERB_DEFAULT_LATE_REVERB_DELAY;
-        effect->Props.Reverb.LateReverbPan[0] = 0.0f;
-        effect->Props.Reverb.LateReverbPan[1] = 0.0f;
-        effect->Props.Reverb.LateReverbPan[2] = 0.0f;
-        effect->Props.Reverb.EchoTime  = 0.25f;
-        effect->Props.Reverb.EchoDepth = 0.0f;
-        effect->Props.Reverb.ModulationTime  = 0.25f;
-        effect->Props.Reverb.ModulationDepth = 0.0f;
-        effect->Props.Reverb.AirAbsorptionGainHF = AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF;
-        effect->Props.Reverb.HFReference = 5000.0f;
-        effect->Props.Reverb.LFReference = 250.0f;
-        effect->Props.Reverb.RoomRolloffFactor = AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR;
-        effect->Props.Reverb.DecayHFLimit = AL_REVERB_DEFAULT_DECAY_HFLIMIT;
-        SET_VTABLE1(ALreverb, effect);
-        break;
-    case AL_EFFECT_CHORUS:
-        effect->Props.Chorus.Waveform = AL_CHORUS_DEFAULT_WAVEFORM;
-        effect->Props.Chorus.Phase = AL_CHORUS_DEFAULT_PHASE;
-        effect->Props.Chorus.Rate = AL_CHORUS_DEFAULT_RATE;
-        effect->Props.Chorus.Depth = AL_CHORUS_DEFAULT_DEPTH;
-        effect->Props.Chorus.Feedback = AL_CHORUS_DEFAULT_FEEDBACK;
-        effect->Props.Chorus.Delay = AL_CHORUS_DEFAULT_DELAY;
-        SET_VTABLE1(ALchorus, effect);
-        break;
-    case AL_EFFECT_COMPRESSOR:
-        effect->Props.Compressor.OnOff = AL_COMPRESSOR_DEFAULT_ONOFF;
-        SET_VTABLE1(ALcompressor, effect);
-        break;
-    case AL_EFFECT_DISTORTION:
-        effect->Props.Distortion.Edge = AL_DISTORTION_DEFAULT_EDGE;
-        effect->Props.Distortion.Gain = AL_DISTORTION_DEFAULT_GAIN;
-        effect->Props.Distortion.LowpassCutoff = AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF;
-        effect->Props.Distortion.EQCenter = AL_DISTORTION_DEFAULT_EQCENTER;
-        effect->Props.Distortion.EQBandwidth = AL_DISTORTION_DEFAULT_EQBANDWIDTH;
-        SET_VTABLE1(ALdistortion, effect);
-        break;
-    case AL_EFFECT_ECHO:
-        effect->Props.Echo.Delay    = AL_ECHO_DEFAULT_DELAY;
-        effect->Props.Echo.LRDelay  = AL_ECHO_DEFAULT_LRDELAY;
-        effect->Props.Echo.Damping  = AL_ECHO_DEFAULT_DAMPING;
-        effect->Props.Echo.Feedback = AL_ECHO_DEFAULT_FEEDBACK;
-        effect->Props.Echo.Spread   = AL_ECHO_DEFAULT_SPREAD;
-        SET_VTABLE1(ALecho, effect);
-        break;
-    case AL_EFFECT_EQUALIZER:
-        effect->Props.Equalizer.LowCutoff = AL_EQUALIZER_DEFAULT_LOW_CUTOFF;
-        effect->Props.Equalizer.LowGain = AL_EQUALIZER_DEFAULT_LOW_GAIN;
-        effect->Props.Equalizer.Mid1Center = AL_EQUALIZER_DEFAULT_MID1_CENTER;
-        effect->Props.Equalizer.Mid1Gain = AL_EQUALIZER_DEFAULT_MID1_GAIN;
-        effect->Props.Equalizer.Mid1Width = AL_EQUALIZER_DEFAULT_MID1_WIDTH;
-        effect->Props.Equalizer.Mid2Center = AL_EQUALIZER_DEFAULT_MID2_CENTER;
-        effect->Props.Equalizer.Mid2Gain = AL_EQUALIZER_DEFAULT_MID2_GAIN;
-        effect->Props.Equalizer.Mid2Width = AL_EQUALIZER_DEFAULT_MID2_WIDTH;
-        effect->Props.Equalizer.HighCutoff = AL_EQUALIZER_DEFAULT_HIGH_CUTOFF;
-        effect->Props.Equalizer.HighGain = AL_EQUALIZER_DEFAULT_HIGH_GAIN;
-        SET_VTABLE1(ALequalizer, effect);
-        break;
-    case AL_EFFECT_FLANGER:
-        effect->Props.Flanger.Waveform = AL_FLANGER_DEFAULT_WAVEFORM;
-        effect->Props.Flanger.Phase = AL_FLANGER_DEFAULT_PHASE;
-        effect->Props.Flanger.Rate = AL_FLANGER_DEFAULT_RATE;
-        effect->Props.Flanger.Depth = AL_FLANGER_DEFAULT_DEPTH;
-        effect->Props.Flanger.Feedback = AL_FLANGER_DEFAULT_FEEDBACK;
-        effect->Props.Flanger.Delay = AL_FLANGER_DEFAULT_DELAY;
-        SET_VTABLE1(ALflanger, effect);
-        break;
-    case AL_EFFECT_RING_MODULATOR:
-        effect->Props.Modulator.Frequency      = AL_RING_MODULATOR_DEFAULT_FREQUENCY;
-        effect->Props.Modulator.HighPassCutoff = AL_RING_MODULATOR_DEFAULT_HIGHPASS_CUTOFF;
-        effect->Props.Modulator.Waveform       = AL_RING_MODULATOR_DEFAULT_WAVEFORM;
-        SET_VTABLE1(ALmodulator, effect);
-        break;
-    case AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT:
-    case AL_EFFECT_DEDICATED_DIALOGUE:
-        effect->Props.Dedicated.Gain = 1.0f;
-        SET_VTABLE1(ALdedicated, effect);
-        break;
-    default:
-        SET_VTABLE1(ALnull, effect);
-        break;
-    }
-    effect->type = type;
-}
-
-
-#include "AL/efx-presets.h"
-
-#define DECL(x) { #x, EFX_REVERB_PRESET_##x }
-static const struct {
-    const char name[32];
-    EFXEAXREVERBPROPERTIES props;
-} reverblist[] = {
-    DECL(GENERIC),
-    DECL(PADDEDCELL),
-    DECL(ROOM),
-    DECL(BATHROOM),
-    DECL(LIVINGROOM),
-    DECL(STONEROOM),
-    DECL(AUDITORIUM),
-    DECL(CONCERTHALL),
-    DECL(CAVE),
-    DECL(ARENA),
-    DECL(HANGAR),
-    DECL(CARPETEDHALLWAY),
-    DECL(HALLWAY),
-    DECL(STONECORRIDOR),
-    DECL(ALLEY),
-    DECL(FOREST),
-    DECL(CITY),
-    DECL(MOUNTAINS),
-    DECL(QUARRY),
-    DECL(PLAIN),
-    DECL(PARKINGLOT),
-    DECL(SEWERPIPE),
-    DECL(UNDERWATER),
-    DECL(DRUGGED),
-    DECL(DIZZY),
-    DECL(PSYCHOTIC),
-
-    DECL(CASTLE_SMALLROOM),
-    DECL(CASTLE_SHORTPASSAGE),
-    DECL(CASTLE_MEDIUMROOM),
-    DECL(CASTLE_LARGEROOM),
-    DECL(CASTLE_LONGPASSAGE),
-    DECL(CASTLE_HALL),
-    DECL(CASTLE_CUPBOARD),
-    DECL(CASTLE_COURTYARD),
-    DECL(CASTLE_ALCOVE),
-
-    DECL(FACTORY_SMALLROOM),
-    DECL(FACTORY_SHORTPASSAGE),
-    DECL(FACTORY_MEDIUMROOM),
-    DECL(FACTORY_LARGEROOM),
-    DECL(FACTORY_LONGPASSAGE),
-    DECL(FACTORY_HALL),
-    DECL(FACTORY_CUPBOARD),
-    DECL(FACTORY_COURTYARD),
-    DECL(FACTORY_ALCOVE),
-
-    DECL(ICEPALACE_SMALLROOM),
-    DECL(ICEPALACE_SHORTPASSAGE),
-    DECL(ICEPALACE_MEDIUMROOM),
-    DECL(ICEPALACE_LARGEROOM),
-    DECL(ICEPALACE_LONGPASSAGE),
-    DECL(ICEPALACE_HALL),
-    DECL(ICEPALACE_CUPBOARD),
-    DECL(ICEPALACE_COURTYARD),
-    DECL(ICEPALACE_ALCOVE),
-
-    DECL(SPACESTATION_SMALLROOM),
-    DECL(SPACESTATION_SHORTPASSAGE),
-    DECL(SPACESTATION_MEDIUMROOM),
-    DECL(SPACESTATION_LARGEROOM),
-    DECL(SPACESTATION_LONGPASSAGE),
-    DECL(SPACESTATION_HALL),
-    DECL(SPACESTATION_CUPBOARD),
-    DECL(SPACESTATION_ALCOVE),
-
-    DECL(WOODEN_SMALLROOM),
-    DECL(WOODEN_SHORTPASSAGE),
-    DECL(WOODEN_MEDIUMROOM),
-    DECL(WOODEN_LARGEROOM),
-    DECL(WOODEN_LONGPASSAGE),
-    DECL(WOODEN_HALL),
-    DECL(WOODEN_CUPBOARD),
-    DECL(WOODEN_COURTYARD),
-    DECL(WOODEN_ALCOVE),
-
-    DECL(SPORT_EMPTYSTADIUM),
-    DECL(SPORT_SQUASHCOURT),
-    DECL(SPORT_SMALLSWIMMINGPOOL),
-    DECL(SPORT_LARGESWIMMINGPOOL),
-    DECL(SPORT_GYMNASIUM),
-    DECL(SPORT_FULLSTADIUM),
-    DECL(SPORT_STADIUMTANNOY),
-
-    DECL(PREFAB_WORKSHOP),
-    DECL(PREFAB_SCHOOLROOM),
-    DECL(PREFAB_PRACTISEROOM),
-    DECL(PREFAB_OUTHOUSE),
-    DECL(PREFAB_CARAVAN),
-
-    DECL(DOME_TOMB),
-    DECL(PIPE_SMALL),
-    DECL(DOME_SAINTPAULS),
-    DECL(PIPE_LONGTHIN),
-    DECL(PIPE_LARGE),
-    DECL(PIPE_RESONANT),
-
-    DECL(OUTDOORS_BACKYARD),
-    DECL(OUTDOORS_ROLLINGPLAINS),
-    DECL(OUTDOORS_DEEPCANYON),
-    DECL(OUTDOORS_CREEK),
-    DECL(OUTDOORS_VALLEY),
-
-    DECL(MOOD_HEAVEN),
-    DECL(MOOD_HELL),
-    DECL(MOOD_MEMORY),
-
-    DECL(DRIVING_COMMENTATOR),
-    DECL(DRIVING_PITGARAGE),
-    DECL(DRIVING_INCAR_RACER),
-    DECL(DRIVING_INCAR_SPORTS),
-    DECL(DRIVING_INCAR_LUXURY),
-    DECL(DRIVING_FULLGRANDSTAND),
-    DECL(DRIVING_EMPTYGRANDSTAND),
-    DECL(DRIVING_TUNNEL),
-
-    DECL(CITY_STREETS),
-    DECL(CITY_SUBWAY),
-    DECL(CITY_MUSEUM),
-    DECL(CITY_LIBRARY),
-    DECL(CITY_UNDERPASS),
-    DECL(CITY_ABANDONED),
-
-    DECL(DUSTYROOM),
-    DECL(CHAPEL),
-    DECL(SMALLWATERROOM),
-};
-#undef DECL
-
-ALvoid LoadReverbPreset(const char *name, ALeffect *effect)
-{
-    size_t i;
-
-    if(strcasecmp(name, "NONE") == 0)
-    {
-        InitEffectParams(effect, AL_EFFECT_NULL);
-        TRACE("Loading reverb '%s'\n", "NONE");
-        return;
-    }
-
-    if(!DisabledEffects[AL__EAXREVERB])
-        InitEffectParams(effect, AL_EFFECT_EAXREVERB);
-    else if(!DisabledEffects[AL__REVERB])
-        InitEffectParams(effect, AL_EFFECT_REVERB);
-    else
-        InitEffectParams(effect, AL_EFFECT_NULL);
-    for(i = 0;i < COUNTOF(reverblist);i++)
-    {
-        const EFXEAXREVERBPROPERTIES *props;
-
-        if(strcasecmp(name, reverblist[i].name) != 0)
-            continue;
-
-        TRACE("Loading reverb '%s'\n", reverblist[i].name);
-        props = &reverblist[i].props;
-        effect->Props.Reverb.Density   = props->flDensity;
-        effect->Props.Reverb.Diffusion = props->flDiffusion;
-        effect->Props.Reverb.Gain   = props->flGain;
-        effect->Props.Reverb.GainHF = props->flGainHF;
-        effect->Props.Reverb.GainLF = props->flGainLF;
-        effect->Props.Reverb.DecayTime    = props->flDecayTime;
-        effect->Props.Reverb.DecayHFRatio = props->flDecayHFRatio;
-        effect->Props.Reverb.DecayLFRatio = props->flDecayLFRatio;
-        effect->Props.Reverb.ReflectionsGain   = props->flReflectionsGain;
-        effect->Props.Reverb.ReflectionsDelay  = props->flReflectionsDelay;
-        effect->Props.Reverb.ReflectionsPan[0] = props->flReflectionsPan[0];
-        effect->Props.Reverb.ReflectionsPan[1] = props->flReflectionsPan[1];
-        effect->Props.Reverb.ReflectionsPan[2] = props->flReflectionsPan[2];
-        effect->Props.Reverb.LateReverbGain   = props->flLateReverbGain;
-        effect->Props.Reverb.LateReverbDelay  = props->flLateReverbDelay;
-        effect->Props.Reverb.LateReverbPan[0] = props->flLateReverbPan[0];
-        effect->Props.Reverb.LateReverbPan[1] = props->flLateReverbPan[1];
-        effect->Props.Reverb.LateReverbPan[2] = props->flLateReverbPan[2];
-        effect->Props.Reverb.EchoTime  = props->flEchoTime;
-        effect->Props.Reverb.EchoDepth = props->flEchoDepth;
-        effect->Props.Reverb.ModulationTime  = props->flModulationTime;
-        effect->Props.Reverb.ModulationDepth = props->flModulationDepth;
-        effect->Props.Reverb.AirAbsorptionGainHF = props->flAirAbsorptionGainHF;
-        effect->Props.Reverb.HFReference = props->flHFReference;
-        effect->Props.Reverb.LFReference = props->flLFReference;
-        effect->Props.Reverb.RoomRolloffFactor = props->flRoomRolloffFactor;
-        effect->Props.Reverb.DecayHFLimit = props->iDecayHFLimit;
-        return;
-    }
-
-    WARN("Reverb preset '%s' not found\n", name);
-}

OpenAL32/alError.c

@@ -1,82 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2000 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <signal.h>
-
-#ifdef HAVE_WINDOWS_H
-#define WIN32_LEAN_AND_MEAN
-#include <windows.h>
-#endif
-
-#include "alMain.h"
-#include "AL/alc.h"
-#include "alError.h"
-
-ALboolean TrapALError = AL_FALSE;
-
-ALvoid alSetError(ALCcontext *Context, ALenum errorCode)
-{
-    ALenum curerr = AL_NO_ERROR;
-
-    WARN("Error generated on context %p, code 0x%04x\n", Context, errorCode);
-    if(TrapALError)
-    {
-#ifdef _WIN32
-        /* DebugBreak will cause an exception if there is no debugger */
-        if(IsDebuggerPresent())
-            DebugBreak();
-#elif defined(SIGTRAP)
-        raise(SIGTRAP);
-#endif
-    }
-
-    (void)(ATOMIC_COMPARE_EXCHANGE_STRONG_SEQ(&Context->LastError, &curerr, errorCode));
-}
-
-AL_API ALenum AL_APIENTRY alGetError(void)
-{
-    ALCcontext *Context;
-    ALenum errorCode;
-
-    Context = GetContextRef();
-    if(!Context)
-    {
-        WARN("Querying error state on null context (implicitly 0x%04x)\n",
-             AL_INVALID_OPERATION);
-        if(TrapALError)
-        {
-#ifdef _WIN32
-            if(IsDebuggerPresent())
-                DebugBreak();
-#elif defined(SIGTRAP)
-            raise(SIGTRAP);
-#endif
-        }
-        return AL_INVALID_OPERATION;
-    }
-
-    errorCode = ATOMIC_EXCHANGE_SEQ(&Context->LastError, AL_NO_ERROR);
-
-    ALCcontext_DecRef(Context);
-
-    return errorCode;
-}

OpenAL32/alExtension.c

@@ -1,103 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-#include <string.h>
-#include <ctype.h>
-
-#include "alError.h"
-#include "alMain.h"
-#include "alFilter.h"
-#include "alEffect.h"
-#include "alAuxEffectSlot.h"
-#include "alSource.h"
-#include "alBuffer.h"
-#include "AL/al.h"
-#include "AL/alc.h"
-
-
-const struct EffectList EffectList[] = {
-    { "eaxreverb",  AL__EAXREVERB,  "AL_EFFECT_EAXREVERB",      AL_EFFECT_EAXREVERB },
-    { "reverb",     AL__REVERB,     "AL_EFFECT_REVERB",         AL_EFFECT_REVERB },
-    { "chorus",     AL__CHORUS,     "AL_EFFECT_CHORUS",         AL_EFFECT_CHORUS },
-    { "compressor", AL__COMPRESSOR, "AL_EFFECT_COMPRESSOR",     AL_EFFECT_COMPRESSOR },
-    { "distortion", AL__DISTORTION, "AL_EFFECT_DISTORTION",     AL_EFFECT_DISTORTION },
-    { "echo",       AL__ECHO,       "AL_EFFECT_ECHO",           AL_EFFECT_ECHO },
-    { "equalizer",  AL__EQUALIZER,  "AL_EFFECT_EQUALIZER",      AL_EFFECT_EQUALIZER },
-    { "flanger",    AL__FLANGER,    "AL_EFFECT_FLANGER",        AL_EFFECT_FLANGER },
-    { "modulator",  AL__MODULATOR,  "AL_EFFECT_RING_MODULATOR", AL_EFFECT_RING_MODULATOR },
-    { "dedicated",  AL__DEDICATED,  "AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT", AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
-    { "dedicated",  AL__DEDICATED,  "AL_EFFECT_DEDICATED_DIALOGUE", AL_EFFECT_DEDICATED_DIALOGUE },
-    { NULL, 0, NULL, (ALenum)0 }
-};
-
-
-AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName)
-{
-    ALboolean ret = AL_FALSE;
-    ALCcontext *context;
-    const char *ptr;
-    size_t len;
-
-    context = GetContextRef();
-    if(!context) return AL_FALSE;
-
-    if(!(extName))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-
-    len = strlen(extName);
-    ptr = context->ExtensionList;
-    while(ptr && *ptr)
-    {
-        if(strncasecmp(ptr, extName, len) == 0 &&
-           (ptr[len] == '\0' || isspace(ptr[len])))
-        {
-            ret = AL_TRUE;
-            break;
-        }
-        if((ptr=strchr(ptr, ' ')) != NULL)
-        {
-            do {
-                ++ptr;
-            } while(isspace(*ptr));
-        }
-    }
-
-done:
-    ALCcontext_DecRef(context);
-    return ret;
-}
-
-
-AL_API ALvoid* AL_APIENTRY alGetProcAddress(const ALchar *funcName)
-{
-    if(!funcName)
-        return NULL;
-    return alcGetProcAddress(NULL, funcName);
-}
-
-AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *enumName)
-{
-    if(!enumName)
-        return (ALenum)0;
-    return alcGetEnumValue(NULL, enumName);
-}

OpenAL32/alFilter.c

@@ -1,719 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alu.h"
-#include "alFilter.h"
-#include "alThunk.h"
-#include "alError.h"
-
-
-extern inline void LockFiltersRead(ALCdevice *device);
-extern inline void UnlockFiltersRead(ALCdevice *device);
-extern inline void LockFiltersWrite(ALCdevice *device);
-extern inline void UnlockFiltersWrite(ALCdevice *device);
-extern inline struct ALfilter *LookupFilter(ALCdevice *device, ALuint id);
-extern inline struct ALfilter *RemoveFilter(ALCdevice *device, ALuint id);
-extern inline void ALfilterState_clear(ALfilterState *filter);
-extern inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src);
-extern inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples);
-extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope);
-extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat freq_mult, ALfloat bandwidth);
-
-static void InitFilterParams(ALfilter *filter, ALenum type);
-
-
-AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
-{
-    ALCdevice *device;
-    ALCcontext *context;
-    ALsizei cur = 0;
-    ALenum err;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(n >= 0))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-
-    device = context->Device;
-    for(cur = 0;cur < n;cur++)
-    {
-        ALfilter *filter = al_calloc(16, sizeof(ALfilter));
-        if(!filter)
-        {
-            alDeleteFilters(cur, filters);
-            SET_ERROR_AND_GOTO(context, AL_OUT_OF_MEMORY, done);
-        }
-        InitFilterParams(filter, AL_FILTER_NULL);
-
-        err = NewThunkEntry(&filter->id);
-        if(err == AL_NO_ERROR)
-            err = InsertUIntMapEntry(&device->FilterMap, filter->id, filter);
-        if(err != AL_NO_ERROR)
-        {
-            FreeThunkEntry(filter->id);
-            memset(filter, 0, sizeof(ALfilter));
-            al_free(filter);
-
-            alDeleteFilters(cur, filters);
-            SET_ERROR_AND_GOTO(context, err, done);
-        }
-
-        filters[cur] = filter->id;
-    }
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters)
-{
-    ALCdevice *device;
-    ALCcontext *context;
-    ALfilter *filter;
-    ALsizei i;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    device = context->Device;
-    LockFiltersWrite(device);
-    if(!(n >= 0))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    for(i = 0;i < n;i++)
-    {
-        if(filters[i] && LookupFilter(device, filters[i]) == NULL)
-            SET_ERROR_AND_GOTO(context, AL_INVALID_NAME, done);
-    }
-    for(i = 0;i < n;i++)
-    {
-        if((filter=RemoveFilter(device, filters[i])) == NULL)
-            continue;
-        FreeThunkEntry(filter->id);
-
-        memset(filter, 0, sizeof(*filter));
-        al_free(filter);
-    }
-
-done:
-    UnlockFiltersWrite(device);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter)
-{
-    ALCcontext *Context;
-    ALboolean  result;
-
-    Context = GetContextRef();
-    if(!Context) return AL_FALSE;
-
-    LockFiltersRead(Context->Device);
-    result = ((!filter || LookupFilter(Context->Device, filter)) ?
-              AL_TRUE : AL_FALSE);
-    UnlockFiltersRead(Context->Device);
-
-    ALCcontext_DecRef(Context);
-
-    return result;
-}
-
-AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersWrite(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        if(param == AL_FILTER_TYPE)
-        {
-            if(value == AL_FILTER_NULL || value == AL_FILTER_LOWPASS ||
-               value == AL_FILTER_HIGHPASS || value == AL_FILTER_BANDPASS)
-                InitFilterParams(ALFilter, value);
-            else
-                alSetError(Context, AL_INVALID_VALUE);
-        }
-        else
-        {
-            /* Call the appropriate handler */
-            ALfilter_SetParami(ALFilter, Context, param, value);
-        }
-    }
-    UnlockFiltersWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    switch(param)
-    {
-        case AL_FILTER_TYPE:
-            alFilteri(filter, param, values[0]);
-            return;
-    }
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersWrite(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        ALfilter_SetParamiv(ALFilter, Context, param, values);
-    }
-    UnlockFiltersWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersWrite(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        ALfilter_SetParamf(ALFilter, Context, param, value);
-    }
-    UnlockFiltersWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersWrite(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        ALfilter_SetParamfv(ALFilter, Context, param, values);
-    }
-    UnlockFiltersWrite(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersRead(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        if(param == AL_FILTER_TYPE)
-            *value = ALFilter->type;
-        else
-        {
-            /* Call the appropriate handler */
-            ALfilter_GetParami(ALFilter, Context, param, value);
-        }
-    }
-    UnlockFiltersRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    switch(param)
-    {
-        case AL_FILTER_TYPE:
-            alGetFilteri(filter, param, values);
-            return;
-    }
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersRead(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        ALfilter_GetParamiv(ALFilter, Context, param, values);
-    }
-    UnlockFiltersRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *value)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersRead(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        ALfilter_GetParamf(ALFilter, Context, param, value);
-    }
-    UnlockFiltersRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *values)
-{
-    ALCcontext *Context;
-    ALCdevice  *Device;
-    ALfilter   *ALFilter;
-
-    Context = GetContextRef();
-    if(!Context) return;
-
-    Device = Context->Device;
-    LockFiltersRead(Device);
-    if((ALFilter=LookupFilter(Device, filter)) == NULL)
-        alSetError(Context, AL_INVALID_NAME);
-    else
-    {
-        /* Call the appropriate handler */
-        ALfilter_GetParamfv(ALFilter, Context, param, values);
-    }
-    UnlockFiltersRead(Device);
-
-    ALCcontext_DecRef(Context);
-}
-
-
-void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_mult, ALfloat rcpQ)
-{
-    ALfloat alpha, sqrtgain_alpha_2;
-    ALfloat w0, sin_w0, cos_w0;
-    ALfloat a[3] = { 1.0f, 0.0f, 0.0f };
-    ALfloat b[3] = { 1.0f, 0.0f, 0.0f };
-
-    // Limit gain to -100dB
-    assert(gain > 0.00001f);
-
-    w0 = F_TAU * freq_mult;
-    sin_w0 = sinf(w0);
-    cos_w0 = cosf(w0);
-    alpha = sin_w0/2.0f * rcpQ;
-
-    /* Calculate filter coefficients depending on filter type */
-    switch(type)
-    {
-        case ALfilterType_HighShelf:
-            sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
-            b[0] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
-            b[1] = -2.0f*gain*((gain-1.0f) + (gain+1.0f)*cos_w0                   );
-            b[2] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2);
-            a[0] =             (gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
-            a[1] =  2.0f*     ((gain-1.0f) - (gain+1.0f)*cos_w0                   );
-            a[2] =             (gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
-            break;
-        case ALfilterType_LowShelf:
-            sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
-            b[0] =       gain*((gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
-            b[1] =  2.0f*gain*((gain-1.0f) - (gain+1.0f)*cos_w0                   );
-            b[2] =       gain*((gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2);
-            a[0] =             (gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
-            a[1] = -2.0f*     ((gain-1.0f) + (gain+1.0f)*cos_w0                   );
-            a[2] =             (gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
-            break;
-        case ALfilterType_Peaking:
-            gain = sqrtf(gain);
-            b[0] =  1.0f + alpha * gain;
-            b[1] = -2.0f * cos_w0;
-            b[2] =  1.0f - alpha * gain;
-            a[0] =  1.0f + alpha / gain;
-            a[1] = -2.0f * cos_w0;
-            a[2] =  1.0f - alpha / gain;
-            break;
-
-        case ALfilterType_LowPass:
-            b[0] = (1.0f - cos_w0) / 2.0f;
-            b[1] =  1.0f - cos_w0;
-            b[2] = (1.0f - cos_w0) / 2.0f;
-            a[0] =  1.0f + alpha;
-            a[1] = -2.0f * cos_w0;
-            a[2] =  1.0f - alpha;
-            break;
-        case ALfilterType_HighPass:
-            b[0] =  (1.0f + cos_w0) / 2.0f;
-            b[1] = -(1.0f + cos_w0);
-            b[2] =  (1.0f + cos_w0) / 2.0f;
-            a[0] =   1.0f + alpha;
-            a[1] =  -2.0f * cos_w0;
-            a[2] =   1.0f - alpha;
-            break;
-        case ALfilterType_BandPass:
-            b[0] =  alpha;
-            b[1] =  0;
-            b[2] = -alpha;
-            a[0] =  1.0f + alpha;
-            a[1] = -2.0f * cos_w0;
-            a[2] =  1.0f - alpha;
-            break;
-    }
-
-    filter->a1 = a[1] / a[0];
-    filter->a2 = a[2] / a[0];
-    filter->b0 = b[0] / a[0];
-    filter->b1 = b[1] / a[0];
-    filter->b2 = b[2] / a[0];
-}
-
-
-static void lp_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void lp_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void lp_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
-{
-    switch(param)
-    {
-        case AL_LOWPASS_GAIN:
-            if(!(val >= AL_LOWPASS_MIN_GAIN && val <= AL_LOWPASS_MAX_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            filter->Gain = val;
-            break;
-
-        case AL_LOWPASS_GAINHF:
-            if(!(val >= AL_LOWPASS_MIN_GAINHF && val <= AL_LOWPASS_MAX_GAINHF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            filter->GainHF = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-static void lp_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    lp_SetParamf(filter, context, param, vals[0]);
-}
-
-static void lp_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void lp_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void lp_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    switch(param)
-    {
-        case AL_LOWPASS_GAIN:
-            *val = filter->Gain;
-            break;
-
-        case AL_LOWPASS_GAINHF:
-            *val = filter->GainHF;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-static void lp_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    lp_GetParamf(filter, context, param, vals);
-}
-
-
-static void hp_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void hp_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void hp_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
-{
-    switch(param)
-    {
-        case AL_HIGHPASS_GAIN:
-            if(!(val >= AL_HIGHPASS_MIN_GAIN && val <= AL_HIGHPASS_MAX_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            filter->Gain = val;
-            break;
-
-        case AL_HIGHPASS_GAINLF:
-            if(!(val >= AL_HIGHPASS_MIN_GAINLF && val <= AL_HIGHPASS_MAX_GAINLF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            filter->GainLF = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-static void hp_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    hp_SetParamf(filter, context, param, vals[0]);
-}
-
-static void hp_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void hp_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void hp_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    switch(param)
-    {
-        case AL_HIGHPASS_GAIN:
-            *val = filter->Gain;
-            break;
-
-        case AL_HIGHPASS_GAINLF:
-            *val = filter->GainLF;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-static void hp_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    hp_GetParamf(filter, context, param, vals);
-}
-
-
-static void bp_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void bp_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void bp_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
-{
-    switch(param)
-    {
-        case AL_BANDPASS_GAIN:
-            if(!(val >= AL_BANDPASS_MIN_GAIN && val <= AL_BANDPASS_MAX_GAIN))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            filter->Gain = val;
-            break;
-
-        case AL_BANDPASS_GAINHF:
-            if(!(val >= AL_BANDPASS_MIN_GAINHF && val <= AL_BANDPASS_MAX_GAINHF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            filter->GainHF = val;
-            break;
-
-        case AL_BANDPASS_GAINLF:
-            if(!(val >= AL_BANDPASS_MIN_GAINLF && val <= AL_BANDPASS_MAX_GAINLF))
-                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
-            filter->GainLF = val;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-static void bp_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
-    bp_SetParamf(filter, context, param, vals[0]);
-}
-
-static void bp_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void bp_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void bp_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
-{
-    switch(param)
-    {
-        case AL_BANDPASS_GAIN:
-            *val = filter->Gain;
-            break;
-
-        case AL_BANDPASS_GAINHF:
-            *val = filter->GainHF;
-            break;
-
-        case AL_BANDPASS_GAINLF:
-            *val = filter->GainLF;
-            break;
-
-        default:
-            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
-    }
-}
-static void bp_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
-{
-    bp_GetParamf(filter, context, param, vals);
-}
-
-
-static void null_SetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void null_SetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void null_SetParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALfloat UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void null_SetParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), const ALfloat *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-
-static void null_GetParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void null_GetParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void null_GetParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALfloat *UNUSED(val))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-static void null_GetParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum UNUSED(param), ALfloat *UNUSED(vals))
-{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
-
-
-ALvoid ReleaseALFilters(ALCdevice *device)
-{
-    ALsizei i;
-    for(i = 0;i < device->FilterMap.size;i++)
-    {
-        ALfilter *temp = device->FilterMap.values[i];
-        device->FilterMap.values[i] = NULL;
-
-        // Release filter structure
-        FreeThunkEntry(temp->id);
-        memset(temp, 0, sizeof(ALfilter));
-        al_free(temp);
-    }
-}
-
-
-static void InitFilterParams(ALfilter *filter, ALenum type)
-{
-    if(type == AL_FILTER_LOWPASS)
-    {
-        filter->Gain = AL_LOWPASS_DEFAULT_GAIN;
-        filter->GainHF = AL_LOWPASS_DEFAULT_GAINHF;
-        filter->HFReference = LOWPASSFREQREF;
-        filter->GainLF = 1.0f;
-        filter->LFReference = HIGHPASSFREQREF;
-
-        filter->SetParami  = lp_SetParami;
-        filter->SetParamiv = lp_SetParamiv;
-        filter->SetParamf  = lp_SetParamf;
-        filter->SetParamfv = lp_SetParamfv;
-        filter->GetParami  = lp_GetParami;
-        filter->GetParamiv = lp_GetParamiv;
-        filter->GetParamf  = lp_GetParamf;
-        filter->GetParamfv = lp_GetParamfv;
-    }
-    else if(type == AL_FILTER_HIGHPASS)
-    {
-        filter->Gain = AL_HIGHPASS_DEFAULT_GAIN;
-        filter->GainHF = 1.0f;
-        filter->HFReference = LOWPASSFREQREF;
-        filter->GainLF = AL_HIGHPASS_DEFAULT_GAINLF;
-        filter->LFReference = HIGHPASSFREQREF;
-
-        filter->SetParami  = hp_SetParami;
-        filter->SetParamiv = hp_SetParamiv;
-        filter->SetParamf  = hp_SetParamf;
-        filter->SetParamfv = hp_SetParamfv;
-        filter->GetParami  = hp_GetParami;
-        filter->GetParamiv = hp_GetParamiv;
-        filter->GetParamf  = hp_GetParamf;
-        filter->GetParamfv = hp_GetParamfv;
-    }
-    else if(type == AL_FILTER_BANDPASS)
-    {
-        filter->Gain = AL_BANDPASS_DEFAULT_GAIN;
-        filter->GainHF = AL_BANDPASS_DEFAULT_GAINHF;
-        filter->HFReference = LOWPASSFREQREF;
-        filter->GainLF = AL_BANDPASS_DEFAULT_GAINLF;
-        filter->LFReference = HIGHPASSFREQREF;
-
-        filter->SetParami  = bp_SetParami;
-        filter->SetParamiv = bp_SetParamiv;
-        filter->SetParamf  = bp_SetParamf;
-        filter->SetParamfv = bp_SetParamfv;
-        filter->GetParami  = bp_GetParami;
-        filter->GetParamiv = bp_GetParamiv;
-        filter->GetParamf  = bp_GetParamf;
-        filter->GetParamfv = bp_GetParamfv;
-    }
-    else
-    {
-        filter->Gain = 1.0f;
-        filter->GainHF = 1.0f;
-        filter->HFReference = LOWPASSFREQREF;
-        filter->GainLF = 1.0f;
-        filter->LFReference = HIGHPASSFREQREF;
-
-        filter->SetParami  = null_SetParami;
-        filter->SetParamiv = null_SetParamiv;
-        filter->SetParamf  = null_SetParamf;
-        filter->SetParamfv = null_SetParamfv;
-        filter->GetParami  = null_GetParami;
-        filter->GetParamiv = null_GetParamiv;
-        filter->GetParamf  = null_GetParamf;
-        filter->GetParamfv = null_GetParamfv;
-    }
-    filter->type = type;
-}

OpenAL32/alListener.c

@@ -1,510 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2000 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include "alMain.h"
-#include "AL/alc.h"
-#include "alError.h"
-#include "alListener.h"
-#include "alSource.h"
-
-AL_API ALvoid AL_APIENTRY alListenerf(ALenum param, ALfloat value)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    switch(param)
-    {
-    case AL_GAIN:
-        if(!(value >= 0.0f && isfinite(value)))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        context->Listener->Gain = value;
-        break;
-
-    case AL_METERS_PER_UNIT:
-        if(!(value >= 0.0f && isfinite(value)))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        context->Listener->MetersPerUnit = value;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alListener3f(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    switch(param)
-    {
-    case AL_POSITION:
-        if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        context->Listener->Position[0] = value1;
-        context->Listener->Position[1] = value2;
-        context->Listener->Position[2] = value3;
-        break;
-
-    case AL_VELOCITY:
-        if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        context->Listener->Velocity[0] = value1;
-        context->Listener->Velocity[1] = value2;
-        context->Listener->Velocity[2] = value3;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alListenerfv(ALenum param, const ALfloat *values)
-{
-    ALCcontext *context;
-
-    if(values)
-    {
-        switch(param)
-        {
-        case AL_GAIN:
-        case AL_METERS_PER_UNIT:
-            alListenerf(param, values[0]);
-            return;
-
-        case AL_POSITION:
-        case AL_VELOCITY:
-            alListener3f(param, values[0], values[1], values[2]);
-            return;
-        }
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    if(!(values))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(param)
-    {
-    case AL_ORIENTATION:
-        if(!(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2]) &&
-             isfinite(values[3]) && isfinite(values[4]) && isfinite(values[5])))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        /* AT then UP */
-        context->Listener->Forward[0] = values[0];
-        context->Listener->Forward[1] = values[1];
-        context->Listener->Forward[2] = values[2];
-        context->Listener->Up[0] = values[3];
-        context->Listener->Up[1] = values[4];
-        context->Listener->Up[2] = values[5];
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alListeneri(ALenum param, ALint UNUSED(value))
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API void AL_APIENTRY alListener3i(ALenum param, ALint value1, ALint value2, ALint value3)
-{
-    ALCcontext *context;
-
-    switch(param)
-    {
-    case AL_POSITION:
-    case AL_VELOCITY:
-        alListener3f(param, (ALfloat)value1, (ALfloat)value2, (ALfloat)value3);
-        return;
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API void AL_APIENTRY alListeneriv(ALenum param, const ALint *values)
-{
-    ALCcontext *context;
-
-    if(values)
-    {
-        ALfloat fvals[6];
-        switch(param)
-        {
-        case AL_POSITION:
-        case AL_VELOCITY:
-            alListener3f(param, (ALfloat)values[0], (ALfloat)values[1], (ALfloat)values[2]);
-            return;
-
-        case AL_ORIENTATION:
-            fvals[0] = (ALfloat)values[0];
-            fvals[1] = (ALfloat)values[1];
-            fvals[2] = (ALfloat)values[2];
-            fvals[3] = (ALfloat)values[3];
-            fvals[4] = (ALfloat)values[4];
-            fvals[5] = (ALfloat)values[5];
-            alListenerfv(param, fvals);
-            return;
-        }
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    if(!(values))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ReadLock(&context->PropLock);
-    if(!(value))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(param)
-    {
-    case AL_GAIN:
-        *value = context->Listener->Gain;
-        break;
-
-    case AL_METERS_PER_UNIT:
-        *value = context->Listener->MetersPerUnit;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ReadUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alGetListener3f(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ReadLock(&context->PropLock);
-    if(!(value1 && value2 && value3))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(param)
-    {
-    case AL_POSITION:
-        *value1 = context->Listener->Position[0];
-        *value2 = context->Listener->Position[1];
-        *value3 = context->Listener->Position[2];
-        break;
-
-    case AL_VELOCITY:
-        *value1 = context->Listener->Velocity[0];
-        *value2 = context->Listener->Velocity[1];
-        *value3 = context->Listener->Velocity[2];
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ReadUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alGetListenerfv(ALenum param, ALfloat *values)
-{
-    ALCcontext *context;
-
-    switch(param)
-    {
-    case AL_GAIN:
-    case AL_METERS_PER_UNIT:
-        alGetListenerf(param, values);
-        return;
-
-    case AL_POSITION:
-    case AL_VELOCITY:
-        alGetListener3f(param, values+0, values+1, values+2);
-        return;
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ReadLock(&context->PropLock);
-    if(!(values))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(param)
-    {
-    case AL_ORIENTATION:
-        // AT then UP
-        values[0] = context->Listener->Forward[0];
-        values[1] = context->Listener->Forward[1];
-        values[2] = context->Listener->Forward[2];
-        values[3] = context->Listener->Up[0];
-        values[4] = context->Listener->Up[1];
-        values[5] = context->Listener->Up[2];
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ReadUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alGetListeneri(ALenum param, ALint *value)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ReadLock(&context->PropLock);
-    if(!(value))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(param)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ReadUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API void AL_APIENTRY alGetListener3i(ALenum param, ALint *value1, ALint *value2, ALint *value3)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ReadLock(&context->PropLock);
-    if(!(value1 && value2 && value3))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch (param)
-    {
-    case AL_POSITION:
-        *value1 = (ALint)context->Listener->Position[0];
-        *value2 = (ALint)context->Listener->Position[1];
-        *value3 = (ALint)context->Listener->Position[2];
-        break;
-
-    case AL_VELOCITY:
-        *value1 = (ALint)context->Listener->Velocity[0];
-        *value2 = (ALint)context->Listener->Velocity[1];
-        *value3 = (ALint)context->Listener->Velocity[2];
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ReadUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API void AL_APIENTRY alGetListeneriv(ALenum param, ALint* values)
-{
-    ALCcontext *context;
-
-    switch(param)
-    {
-    case AL_POSITION:
-    case AL_VELOCITY:
-        alGetListener3i(param, values+0, values+1, values+2);
-        return;
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ReadLock(&context->PropLock);
-    if(!(values))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(param)
-    {
-    case AL_ORIENTATION:
-        // AT then UP
-        values[0] = (ALint)context->Listener->Forward[0];
-        values[1] = (ALint)context->Listener->Forward[1];
-        values[2] = (ALint)context->Listener->Forward[2];
-        values[3] = (ALint)context->Listener->Up[0];
-        values[4] = (ALint)context->Listener->Up[1];
-        values[5] = (ALint)context->Listener->Up[2];
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ReadUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-
-void UpdateListenerProps(ALCcontext *context)
-{
-    ALlistener *listener = context->Listener;
-    struct ALlistenerProps *props;
-
-    /* Get an unused proprty container, or allocate a new one as needed. */
-    props = ATOMIC_LOAD(&listener->FreeList, almemory_order_acquire);
-    if(!props)
-        props = al_calloc(16, sizeof(*props));
-    else
-    {
-        struct ALlistenerProps *next;
-        do {
-            next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
-        } while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&listener->FreeList, &props, next,
-                almemory_order_seq_cst, almemory_order_acquire) == 0);
-    }
-
-    /* Copy in current property values. */
-    props->Position[0] = listener->Position[0];
-    props->Position[1] = listener->Position[1];
-    props->Position[2] = listener->Position[2];
-
-    props->Velocity[0] = listener->Velocity[0];
-    props->Velocity[1] = listener->Velocity[1];
-    props->Velocity[2] = listener->Velocity[2];
-
-    props->Forward[0] = listener->Forward[0];
-    props->Forward[1] = listener->Forward[1];
-    props->Forward[2] = listener->Forward[2];
-    props->Up[0] = listener->Up[0];
-    props->Up[1] = listener->Up[1];
-    props->Up[2] = listener->Up[2];
-
-    props->Gain = listener->Gain;
-    props->MetersPerUnit = listener->MetersPerUnit;
-
-    props->DopplerFactor = context->DopplerFactor;
-    props->DopplerVelocity = context->DopplerVelocity;
-    props->SpeedOfSound = context->SpeedOfSound;
-
-    props->SourceDistanceModel = context->SourceDistanceModel;
-    props->DistanceModel = context->DistanceModel;;
-
-    /* Set the new container for updating internal parameters. */
-    props = ATOMIC_EXCHANGE_PTR(&listener->Update, props, almemory_order_acq_rel);
-    if(props)
-    {
-        /* If there was an unused update container, put it back in the
-         * freelist.
-         */
-        ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &listener->FreeList, props);
-    }
-}

OpenAL32/alState.c

@@ -1,779 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2000 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include "version.h"
-
-#include <stdlib.h>
-#include "alMain.h"
-#include "AL/alc.h"
-#include "AL/al.h"
-#include "AL/alext.h"
-#include "alError.h"
-#include "alListener.h"
-#include "alSource.h"
-#include "alAuxEffectSlot.h"
-
-#include "backends/base.h"
-
-
-static const ALchar alVendor[] = "OpenAL Community";
-static const ALchar alVersion[] = "1.1 ALSOFT "ALSOFT_VERSION;
-static const ALchar alRenderer[] = "OpenAL Soft";
-
-// Error Messages
-static const ALchar alNoError[] = "No Error";
-static const ALchar alErrInvalidName[] = "Invalid Name";
-static const ALchar alErrInvalidEnum[] = "Invalid Enum";
-static const ALchar alErrInvalidValue[] = "Invalid Value";
-static const ALchar alErrInvalidOp[] = "Invalid Operation";
-static const ALchar alErrOutOfMemory[] = "Out of Memory";
-
-/* Resampler strings */
-static const ALchar alPointResampler[] = "Nearest";
-static const ALchar alLinearResampler[] = "Linear";
-static const ALchar alSinc4Resampler[] = "4-Point Sinc";
-static const ALchar alBSincResampler[] = "Band-limited Sinc (12/24)";
-
-AL_API ALvoid AL_APIENTRY alEnable(ALenum capability)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    switch(capability)
-    {
-    case AL_SOURCE_DISTANCE_MODEL:
-        context->SourceDistanceModel = AL_TRUE;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alDisable(ALenum capability)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    WriteLock(&context->PropLock);
-    switch(capability)
-    {
-    case AL_SOURCE_DISTANCE_MODEL:
-        context->SourceDistanceModel = AL_FALSE;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-
-done:
-    WriteUnlock(&context->PropLock);
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALboolean AL_APIENTRY alIsEnabled(ALenum capability)
-{
-    ALCcontext *context;
-    ALboolean value=AL_FALSE;
-
-    context = GetContextRef();
-    if(!context) return AL_FALSE;
-
-    switch(capability)
-    {
-    case AL_SOURCE_DISTANCE_MODEL:
-        value = context->SourceDistanceModel;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}
-
-AL_API ALboolean AL_APIENTRY alGetBoolean(ALenum pname)
-{
-    ALCcontext *context;
-    ALboolean value=AL_FALSE;
-
-    context = GetContextRef();
-    if(!context) return AL_FALSE;
-
-    switch(pname)
-    {
-    case AL_DOPPLER_FACTOR:
-        if(context->DopplerFactor != 0.0f)
-            value = AL_TRUE;
-        break;
-
-    case AL_DOPPLER_VELOCITY:
-        if(context->DopplerVelocity != 0.0f)
-            value = AL_TRUE;
-        break;
-
-    case AL_DISTANCE_MODEL:
-        if(context->DistanceModel == AL_INVERSE_DISTANCE_CLAMPED)
-            value = AL_TRUE;
-        break;
-
-    case AL_SPEED_OF_SOUND:
-        if(context->SpeedOfSound != 0.0f)
-            value = AL_TRUE;
-        break;
-
-    case AL_DEFERRED_UPDATES_SOFT:
-        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-            value = AL_TRUE;
-        break;
-
-    case AL_GAIN_LIMIT_SOFT:
-        if(GAIN_MIX_MAX/context->GainBoost != 0.0f)
-            value = AL_TRUE;
-        break;
-
-    case AL_NUM_RESAMPLERS_SOFT:
-        /* Always non-0. */
-        value = AL_TRUE;
-        break;
-
-    case AL_DEFAULT_RESAMPLER_SOFT:
-        value = ResamplerDefault ? AL_TRUE : AL_FALSE;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}
-
-AL_API ALdouble AL_APIENTRY alGetDouble(ALenum pname)
-{
-    ALCcontext *context;
-    ALdouble value = 0.0;
-
-    context = GetContextRef();
-    if(!context) return 0.0;
-
-    switch(pname)
-    {
-    case AL_DOPPLER_FACTOR:
-        value = (ALdouble)context->DopplerFactor;
-        break;
-
-    case AL_DOPPLER_VELOCITY:
-        value = (ALdouble)context->DopplerVelocity;
-        break;
-
-    case AL_DISTANCE_MODEL:
-        value = (ALdouble)context->DistanceModel;
-        break;
-
-    case AL_SPEED_OF_SOUND:
-        value = (ALdouble)context->SpeedOfSound;
-        break;
-
-    case AL_DEFERRED_UPDATES_SOFT:
-        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-            value = (ALdouble)AL_TRUE;
-        break;
-
-    case AL_GAIN_LIMIT_SOFT:
-        value = (ALdouble)GAIN_MIX_MAX/context->GainBoost;
-        break;
-
-    case AL_NUM_RESAMPLERS_SOFT:
-        value = (ALdouble)(ResamplerMax + 1);
-        break;
-
-    case AL_DEFAULT_RESAMPLER_SOFT:
-        value = (ALdouble)ResamplerDefault;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}
-
-AL_API ALfloat AL_APIENTRY alGetFloat(ALenum pname)
-{
-    ALCcontext *context;
-    ALfloat value = 0.0f;
-
-    context = GetContextRef();
-    if(!context) return 0.0f;
-
-    switch(pname)
-    {
-    case AL_DOPPLER_FACTOR:
-        value = context->DopplerFactor;
-        break;
-
-    case AL_DOPPLER_VELOCITY:
-        value = context->DopplerVelocity;
-        break;
-
-    case AL_DISTANCE_MODEL:
-        value = (ALfloat)context->DistanceModel;
-        break;
-
-    case AL_SPEED_OF_SOUND:
-        value = context->SpeedOfSound;
-        break;
-
-    case AL_DEFERRED_UPDATES_SOFT:
-        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-            value = (ALfloat)AL_TRUE;
-        break;
-
-    case AL_GAIN_LIMIT_SOFT:
-        value = GAIN_MIX_MAX/context->GainBoost;
-        break;
-
-    case AL_NUM_RESAMPLERS_SOFT:
-        value = (ALfloat)(ResamplerMax + 1);
-        break;
-
-    case AL_DEFAULT_RESAMPLER_SOFT:
-        value = (ALfloat)ResamplerDefault;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}
-
-AL_API ALint AL_APIENTRY alGetInteger(ALenum pname)
-{
-    ALCcontext *context;
-    ALint value = 0;
-
-    context = GetContextRef();
-    if(!context) return 0;
-
-    switch(pname)
-    {
-    case AL_DOPPLER_FACTOR:
-        value = (ALint)context->DopplerFactor;
-        break;
-
-    case AL_DOPPLER_VELOCITY:
-        value = (ALint)context->DopplerVelocity;
-        break;
-
-    case AL_DISTANCE_MODEL:
-        value = (ALint)context->DistanceModel;
-        break;
-
-    case AL_SPEED_OF_SOUND:
-        value = (ALint)context->SpeedOfSound;
-        break;
-
-    case AL_DEFERRED_UPDATES_SOFT:
-        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-            value = (ALint)AL_TRUE;
-        break;
-
-    case AL_GAIN_LIMIT_SOFT:
-        value = (ALint)(GAIN_MIX_MAX/context->GainBoost);
-        break;
-
-    case AL_NUM_RESAMPLERS_SOFT:
-        value = ResamplerMax + 1;
-        break;
-
-    case AL_DEFAULT_RESAMPLER_SOFT:
-        value = ResamplerDefault;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}
-
-AL_API ALint64SOFT AL_APIENTRY alGetInteger64SOFT(ALenum pname)
-{
-    ALCcontext *context;
-    ALint64SOFT value = 0;
-
-    context = GetContextRef();
-    if(!context) return 0;
-
-    switch(pname)
-    {
-    case AL_DOPPLER_FACTOR:
-        value = (ALint64SOFT)context->DopplerFactor;
-        break;
-
-    case AL_DOPPLER_VELOCITY:
-        value = (ALint64SOFT)context->DopplerVelocity;
-        break;
-
-    case AL_DISTANCE_MODEL:
-        value = (ALint64SOFT)context->DistanceModel;
-        break;
-
-    case AL_SPEED_OF_SOUND:
-        value = (ALint64SOFT)context->SpeedOfSound;
-        break;
-
-    case AL_DEFERRED_UPDATES_SOFT:
-        if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-            value = (ALint64SOFT)AL_TRUE;
-        break;
-
-    case AL_GAIN_LIMIT_SOFT:
-        value = (ALint64SOFT)(GAIN_MIX_MAX/context->GainBoost);
-        break;
-
-    case AL_NUM_RESAMPLERS_SOFT:
-        value = (ALint64SOFT)(ResamplerMax + 1);
-        break;
-
-    case AL_DEFAULT_RESAMPLER_SOFT:
-        value = (ALint64SOFT)ResamplerDefault;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}
-
-AL_API ALvoid AL_APIENTRY alGetBooleanv(ALenum pname, ALboolean *values)
-{
-    ALCcontext *context;
-
-    if(values)
-    {
-        switch(pname)
-        {
-            case AL_DOPPLER_FACTOR:
-            case AL_DOPPLER_VELOCITY:
-            case AL_DISTANCE_MODEL:
-            case AL_SPEED_OF_SOUND:
-            case AL_DEFERRED_UPDATES_SOFT:
-            case AL_GAIN_LIMIT_SOFT:
-            case AL_NUM_RESAMPLERS_SOFT:
-            case AL_DEFAULT_RESAMPLER_SOFT:
-                values[0] = alGetBoolean(pname);
-                return;
-        }
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(values))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(pname)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetDoublev(ALenum pname, ALdouble *values)
-{
-    ALCcontext *context;
-
-    if(values)
-    {
-        switch(pname)
-        {
-            case AL_DOPPLER_FACTOR:
-            case AL_DOPPLER_VELOCITY:
-            case AL_DISTANCE_MODEL:
-            case AL_SPEED_OF_SOUND:
-            case AL_DEFERRED_UPDATES_SOFT:
-            case AL_GAIN_LIMIT_SOFT:
-            case AL_NUM_RESAMPLERS_SOFT:
-            case AL_DEFAULT_RESAMPLER_SOFT:
-                values[0] = alGetDouble(pname);
-                return;
-        }
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(values))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(pname)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetFloatv(ALenum pname, ALfloat *values)
-{
-    ALCcontext *context;
-
-    if(values)
-    {
-        switch(pname)
-        {
-            case AL_DOPPLER_FACTOR:
-            case AL_DOPPLER_VELOCITY:
-            case AL_DISTANCE_MODEL:
-            case AL_SPEED_OF_SOUND:
-            case AL_DEFERRED_UPDATES_SOFT:
-            case AL_GAIN_LIMIT_SOFT:
-            case AL_NUM_RESAMPLERS_SOFT:
-            case AL_DEFAULT_RESAMPLER_SOFT:
-                values[0] = alGetFloat(pname);
-                return;
-        }
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(values))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-    switch(pname)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alGetIntegerv(ALenum pname, ALint *values)
-{
-    ALCcontext *context;
-
-    if(values)
-    {
-        switch(pname)
-        {
-            case AL_DOPPLER_FACTOR:
-            case AL_DOPPLER_VELOCITY:
-            case AL_DISTANCE_MODEL:
-            case AL_SPEED_OF_SOUND:
-            case AL_DEFERRED_UPDATES_SOFT:
-            case AL_GAIN_LIMIT_SOFT:
-            case AL_NUM_RESAMPLERS_SOFT:
-            case AL_DEFAULT_RESAMPLER_SOFT:
-                values[0] = alGetInteger(pname);
-                return;
-        }
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    switch(pname)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API void AL_APIENTRY alGetInteger64vSOFT(ALenum pname, ALint64SOFT *values)
-{
-    ALCcontext *context;
-
-    if(values)
-    {
-        switch(pname)
-        {
-            case AL_DOPPLER_FACTOR:
-            case AL_DOPPLER_VELOCITY:
-            case AL_DISTANCE_MODEL:
-            case AL_SPEED_OF_SOUND:
-            case AL_DEFERRED_UPDATES_SOFT:
-            case AL_GAIN_LIMIT_SOFT:
-            case AL_NUM_RESAMPLERS_SOFT:
-            case AL_DEFAULT_RESAMPLER_SOFT:
-                values[0] = alGetInteger64SOFT(pname);
-                return;
-        }
-    }
-
-    context = GetContextRef();
-    if(!context) return;
-
-    switch(pname)
-    {
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API const ALchar* AL_APIENTRY alGetString(ALenum pname)
-{
-    const ALchar *value = NULL;
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return NULL;
-
-    switch(pname)
-    {
-    case AL_VENDOR:
-        value = alVendor;
-        break;
-
-    case AL_VERSION:
-        value = alVersion;
-        break;
-
-    case AL_RENDERER:
-        value = alRenderer;
-        break;
-
-    case AL_EXTENSIONS:
-        value = context->ExtensionList;
-        break;
-
-    case AL_NO_ERROR:
-        value = alNoError;
-        break;
-
-    case AL_INVALID_NAME:
-        value = alErrInvalidName;
-        break;
-
-    case AL_INVALID_ENUM:
-        value = alErrInvalidEnum;
-        break;
-
-    case AL_INVALID_VALUE:
-        value = alErrInvalidValue;
-        break;
-
-    case AL_INVALID_OPERATION:
-        value = alErrInvalidOp;
-        break;
-
-    case AL_OUT_OF_MEMORY:
-        value = alErrOutOfMemory;
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}
-
-AL_API ALvoid AL_APIENTRY alDopplerFactor(ALfloat value)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(value >= 0.0f && isfinite(value)))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-
-    WriteLock(&context->PropLock);
-    context->DopplerFactor = value;
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-    WriteUnlock(&context->PropLock);
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alDopplerVelocity(ALfloat value)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(value >= 0.0f && isfinite(value)))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-
-    WriteLock(&context->PropLock);
-    context->DopplerVelocity = value;
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-    WriteUnlock(&context->PropLock);
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alSpeedOfSound(ALfloat value)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(value > 0.0f && isfinite(value)))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-
-    WriteLock(&context->PropLock);
-    context->SpeedOfSound = value;
-    if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-        UpdateListenerProps(context);
-    WriteUnlock(&context->PropLock);
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alDistanceModel(ALenum value)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    if(!(value == AL_INVERSE_DISTANCE || value == AL_INVERSE_DISTANCE_CLAMPED ||
-         value == AL_LINEAR_DISTANCE || value == AL_LINEAR_DISTANCE_CLAMPED ||
-         value == AL_EXPONENT_DISTANCE || value == AL_EXPONENT_DISTANCE_CLAMPED ||
-         value == AL_NONE))
-        SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-
-    WriteLock(&context->PropLock);
-    context->DistanceModel = value;
-    if(!context->SourceDistanceModel)
-    {
-        if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
-            UpdateListenerProps(context);
-    }
-    WriteUnlock(&context->PropLock);
-
-done:
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API ALvoid AL_APIENTRY alDeferUpdatesSOFT(void)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ALCcontext_DeferUpdates(context);
-
-    ALCcontext_DecRef(context);
-}
-
-AL_API ALvoid AL_APIENTRY alProcessUpdatesSOFT(void)
-{
-    ALCcontext *context;
-
-    context = GetContextRef();
-    if(!context) return;
-
-    ALCcontext_ProcessUpdates(context);
-
-    ALCcontext_DecRef(context);
-}
-
-
-AL_API const ALchar* AL_APIENTRY alGetStringiSOFT(ALenum pname, ALsizei index)
-{
-    const char *ResamplerNames[] = {
-        alPointResampler, alLinearResampler,
-        alSinc4Resampler, alBSincResampler,
-    };
-    const ALchar *value = NULL;
-    ALCcontext *context;
-
-    static_assert(COUNTOF(ResamplerNames) == ResamplerMax+1, "Incorrect ResamplerNames list");
-
-    context = GetContextRef();
-    if(!context) return NULL;
-
-    switch(pname)
-    {
-    case AL_RESAMPLER_NAME_SOFT:
-        if(index < 0 || (size_t)index >= COUNTOF(ResamplerNames))
-            SET_ERROR_AND_GOTO(context, AL_INVALID_VALUE, done);
-        value = ResamplerNames[index];
-        break;
-
-    default:
-        SET_ERROR_AND_GOTO(context, AL_INVALID_ENUM, done);
-    }
-
-done:
-    ALCcontext_DecRef(context);
-
-    return value;
-}

OpenAL32/alThunk.c

@@ -1,108 +0,0 @@
-/**
- * OpenAL cross platform audio library
- * Copyright (C) 1999-2007 by authors.
- * This library is free software; you can redistribute it and/or
- *  modify it under the terms of the GNU Library General Public
- *  License as published by the Free Software Foundation; either
- *  version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- *  but WITHOUT ANY WARRANTY; without even the implied warranty of
- *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- *  Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- *  License along with this library; if not, write to the
- *  Free Software Foundation, Inc.,
- *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alThunk.h"
-
-#include "almalloc.h"
-
-
-static ATOMIC_FLAG *ThunkArray;
-static ALsizei      ThunkArraySize;
-static RWLock ThunkLock;
-
-void ThunkInit(void)
-{
-    RWLockInit(&ThunkLock);
-    ThunkArraySize = 1024;
-    ThunkArray = al_calloc(16, ThunkArraySize * sizeof(*ThunkArray));
-}
-
-void ThunkExit(void)
-{
-    al_free(ThunkArray);
-    ThunkArray = NULL;
-    ThunkArraySize = 0;
-}
-
-ALenum NewThunkEntry(ALuint *index)
-{
-    void *NewList;
-    ALsizei i;
-
-    ReadLock(&ThunkLock);
-    for(i = 0;i < ThunkArraySize;i++)
-    {
-        if(!ATOMIC_FLAG_TEST_AND_SET(&ThunkArray[i], almemory_order_acq_rel))
-        {
-            ReadUnlock(&ThunkLock);
-            *index = i+1;
-            return AL_NO_ERROR;
-        }
-    }
-    ReadUnlock(&ThunkLock);
-
-    WriteLock(&ThunkLock);
-    /* Double-check that there's still no free entries, in case another
-     * invocation just came through and increased the size of the array.
-     */
-    for(;i < ThunkArraySize;i++)
-    {
-        if(!ATOMIC_FLAG_TEST_AND_SET(&ThunkArray[i], almemory_order_acq_rel))
-        {
-            WriteUnlock(&ThunkLock);
-            *index = i+1;
-            return AL_NO_ERROR;
-        }
-    }
-
-    NewList = al_calloc(16, ThunkArraySize*2 * sizeof(*ThunkArray));
-    if(!NewList)
-    {
-        WriteUnlock(&ThunkLock);
-        ERR("Realloc failed to increase to %u entries!\n", ThunkArraySize*2);
-        return AL_OUT_OF_MEMORY;
-    }
-    memcpy(NewList, ThunkArray, ThunkArraySize*sizeof(*ThunkArray));
-    al_free(ThunkArray);
-    ThunkArray = NewList;
-    ThunkArraySize *= 2;
-
-    ATOMIC_FLAG_TEST_AND_SET(&ThunkArray[i], almemory_order_seq_cst);
-    *index = ++i;
-
-    for(;i < ThunkArraySize;i++)
-        ATOMIC_FLAG_CLEAR(&ThunkArray[i], almemory_order_relaxed);
-    WriteUnlock(&ThunkLock);
-
-    return AL_NO_ERROR;
-}
-
-void FreeThunkEntry(ALuint index)
-{
-    ReadLock(&ThunkLock);
-    if(index > 0 && (ALsizei)index <= ThunkArraySize)
-        ATOMIC_FLAG_CLEAR(&ThunkArray[index-1], almemory_order_release);
-    ReadUnlock(&ThunkLock);
-}

README

@@ -1,55 +0,0 @@
-Source Install
-==============
-
-To install OpenAL Soft, use your favorite shell to go into the build/
-directory, and run:
-
-cmake ..
-
-Assuming configuration went well, you can then build it, typically using GNU
-Make (KDevelop, MSVC, and others are possible depending on your system setup
-and CMake configuration).
-
-Please Note: Double check that the appropriate backends were detected. Often,
-complaints of no sound, crashing, and missing devices can be solved by making
-sure the correct backends are being used. CMake's output will identify which
-backends were enabled.
-
-For most systems, you will likely want to make sure ALSA, OSS, and PulseAudio
-were detected (if your target system uses them). For Windows, make sure
-DirectSound was detected.
-
-
-Utilities
-=========
-
-The source package comes with an informational utility, openal-info, and is
-built by default. It prints out information provided by the ALC and AL sub-
-systems, including discovered devices, version information, and extensions.
-
-
-Configuration
-=============
-
-OpenAL Soft can be configured on a per-user and per-system basis. This allows
-users and sysadmins to control information provided to applications, as well
-as application-agnostic behavior of the library. See alsoftrc.sample for
-available settings.
-
-
-Acknowledgements
-================
-
-Special thanks go to:
-
-Creative Labs for the original source code this is based off of.
-
-Christopher Fitzgerald for the current reverb effect implementation, and
-helping with the low-pass and HRTF filters.
-
-Christian Borss for the 3D panning code previous versions used as a base.
-
-Ben Davis for the idea behind a previous version of the click-removal code.
-
-Richard Furse for helping with my understanding of Ambisonics that is used by
-the various parts of the library.

README.md

@@ -0,0 +1,16 @@
+# openal-soft-vita
+
+This repo contains patchsets for vita [openal-soft](https://openal-soft.org/) port.
+
+Format is openal-soft-<openal_version>-vita-<patchset_version>.patch
+
+For full source see vita-* branches
+
+
+# building
+
+```
+mkdir build && cd build
+cmake -DCMAKE_TOOLCHAIN_FILE=${VITASDK}/share/vita.toolchain.cmake -DCMAKE_BUILD_TYPE=Release ..
+make install
+```

XCompile-Android.txt

@@ -1,39 +0,0 @@
-# Cross-compiling requires CMake 2.6 or newer. Example:
-# cmake .. -DCMAKE_TOOLCHAIN_FILE=../XCompile-Android.txt -DHOST=arm-linux-androideabi
-# Where 'arm-linux-androideabi' is the host prefix for the cross-compiler. If
-# you already have a toolchain file setup, you may use that instead of this
-# file. Make sure to set CMAKE_FIND_ROOT_PATH to where the NDK toolchain was
-# installed (e.g. "$ENV{HOME}/toolchains/arm-linux-androideabi-r10c-21").
-
-# the name of the target operating system
-SET(CMAKE_SYSTEM_NAME Linux)
-
-# which compilers to use for C and C++
-SET(CMAKE_C_COMPILER "${HOST}-gcc")
-SET(CMAKE_CXX_COMPILER "${HOST}-g++")
-SET(CMAKE_RC_COMPILER "${HOST}-windres")
-
-# here is the target environment located
-SET(CMAKE_FIND_ROOT_PATH "SET THIS TO THE NDK TOOLCHAIN'S INSTALL PATH")
-
-# here is where stuff gets installed to
-SET(CMAKE_INSTALL_PREFIX "${CMAKE_FIND_ROOT_PATH}" CACHE STRING "Install path prefix, prepended onto install directories." FORCE)
-
-# adjust the default behaviour of the FIND_XXX() commands:
-# search headers and libraries in the target environment, search 
-# programs in the host environment
-set(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER)
-set(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY)
-set(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY)
-
-# set env vars so that pkg-config will look in the appropriate directory for
-# .pc files (as there seems to be no way to force using ${HOST}-pkg-config)
-set(ENV{PKG_CONFIG_LIBDIR} "${CMAKE_INSTALL_PREFIX}/lib/pkgconfig")
-set(ENV{PKG_CONFIG_PATH} "")
-
-# Qt4 tools
-SET(QT_QMAKE_EXECUTABLE ${HOST}-qmake)
-SET(QT_MOC_EXECUTABLE ${HOST}-moc)
-SET(QT_RCC_EXECUTABLE ${HOST}-rcc)
-SET(QT_UIC_EXECUTABLE ${HOST}-uic)
-SET(QT_LRELEASE_EXECUTABLE ${HOST}-lrelease)

XCompile.txt

@@ -1,37 +0,0 @@
-# Cross-compiling requires CMake 2.6 or newer. Example:
-# cmake .. -DCMAKE_TOOLCHAIN_FILE=../XCompile.txt -DHOST=i686-w64-mingw32
-# Where 'i686-w64-mingw32' is the host prefix for your cross-compiler. If you
-# already have a toolchain file setup, you may use that instead of this file.
-
-# the name of the target operating system
-SET(CMAKE_SYSTEM_NAME Windows)
-
-# which compilers to use for C and C++
-SET(CMAKE_C_COMPILER "${HOST}-gcc")
-SET(CMAKE_CXX_COMPILER "${HOST}-g++")
-SET(CMAKE_RC_COMPILER "${HOST}-windres")
-
-# here is the target environment located
-SET(CMAKE_FIND_ROOT_PATH "/usr/${HOST}")
-
-# here is where stuff gets installed to
-SET(CMAKE_INSTALL_PREFIX "${CMAKE_FIND_ROOT_PATH}" CACHE STRING "Install path prefix, prepended onto install directories." FORCE)
-
-# adjust the default behaviour of the FIND_XXX() commands:
-# search headers and libraries in the target environment, search 
-# programs in the host environment
-set(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER)
-set(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY)
-set(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY)
-
-# set env vars so that pkg-config will look in the appropriate directory for
-# .pc files (as there seems to be no way to force using ${HOST}-pkg-config)
-set(ENV{PKG_CONFIG_LIBDIR} "${CMAKE_INSTALL_PREFIX}/lib/pkgconfig")
-set(ENV{PKG_CONFIG_PATH} "")
-
-# Qt4 tools
-SET(QT_QMAKE_EXECUTABLE ${HOST}-qmake)
-SET(QT_MOC_EXECUTABLE ${HOST}-moc)
-SET(QT_RCC_EXECUTABLE ${HOST}-rcc)
-SET(QT_UIC_EXECUTABLE ${HOST}-uic)
-SET(QT_LRELEASE_EXECUTABLE ${HOST}-lrelease)

alsoftrc.sample

@@ -1,487 +0,0 @@
-# OpenAL config file.
-#
-# Option blocks may appear multiple times, and duplicated options will take the
-# last value specified. Environment variables may be specified within option
-# values, and are automatically substituted when the config file is loaded.
-# Environment variable names may only contain alpha-numeric characters (a-z,
-# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,
-# specifying "$HOME/file.ext" would typically result in something like
-# "/home/user/file.ext". To specify an actual "$" character, use "$$".
-#
-# Device-specific values may be specified by including the device name in the
-# block name, with "general" replaced by the device name. That is, general
-# options for the device "Name of Device" would be in the [Name of Device]
-# block, while ALSA options would be in the [alsa/Name of Device] block.
-# Options marked as "(global)" are not influenced by the device.
-#
-# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
-# specific override settings in $HOME/.alsoftrc.
-# For Windows, these settings should go into $AppData\alsoft.ini
-#
-# Option and block names are case-senstive. The supplied values are only hints
-# and may not be honored (though generally it'll try to get as close as
-# possible). Note: options that are left unset may default to app- or system-
-# specified values. These are the current available settings:
-
-##
-## General stuff
-##
-[general]
-
-## disable-cpu-exts: (global)
-#  Disables use of specialized methods that use specific CPU intrinsics.
-#  Certain methods may utilize CPU extensions for improved performance, and
-#  this option is useful for preventing some or all of those methods from being
-#  used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
-#  Specifying 'all' disables use of all such specialized methods.
-#disable-cpu-exts =
-
-## drivers: (global)
-#  Sets the backend driver list order, comma-seperated. Unknown backends and
-#  duplicated names are ignored. Unlisted backends won't be considered for use
-#  unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
-#  other backends, while 'oss' will try OSS only). Backends prepended with -
-#  won't be considered for use (e.g. '-oss,' will try all available backends
-#  except OSS). An empty list means to try all backends.
-#drivers =
-
-## channels:
-#  Sets the output channel configuration. If left unspecified, one will try to
-#  be detected from the system, and defaulting to stereo. The available values
-#  are: mono, stereo, quad, surround51, surround51rear, surround61, surround71,
-#  ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic
-#  channels of the given order (using ACN ordering and SN3D normalization by
-#  default), which need to be decoded to play correctly on speakers.
-#channels =
-
-## sample-type:
-#  Sets the output sample type. Currently, all mixing is done with 32-bit float
-#  and converted to the output sample type as needed. Available values are:
-#  int8    - signed 8-bit int
-#  uint8   - unsigned 8-bit int
-#  int16   - signed 16-bit int
-#  uint16  - unsigned 16-bit int
-#  int32   - signed 32-bit int
-#  uint32  - unsigned 32-bit int
-#  float32 - 32-bit float
-#sample-type = float32
-
-## frequency:
-#  Sets the output frequency. If left unspecified it will try to detect a
-#  default from the system, otherwise it will default to 44100.
-#frequency =
-
-## period_size:
-#  Sets the update period size, in frames. This is the number of frames needed
-#  for each mixing update. Acceptable values range between 64 and 8192.
-#period_size = 1024
-
-## periods:
-#  Sets the number of update periods. Higher values create a larger mix ahead,
-#  which helps protect against skips when the CPU is under load, but increases
-#  the delay between a sound getting mixed and being heard. Acceptable values
-#  range between 2 and 16.
-#periods = 3
-
-## stereo-mode:
-#  Specifies if stereo output is treated as being headphones or speakers. With
-#  headphones, HRTF or crossfeed filters may be used for better audio quality.
-#  Valid settings are auto, speakers, and headphones.
-#stereo-mode = auto
-
-## stereo-encoding:
-#  Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)
-#  uses standard amplitude panning (aka pair-wise, stereo pair, etc) between
-#  -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ
-#  output, which encodes some surround sound information into stereo output
-#  that can be decoded with a surround sound receiver. If crossfeed filters are
-#  used, UHJ is disabled.
-#stereo-encoding = panpot
-
-## ambi-format:
-#  Specifies the channel order and normalization for the "ambi*" set of channel
-#  configurations. Valid settings are: fuma, acn+sn3d, acn+n3d
-#ambi-format = acn+sn3d
-
-## hrtf:
-#  Controls HRTF processing. These filters provide better spatialization of
-#  sounds while using headphones, but do require a bit more CPU power. The
-#  default filters will only work with 44100hz or 48000hz stereo output. While
-#  HRTF is used, the cf_level option is ignored. Setting this to auto (default)
-#  will allow HRTF to be used when headphones are detected or the app requests
-#  it, while setting true or false will forcefully enable or disable HRTF
-#  respectively.
-#hrtf = auto
-
-## default-hrtf:
-#  Specifies the default HRTF to use. When multiple HRTFs are available, this
-#  determines the preferred one to use if none are specifically requested. Note
-#  that this is the enumerated HRTF name, not necessarily the filename.
-#default-hrtf =
-
-## hrtf-paths:
-#  Specifies a comma-separated list of paths containing HRTF data sets. The
-#  format of the files are described in docs/hrtf.txt. The files within the
-#  directories must have the .mhr file extension to be recognized. By default,
-#  OS-dependent data paths will be used. They will also be used if the list
-#  ends with a comma. On Windows this is:
-#  $AppData\openal\hrtf
-#  And on other systems, it's (in order):
-#  $XDG_DATA_HOME/openal/hrtf  (defaults to $HOME/.local/share/openal/hrtf)
-#  $XDG_DATA_DIRS/openal/hrtf  (defaults to /usr/local/share/openal/hrtf and
-#                               /usr/share/openal/hrtf)
-#hrtf-paths =
-
-## cf_level:
-#  Sets the crossfeed level for stereo output. Valid values are:
-#  0 - No crossfeed
-#  1 - Low crossfeed
-#  2 - Middle crossfeed
-#  3 - High crossfeed (virtual speakers are closer to itself)
-#  4 - Low easy crossfeed
-#  5 - Middle easy crossfeed
-#  6 - High easy crossfeed
-#  Users of headphones may want to try various settings. Has no effect on non-
-#  stereo modes.
-#cf_level = 0
-
-## resampler: (global)
-#  Selects the resampler used when mixing sources. Valid values are:
-#  point - nearest sample, no interpolation
-#  linear - extrapolates samples using a linear slope between samples
-#  sinc4 - extrapolates samples using a 4-point Sinc filter
-#  bsinc - extrapolates samples using a band-limited Sinc filter (varying
-#          between 12 and 24 points, with anti-aliasing)
-#  Specifying other values will result in using the default (linear).
-#resampler = linear
-
-## rt-prio: (global)
-#  Sets real-time priority for the mixing thread. Not all drivers may use this
-#  (eg. PortAudio) as they already control the priority of the mixing thread.
-#  0 and negative values will disable it. Note that this may constitute a
-#  security risk since a real-time priority thread can indefinitely block
-#  normal-priority threads if it fails to wait. As such, the default is
-#  disabled.
-#rt-prio = 0
-
-## sources:
-#  Sets the maximum number of allocatable sources. Lower values may help for
-#  systems with apps that try to play more sounds than the CPU can handle.
-#sources = 256
-
-## slots:
-#  Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
-#  can use a non-negligible amount of CPU time if an effect is set on it even
-#  if no sources are feeding it, so this may help when apps use more than the
-#  system can handle.
-#slots = 64
-
-## sends:
-#  Limits the number of auxiliary sends allowed per source. Setting this higher
-#  than the default has no effect.
-#sends = 16
-
-## output-limiter:
-#  Applies a gain limiter on the final mixed output. This reduces the volume
-#  when the output samples would otherwise clamp, avoiding excessive clipping
-#  noise.
-#output-limiter = true
-
-## dither:
-#  Applies dithering on the final mix, for 8- and 16-bit output by default.
-#  This replaces the distortion created by nearest-value quantization with low-
-#  level whitenoise.
-#dither = true
-
-## dither-depth:
-#  Quantization bit-depth for dithered output. A value of 0 (or less) will
-#  match the output sample depth. For int32, uint32, and float32 output, 0 will
-#  disable dithering because they're at or beyond the rendered precision. The
-#  maximum dither depth is 24.
-#dither-depth = 0
-
-## volume-adjust:
-#  A global volume adjustment for source output, expressed in decibels. The
-#  value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
-#  be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
-#  value of 0 means no change.
-#volume-adjust = 0
-
-## excludefx: (global)
-#  Sets which effects to exclude, preventing apps from using them. This can
-#  help for apps that try to use effects which are too CPU intensive for the
-#  system to handle. Available effects are: eaxreverb,reverb,chorus,compressor,
-#  distortion,echo,equalizer,flanger,modulator,dedicated
-#excludefx =
-
-## default-reverb: (global)
-#  A reverb preset that applies by default to all sources on send 0
-#  (applications that set their own slots on send 0 will override this).
-#  Available presets are: None, Generic, PaddedCell, Room, Bathroom,
-#  Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
-#  CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
-#  Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
-#default-reverb =
-
-## trap-alc-error: (global)
-#  Generates a SIGTRAP signal when an ALC device error is generated, on systems
-#  that support it. This helps when debugging, while trying to find the cause
-#  of a device error. On Windows, a breakpoint exception is generated.
-#trap-alc-error = false
-
-## trap-al-error: (global)
-#  Generates a SIGTRAP signal when an AL context error is generated, on systems
-#  that support it. This helps when debugging, while trying to find the cause
-#  of a context error. On Windows, a breakpoint exception is generated.
-#trap-al-error = false
-
-##
-## Ambisonic decoder stuff
-##
-[decoder]
-
-## hq-mode:
-#  Enables a high-quality ambisonic decoder. This mode is capable of frequency-
-#  dependent processing, creating a better reproduction of 3D sound rendering
-#  over surround sound speakers. Enabling this also requires specifying decoder
-#  configuration files for the appropriate speaker configuration you intend to
-#  use (see the quad, surround51, etc options below). Currently, up to third-
-#  order decoding is supported.
-hq-mode = false
-
-## distance-comp:
-#  Enables compensation for the speakers' relative distances to the listener.
-#  This applies the necessary delays and attenuation to make the speakers
-#  behave as though they are all equidistant, which is important for proper
-#  playback of 3D sound rendering. Requires the proper distances to be
-#  specified in the decoder configuration file.
-distance-comp = true
-
-## nfc:
-#  Enables near-field control filters. This simulates and compensates for low-
-#  frequency effects caused by the curvature of nearby sound-waves, which
-#  creates a more realistic perception of sound distance. Note that the effect
-#  may be stronger or weaker than intended if the application doesn't use or
-#  specify an appropriate unit scale, or if incorrect speaker distances are set
-#  in the decoder configuration file. Requires hq-mode to be enabled.
-nfc = true
-
-## nfc-ref-delay
-#  Specifies the reference delay value for ambisonic output. When channels is
-#  set to one of the ambi* formats, this option enables NFC-HOA output with the
-#  specified Reference Delay parameter. The specified value can then be shared
-#  with an appropriate NFC-HOA decoder to reproduce correct near-field effects.
-#  Keep in mind that despite being designed for higher-order ambisonics, this
-#  applies to first-order output all the same. When left unset, normal output
-#  is created with no near-field simulation.
-nfc-ref-delay =
-
-## quad:
-#  Decoder configuration file for Quadrophonic channel output. See
-#  docs/ambdec.txt for a description of the file format.
-quad =
-
-## surround51:
-#  Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
-#  See docs/ambdec.txt for a description of the file format.
-surround51 =
-
-## surround61:
-#  Decoder configuration file for 6.1 Surround channel output. See
-#  docs/ambdec.txt for a description of the file format.
-surround61 =
-
-## surround71:
-#  Decoder configuration file for 7.1 Surround channel output. See
-#  docs/ambdec.txt for a description of the file format. Note: This can be used
-#  to enable 3D7.1 with the appropriate configuration and speaker placement,
-#  see docs/3D7.1.txt.
-surround71 =
-
-##
-## Reverb effect stuff (includes EAX reverb)
-##
-[reverb]
-
-## boost: (global)
-#  A global amplification for reverb output, expressed in decibels. The value
-#  is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
-#  scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
-#  value of 0 means no change.
-#boost = 0
-
-## emulate-eax: (global)
-#  Allows the standard reverb effect to be used in place of EAX reverb. EAX
-#  reverb processing is a bit more CPU intensive than standard, so this option
-#  allows a simpler effect to be used at the loss of some quality.
-#emulate-eax = false
-
-##
-## PulseAudio backend stuff
-##
-[pulse]
-
-## spawn-server: (global)
-#  Attempts to autospawn a PulseAudio server whenever needed (initializing the
-#  backend, enumerating devices, etc). Setting autospawn to false in Pulse's
-#  client.conf will still prevent autospawning even if this is set to true.
-#spawn-server = true
-
-## allow-moves: (global)
-#  Allows PulseAudio to move active streams to different devices. Note that the
-#  device specifier (seen by applications) will not be updated when this
-#  occurs, and neither will the AL device configuration (sample rate, format,
-#  etc).
-#allow-moves = false
-
-## fix-rate:
-#  Specifies whether to match the playback stream's sample rate to the device's
-#  sample rate. Enabling this forces OpenAL Soft to mix sources and effects
-#  directly to the actual output rate, avoiding a second resample pass by the
-#  PulseAudio server.
-#fix-rate = false
-
-##
-## ALSA backend stuff
-##
-[alsa]
-
-## device: (global)
-#  Sets the device name for the default playback device.
-#device = default
-
-## device-prefix: (global)
-#  Sets the prefix used by the discovered (non-default) playback devices. This
-#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the
-#  device index for the requested device name.
-#device-prefix = plughw:
-
-## device-prefix-*: (global)
-#  Card- and device-specific prefixes may be used to override the device-prefix
-#  option. The option may specify the card id (eg, device-prefix-NVidia), or
-#  the card id and device index (eg, device-prefix-NVidia-0). The card id is
-#  case-sensitive.
-#device-prefix- =
-
-## capture: (global)
-#  Sets the device name for the default capture device.
-#capture = default
-
-## capture-prefix: (global)
-#  Sets the prefix used by the discovered (non-default) capture devices. This
-#  will be appended with "CARD=c,DEV=d", where c is the card id and d is the
-#  device number for the requested device name.
-#capture-prefix = plughw:
-
-## capture-prefix-*: (global)
-#  Card- and device-specific prefixes may be used to override the
-#  capture-prefix option. The option may specify the card id (eg,
-#  capture-prefix-NVidia), or the card id and device index (eg,
-#  capture-prefix-NVidia-0). The card id is case-sensitive.
-#capture-prefix- =
-
-## mmap:
-#  Sets whether to try using mmap mode (helps reduce latencies and CPU
-#  consumption). If mmap isn't available, it will automatically fall back to
-#  non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
-#  and anything else will force mmap off.
-#mmap = true
-
-## allow-resampler:
-#  Specifies whether to allow ALSA's built-in resampler. Enabling this will
-#  allow the playback device to be set to a different sample rate than the
-#  actual output, causing ALSA to apply its own resampling pass after OpenAL
-#  Soft resamples and mixes the sources and effects for output.
-#allow-resampler = false
-
-##
-## OSS backend stuff
-##
-[oss]
-
-## device: (global)
-#  Sets the device name for OSS output.
-#device = /dev/dsp
-
-## capture: (global)
-#  Sets the device name for OSS capture.
-#capture = /dev/dsp
-
-##
-## Solaris backend stuff
-##
-[solaris]
-
-## device: (global)
-#  Sets the device name for Solaris output.
-#device = /dev/audio
-
-##
-## QSA backend stuff
-##
-[qsa]
-
-##
-## JACK backend stuff
-##
-[jack]
-
-## spawn-server: (global)
-#  Attempts to autospawn a JACK server whenever needed (initializing the
-#  backend, opening devices, etc).
-#spawn-server = false
-
-## buffer-size:
-#  Sets the update buffer size, in samples, that the backend will keep buffered
-#  to handle the server's real-time processing requests. This value must be a
-#  power of 2, or else it will be rounded up to the next power of 2. If it is
-#  less than JACK's buffer update size, it will be clamped. This option may
-#  be useful in case the server's update size is too small and doesn't give the
-#  mixer time to keep enough audio available for the processing requests.
-#buffer-size = 0
-
-##
-## MMDevApi backend stuff
-##
-[mmdevapi]
-
-##
-## DirectSound backend stuff
-##
-[dsound]
-
-##
-## Windows Multimedia backend stuff
-##
-[winmm]
-
-##
-## PortAudio backend stuff
-##
-[port]
-
-## device: (global)
-#  Sets the device index for output. Negative values will use the default as
-#  given by PortAudio itself.
-#device = -1
-
-## capture: (global)
-#  Sets the device index for capture. Negative values will use the default as
-#  given by PortAudio itself.
-#capture = -1
-
-##
-## Wave File Writer stuff
-##
-[wave]
-
-## file: (global)
-#  Sets the filename of the wave file to write to. An empty name prevents the
-#  backend from opening, even when explicitly requested.
-#  THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
-#file =
-
-## bformat: (global)
-#  Creates AMB format files using first-order ambisonics instead of a standard
-#  single- or multi-channel .wav file.
-#bformat = false

appveyor.yml

@@ -1,19 +0,0 @@
-version: 1.18.1.{build}
-
-environment:
-    matrix:
-      - GEN: "Visual Studio 14 2015"
-        CFG: Release
-      - GEN: "Visual Studio 14 2015 Win64"
-        CFG: Release
-
-install:
-    # Remove the VS Xamarin targets to reduce AppVeyor specific noise in build
-    # logs.  See also http://help.appveyor.com/discussions/problems/4569
-    - del "C:\Program Files (x86)\MSBuild\14.0\Microsoft.Common.targets\ImportAfter\Xamarin.Common.targets"
-
-build_script:
-    - cd build
-    - cmake -G"%GEN%" -DALSOFT_REQUIRE_WINMM=ON -DALSOFT_REQUIRE_DSOUND=ON -DALSOFT_REQUIRE_MMDEVAPI=ON -DALSOFT_EMBED_HRTF_DATA=YES ..
-    - cmake --build . --config %CFG% --clean-first
-

cmake/CheckFileOffsetBits.c

@@ -1,9 +0,0 @@
-#include <sys/types.h>
-
-#define KB ((off_t)(1024))
-#define MB ((off_t)(KB*1024))
-#define GB ((off_t)(MB*1024))
-int tb[((GB+GB+GB) > GB) ? 1 : -1];
-
-int main()
-{ return 0; }

cmake/CheckFileOffsetBits.cmake

@@ -1,39 +0,0 @@
-# - Check if the _FILE_OFFSET_BITS macro is needed for large files
-# CHECK_FILE_OFFSET_BITS()
-#
-# The following variables may be set before calling this macro to
-# modify the way the check is run:
-#
-#  CMAKE_REQUIRED_FLAGS = string of compile command line flags
-#  CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar)
-#  CMAKE_REQUIRED_INCLUDES = list of include directories
-# Copyright (c) 2009, Chris Robinson
-#
-# Redistribution and use is allowed according to the terms of the LGPL license.
-
-
-MACRO(CHECK_FILE_OFFSET_BITS)
-
-  IF(NOT DEFINED _FILE_OFFSET_BITS)
-    MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files")
-    TRY_COMPILE(__WITHOUT_FILE_OFFSET_BITS_64
-      ${CMAKE_CURRENT_BINARY_DIR}
-      ${CMAKE_CURRENT_SOURCE_DIR}/cmake/CheckFileOffsetBits.c
-      COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS})
-    IF(NOT __WITHOUT_FILE_OFFSET_BITS_64)
-      TRY_COMPILE(__WITH_FILE_OFFSET_BITS_64
-        ${CMAKE_CURRENT_BINARY_DIR}
-        ${CMAKE_CURRENT_SOURCE_DIR}/cmake/CheckFileOffsetBits.c
-        COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS} -D_FILE_OFFSET_BITS=64)
-    ENDIF(NOT __WITHOUT_FILE_OFFSET_BITS_64)
-
-    IF(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64)
-      SET(_FILE_OFFSET_BITS 64 CACHE INTERNAL "_FILE_OFFSET_BITS macro needed for large files")
-      MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files - 64")
-    ELSE(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64)
-      SET(_FILE_OFFSET_BITS "" CACHE INTERNAL "_FILE_OFFSET_BITS macro needed for large files")
-      MESSAGE(STATUS "Checking _FILE_OFFSET_BITS for large files - not needed")
-    ENDIF(NOT __WITHOUT_FILE_OFFSET_BITS_64 AND __WITH_FILE_OFFSET_BITS_64)
-  ENDIF(NOT DEFINED _FILE_OFFSET_BITS)
-
-ENDMACRO(CHECK_FILE_OFFSET_BITS)

cmake/CheckSharedFunctionExists.cmake

@@ -1,92 +0,0 @@
-# - Check if a symbol exists as a function, variable, or macro
-# CHECK_SYMBOL_EXISTS(<symbol> <files> <variable>)
-#
-# Check that the <symbol> is available after including given header
-# <files> and store the result in a <variable>.  Specify the list
-# of files in one argument as a semicolon-separated list.
-#
-# If the header files define the symbol as a macro it is considered
-# available and assumed to work.  If the header files declare the
-# symbol as a function or variable then the symbol must also be
-# available for linking.  If the symbol is a type or enum value
-# it will not be recognized (consider using CheckTypeSize or
-# CheckCSourceCompiles).
-#
-# The following variables may be set before calling this macro to
-# modify the way the check is run:
-#
-#  CMAKE_REQUIRED_FLAGS = string of compile command line flags
-#  CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar)
-#  CMAKE_REQUIRED_INCLUDES = list of include directories
-#  CMAKE_REQUIRED_LIBRARIES = list of libraries to link
-
-#=============================================================================
-# Copyright 2003-2011 Kitware, Inc.
-#
-# Distributed under the OSI-approved BSD License (the "License");
-# see accompanying file Copyright.txt for details.
-#
-# This software is distributed WITHOUT ANY WARRANTY; without even the
-# implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
-# See the License for more information.
-#=============================================================================
-# (To distribute this file outside of CMake, substitute the full
-#  License text for the above reference.)
-
-MACRO(CHECK_SHARED_FUNCTION_EXISTS SYMBOL FILES LIBRARY LOCATION VARIABLE)
-  IF(NOT DEFINED "${VARIABLE}" OR "x${${VARIABLE}}" STREQUAL "x${VARIABLE}")
-    SET(CMAKE_CONFIGURABLE_FILE_CONTENT "/* */\n")
-    SET(MACRO_CHECK_SYMBOL_EXISTS_FLAGS ${CMAKE_REQUIRED_FLAGS})
-    IF(CMAKE_REQUIRED_LIBRARIES)
-      SET(CHECK_SYMBOL_EXISTS_LIBS 
-        "-DLINK_LIBRARIES:STRING=${CMAKE_REQUIRED_LIBRARIES};${LIBRARY}")
-    ELSE(CMAKE_REQUIRED_LIBRARIES)
-      SET(CHECK_SYMBOL_EXISTS_LIBS
-        "-DLINK_LIBRARIES:STRING=${LIBRARY}")
-    ENDIF(CMAKE_REQUIRED_LIBRARIES)
-    IF(CMAKE_REQUIRED_INCLUDES)
-      SET(CMAKE_SYMBOL_EXISTS_INCLUDES
-        "-DINCLUDE_DIRECTORIES:STRING=${CMAKE_REQUIRED_INCLUDES}")
-    ELSE(CMAKE_REQUIRED_INCLUDES)
-      SET(CMAKE_SYMBOL_EXISTS_INCLUDES)
-    ENDIF(CMAKE_REQUIRED_INCLUDES)
-    FOREACH(FILE ${FILES})
-      SET(CMAKE_CONFIGURABLE_FILE_CONTENT
-        "${CMAKE_CONFIGURABLE_FILE_CONTENT}#include <${FILE}>\n")
-    ENDFOREACH(FILE)
-    SET(CMAKE_CONFIGURABLE_FILE_CONTENT
-      "${CMAKE_CONFIGURABLE_FILE_CONTENT}\nvoid cmakeRequireSymbol(int dummy,...){(void)dummy;}\nint main()\n{\n  cmakeRequireSymbol(0,&${SYMBOL});\n  return 0;\n}\n")
-
-    CONFIGURE_FILE("${CMAKE_ROOT}/Modules/CMakeConfigurableFile.in"
-      "${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c" @ONLY)
-
-    MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY}")
-    TRY_COMPILE(${VARIABLE}
-      ${CMAKE_CURRENT_BINARY_DIR}
-      ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c
-      COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS}
-      CMAKE_FLAGS 
-      -DCOMPILE_DEFINITIONS:STRING=${MACRO_CHECK_SYMBOL_EXISTS_FLAGS}
-      -DLINK_DIRECTORIES:STRING=${LOCATION}
-      "${CHECK_SYMBOL_EXISTS_LIBS}"
-      "${CMAKE_SYMBOL_EXISTS_INCLUDES}"
-      OUTPUT_VARIABLE OUTPUT)
-    IF(${VARIABLE})
-      MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - found")
-      SET(${VARIABLE} 1 CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}")
-      FILE(APPEND ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeOutput.log
-        "Determining if the ${SYMBOL} "
-        "exist in ${LIBRARY} passed with the following output:\n"
-        "${OUTPUT}\nFile ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n"
-        "${CMAKE_CONFIGURABLE_FILE_CONTENT}\n")
-    ELSE(${VARIABLE})
-      MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - not found.")
-      SET(${VARIABLE} "" CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}")
-      FILE(APPEND ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeError.log
-        "Determining if the ${SYMBOL} "
-        "exist in ${LIBRARY} failed with the following output:\n"
-        "${OUTPUT}\nFile ${CMAKE_CURRENT_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n"
-        "${CMAKE_CONFIGURABLE_FILE_CONTENT}\n")
-    ENDIF(${VARIABLE})
-  ENDIF(NOT DEFINED "${VARIABLE}" OR "x${${VARIABLE}}" STREQUAL "x${VARIABLE}")
-ENDMACRO(CHECK_SHARED_FUNCTION_EXISTS)

cmake/FindALSA.cmake

@@ -1,73 +0,0 @@
-# - Find alsa
-# Find the alsa libraries (asound)
-#
-#  This module defines the following variables:
-#     ALSA_FOUND       - True if ALSA_INCLUDE_DIR & ALSA_LIBRARY are found
-#     ALSA_LIBRARIES   - Set when ALSA_LIBRARY is found
-#     ALSA_INCLUDE_DIRS - Set when ALSA_INCLUDE_DIR is found
-#
-#     ALSA_INCLUDE_DIR - where to find asoundlib.h, etc.
-#     ALSA_LIBRARY     - the asound library
-#     ALSA_VERSION_STRING - the version of alsa found (since CMake 2.8.8)
-#
-
-#=============================================================================
-# Copyright 2009-2011 Kitware, Inc.
-# Copyright 2009-2011 Philip Lowman <philip@yhbt.com>
-#
-# Redistribution and use in source and binary forms, with or without
-# modification, are permitted provided that the following conditions are
-# met:
-#
-#  * Redistributions of source code must retain the above copyright notice,
-#    this list of conditions and the following disclaimer.
-#
-#  * Redistributions in binary form must reproduce the above copyright notice,
-#    this list of conditions and the following disclaimer in the documentation
-#    and/or other materials provided with the distribution.
-#
-#  * The names of Kitware, Inc., the Insight Consortium, or the names of
-#    any consortium members, or of any contributors, may not be used to
-#    endorse or promote products derived from this software without
-#    specific prior written permission.
-#
-# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDER AND CONTRIBUTORS ``AS IS''
-# AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
-# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
-# ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHORS OR CONTRIBUTORS BE LIABLE FOR
-# ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
-# DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
-# SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
-# CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
-# OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-# OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-#=============================================================================
-
-find_path(ALSA_INCLUDE_DIR NAMES alsa/asoundlib.h
-          DOC "The ALSA (asound) include directory"
-)
-
-find_library(ALSA_LIBRARY NAMES asound
-             DOC "The ALSA (asound) library"
-)
-
-if(ALSA_INCLUDE_DIR AND EXISTS "${ALSA_INCLUDE_DIR}/alsa/version.h")
-  file(STRINGS "${ALSA_INCLUDE_DIR}/alsa/version.h" alsa_version_str REGEX "^#define[\t ]+SND_LIB_VERSION_STR[\t ]+\".*\"")
-
-  string(REGEX REPLACE "^.*SND_LIB_VERSION_STR[\t ]+\"([^\"]*)\".*$" "\\1" ALSA_VERSION_STRING "${alsa_version_str}")
-  unset(alsa_version_str)
-endif()
-
-# handle the QUIETLY and REQUIRED arguments and set ALSA_FOUND to TRUE if
-# all listed variables are TRUE
-include(FindPackageHandleStandardArgs)
-find_package_handle_standard_args(ALSA
-                                  REQUIRED_VARS ALSA_LIBRARY ALSA_INCLUDE_DIR
-                                  VERSION_VAR ALSA_VERSION_STRING)
-
-if(ALSA_FOUND)
-  set( ALSA_LIBRARIES ${ALSA_LIBRARY} )
-  set( ALSA_INCLUDE_DIRS ${ALSA_INCLUDE_DIR} )
-endif()
-
-mark_as_advanced(ALSA_INCLUDE_DIR ALSA_LIBRARY)