Base the convolution example on the simpler stream example
This commit is contained in:
Chris Robinson
+1
-1
@@ -1508,7 +1508,7 @@ if(ALSOFT_EXAMPLES)
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target_link_libraries(alstreamcb PRIVATE ${LINKER_FLAGS} SndFile::SndFile ex-common
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${UNICODE_FLAG})
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add_executable(alconvolve examples/alconvolve.cpp)
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add_executable(alconvolve examples/alconvolve.c)
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target_link_libraries(alconvolve PRIVATE ${LINKER_FLAGS} common SndFile::SndFile ex-common
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${UNICODE_FLAG})
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@@ -0,0 +1,515 @@
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/*
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* OpenAL Convolution Reverb Example
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*
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* Copyright (c) 2020 by Chris Robinson <chris.kcat@gmail.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/* This file contains an example for applying convolution reverb to a source. */
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#include <assert.h>
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#include <inttypes.h>
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#include <limits.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include "sndfile.h"
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#include "AL/al.h"
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#include "AL/alext.h"
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#include "common/alhelpers.h"
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#ifndef AL_SOFT_convolution_reverb
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#define AL_SOFT_convolution_reverb
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#define AL_EFFECT_CONVOLUTION_REVERB_SOFT 0xA000
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#endif
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/* Effect object functions */
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static LPALGENEFFECTS alGenEffects;
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static LPALDELETEEFFECTS alDeleteEffects;
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static LPALISEFFECT alIsEffect;
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static LPALEFFECTI alEffecti;
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static LPALEFFECTIV alEffectiv;
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static LPALEFFECTF alEffectf;
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static LPALEFFECTFV alEffectfv;
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static LPALGETEFFECTI alGetEffecti;
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static LPALGETEFFECTIV alGetEffectiv;
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static LPALGETEFFECTF alGetEffectf;
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static LPALGETEFFECTFV alGetEffectfv;
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/* Auxiliary Effect Slot object functions */
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static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
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static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
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static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
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static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
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static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
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static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
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static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
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static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
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static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
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static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
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static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
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/* This stuff defines a simple streaming player object, the same as alstream.c.
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* Comments are removed for brevity, see alstream.c for more details.
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*/
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#define NUM_BUFFERS 4
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#define BUFFER_SAMPLES 8192
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typedef struct StreamPlayer {
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ALuint buffers[NUM_BUFFERS];
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ALuint source;
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SNDFILE *sndfile;
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SF_INFO sfinfo;
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float *membuf;
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ALenum format;
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} StreamPlayer;
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static StreamPlayer *NewPlayer(void)
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{
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StreamPlayer *player;
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player = calloc(1, sizeof(*player));
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assert(player != NULL);
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alGenBuffers(NUM_BUFFERS, player->buffers);
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assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
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alGenSources(1, &player->source);
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assert(alGetError() == AL_NO_ERROR && "Could not create source");
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alSource3i(player->source, AL_POSITION, 0, 0, -1);
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alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
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alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
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assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
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return player;
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}
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static void ClosePlayerFile(StreamPlayer *player)
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{
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if(player->sndfile)
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sf_close(player->sndfile);
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player->sndfile = NULL;
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free(player->membuf);
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player->membuf = NULL;
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}
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static void DeletePlayer(StreamPlayer *player)
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{
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ClosePlayerFile(player);
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alDeleteSources(1, &player->source);
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alDeleteBuffers(NUM_BUFFERS, player->buffers);
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if(alGetError() != AL_NO_ERROR)
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fprintf(stderr, "Failed to delete object IDs\n");
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memset(player, 0, sizeof(*player));
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free(player);
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}
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static int OpenPlayerFile(StreamPlayer *player, const char *filename)
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{
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size_t frame_size;
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ClosePlayerFile(player);
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player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
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if(!player->sndfile)
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{
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fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
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return 0;
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}
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if(player->sfinfo.channels == 1)
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player->format = AL_FORMAT_MONO_FLOAT32;
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else if(player->sfinfo.channels == 2)
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player->format = AL_FORMAT_STEREO_FLOAT32;
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else if(player->sfinfo.channels == 6)
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player->format = AL_FORMAT_51CHN32;
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else
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{
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fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
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sf_close(player->sndfile);
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player->sndfile = NULL;
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return 0;
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}
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frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(float);
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player->membuf = malloc(frame_size);
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return 1;
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}
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static int StartPlayer(StreamPlayer *player)
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{
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ALsizei i;
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alSourceRewind(player->source);
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alSourcei(player->source, AL_BUFFER, 0);
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for(i = 0;i < NUM_BUFFERS;i++)
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{
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sf_count_t slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
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if(slen < 1) break;
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slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
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alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
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player->sfinfo.samplerate);
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}
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error buffering for playback\n");
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return 0;
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}
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alSourceQueueBuffers(player->source, i, player->buffers);
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alSourcePlay(player->source);
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error starting playback\n");
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return 0;
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}
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return 1;
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}
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static int UpdatePlayer(StreamPlayer *player)
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{
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ALint processed, state;
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alGetSourcei(player->source, AL_SOURCE_STATE, &state);
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alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error checking source state\n");
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return 0;
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}
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while(processed > 0)
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{
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ALuint bufid;
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sf_count_t slen;
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alSourceUnqueueBuffers(player->source, 1, &bufid);
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processed--;
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slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
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if(slen > 0)
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{
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slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
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alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
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player->sfinfo.samplerate);
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alSourceQueueBuffers(player->source, 1, &bufid);
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}
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error buffering data\n");
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return 0;
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}
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}
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if(state != AL_PLAYING && state != AL_PAUSED)
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{
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ALint queued;
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alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
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if(queued == 0)
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return 0;
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alSourcePlay(player->source);
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if(alGetError() != AL_NO_ERROR)
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{
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fprintf(stderr, "Error restarting playback\n");
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return 0;
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}
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}
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return 1;
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}
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/* CreateEffect creates a new OpenAL effect object with a convolution reverb
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* type, and returns the new effect ID.
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*/
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static ALuint CreateEffect(void)
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{
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ALuint effect = 0;
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ALenum err;
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printf("Using Convolution Reverb\n");
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/* Create the effect object and set the convolution reverb effect type. */
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alGenEffects(1, &effect);
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alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
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/* Check if an error occured, and clean up if so. */
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err = alGetError();
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if(err != AL_NO_ERROR)
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{
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fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
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if(alIsEffect(effect))
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alDeleteEffects(1, &effect);
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return 0;
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}
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return effect;
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}
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/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
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* returns the new buffer ID.
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*/
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static ALuint LoadSound(const char *filename)
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{
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ALenum err, format;
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ALuint buffer;
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SNDFILE *sndfile;
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SF_INFO sfinfo;
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float *membuf;
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sf_count_t num_frames;
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ALsizei num_bytes;
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/* Open the audio file and check that it's usable. */
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sndfile = sf_open(filename, SFM_READ, &sfinfo);
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if(!sndfile)
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{
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fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
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return 0;
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}
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if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(float))/sfinfo.channels)
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{
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fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
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sf_close(sndfile);
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return 0;
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}
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/* Get the sound format, and figure out the OpenAL format. Use floats since
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* impulse responses will usually have more than 16-bit precision.
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*/
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if(sfinfo.channels == 1)
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format = AL_FORMAT_MONO_FLOAT32;
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else if(sfinfo.channels == 2)
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format = AL_FORMAT_STEREO_FLOAT32;
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else
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{
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fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
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sf_close(sndfile);
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return 0;
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}
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/* Decode the whole audio file to a buffer. */
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membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(float));
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num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
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if(num_frames < 1)
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{
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free(membuf);
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sf_close(sndfile);
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fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
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return 0;
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}
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num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(float);
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/* Buffer the audio data into a new buffer object, then free the data and
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* close the file.
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*/
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buffer = 0;
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alGenBuffers(1, &buffer);
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alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
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free(membuf);
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sf_close(sndfile);
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/* Check if an error occured, and clean up if so. */
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err = alGetError();
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if(err != AL_NO_ERROR)
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{
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fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
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if(buffer && alIsBuffer(buffer))
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alDeleteBuffers(1, &buffer);
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return 0;
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}
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return buffer;
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}
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int main(int argc, char **argv)
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{
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ALuint ir_buffer, effect, slot;
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StreamPlayer *player;
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int i;
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/* Print out usage if no arguments were specified */
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if(argc < 2)
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{
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fprintf(stderr, "Usage: %s [-device <name>] <impulse response file> <filenames...>\n",
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argv[0]);
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return 1;
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}
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argv++; argc--;
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if(InitAL(&argv, &argc) != 0)
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return 1;
|
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|
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if(!alcIsExtensionPresent(alcGetContextsDevice(alcGetCurrentContext()), "ALC_EXT_EFX"))
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{
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CloseAL();
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fprintf(stderr, "Error: EFX not supported\n");
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return 1;
|
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}
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if(argc < 2)
|
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{
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CloseAL();
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fprintf(stderr, "Error: Missing impulse response or sound files\n");
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return 1;
|
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}
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/* Define a macro to help load the function pointers. */
|
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#define LOAD_PROC(T, x) ((x) = (T)alGetProcAddress(#x))
|
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LOAD_PROC(LPALGENEFFECTS, alGenEffects);
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LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
|
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LOAD_PROC(LPALISEFFECT, alIsEffect);
|
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LOAD_PROC(LPALEFFECTI, alEffecti);
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LOAD_PROC(LPALEFFECTIV, alEffectiv);
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LOAD_PROC(LPALEFFECTF, alEffectf);
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LOAD_PROC(LPALEFFECTFV, alEffectfv);
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LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
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LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
|
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LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
|
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LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
|
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|
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LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
|
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LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
|
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LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
|
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LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
|
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LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
|
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LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
|
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LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
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LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
|
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#undef LOAD_PROC
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/* Load the impulse response sound into a buffer. */
|
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ir_buffer = LoadSound(argv[0]);
|
||||
if(!ir_buffer)
|
||||
{
|
||||
CloseAL();
|
||||
return 1;
|
||||
}
|
||||
|
||||
/* Load the reverb into an effect. */
|
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effect = CreateEffect();
|
||||
if(!effect)
|
||||
{
|
||||
alDeleteBuffers(1, &ir_buffer);
|
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CloseAL();
|
||||
return 1;
|
||||
}
|
||||
|
||||
/* Create the effect slot object. This is what "plays" an effect on sources
|
||||
* that connect to it.
|
||||
*/
|
||||
slot = 0;
|
||||
alGenAuxiliaryEffectSlots(1, &slot);
|
||||
|
||||
/* Set the impulse response sound buffer on the effect slot. This allows
|
||||
* effects to access it as needed. In this case, convolution reverb uses it
|
||||
* as the filter source. NOTE: Unlike the effect object, the buffer *is*
|
||||
* kept referenced and may not be changed or deleted as long as it's set,
|
||||
* just like with a source. When another buffer is set, or the effect slot
|
||||
* is deleted, the buffer reference is released.
|
||||
*
|
||||
* The effect slot's gain is reduced because the impulse responses I've
|
||||
* tested with result in excessively loud reverb. Is that normal? Even with
|
||||
* this, it seems a bit on the loud side.
|
||||
*
|
||||
* Also note: unlike standard or EAX reverb, there is no automatic
|
||||
* attenuation of a source's reverb response with distance, so the reverb
|
||||
* will remain full volume regardless of a given sound's distance from the
|
||||
* listener. You can use a send filter to alter a given source's
|
||||
* contribution to reverb.
|
||||
*/
|
||||
alAuxiliaryEffectSloti(slot, AL_BUFFER, (ALint)ir_buffer);
|
||||
alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
|
||||
alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, (ALint)effect);
|
||||
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
|
||||
|
||||
player = NewPlayer();
|
||||
/* Connect the player's source to the effect slot. */
|
||||
alSource3i(player->source, AL_AUXILIARY_SEND_FILTER, (ALint)slot, 0, AL_FILTER_NULL);
|
||||
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
|
||||
|
||||
/* Play each file listed on the command line */
|
||||
for(i = 1;i < argc;i++)
|
||||
{
|
||||
const char *namepart;
|
||||
|
||||
if(!OpenPlayerFile(player, argv[i]))
|
||||
continue;
|
||||
|
||||
namepart = strrchr(argv[i], '/');
|
||||
if(namepart || (namepart=strrchr(argv[i], '\\')))
|
||||
namepart++;
|
||||
else
|
||||
namepart = argv[i];
|
||||
|
||||
printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
|
||||
player->sfinfo.samplerate);
|
||||
fflush(stdout);
|
||||
|
||||
if(!StartPlayer(player))
|
||||
{
|
||||
ClosePlayerFile(player);
|
||||
continue;
|
||||
}
|
||||
|
||||
while(UpdatePlayer(player))
|
||||
al_nssleep(10000000);
|
||||
|
||||
ClosePlayerFile(player);
|
||||
}
|
||||
printf("Done.\n");
|
||||
|
||||
/* All files done. Delete the player and effect resources, and close down
|
||||
* OpenAL.
|
||||
*/
|
||||
DeletePlayer(player);
|
||||
player = NULL;
|
||||
|
||||
alDeleteAuxiliaryEffectSlots(1, &slot);
|
||||
alDeleteEffects(1, &effect);
|
||||
alDeleteBuffers(1, &ir_buffer);
|
||||
|
||||
CloseAL();
|
||||
|
||||
return 0;
|
||||
}
|
||||
@@ -1,536 +0,0 @@
|
||||
/*
|
||||
* OpenAL Convolution Reverb Example
|
||||
*
|
||||
* Copyright (c) 2020 by Chris Robinson <chris.kcat@gmail.com>
|
||||
*
|
||||
* 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.
|
||||
*/
|
||||
|
||||
/* This file contains a streaming audio player, using the convolution reverb
|
||||
* effect.
|
||||
*/
|
||||
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <stdio.h>
|
||||
|
||||
#include <atomic>
|
||||
#include <cassert>
|
||||
#include <chrono>
|
||||
#include <limits>
|
||||
#include <memory>
|
||||
#include <stdexcept>
|
||||
#include <string>
|
||||
#include <thread>
|
||||
#include <vector>
|
||||
|
||||
#include "sndfile.h"
|
||||
|
||||
#include "AL/al.h"
|
||||
#include "AL/alc.h"
|
||||
#include "AL/alext.h"
|
||||
|
||||
#include "common/alhelpers.h"
|
||||
|
||||
|
||||
#ifndef AL_SOFT_callback_buffer
|
||||
#define AL_SOFT_callback_buffer
|
||||
typedef unsigned int ALbitfieldSOFT;
|
||||
#define AL_BUFFER_CALLBACK_FUNCTION_SOFT 0x19A0
|
||||
#define AL_BUFFER_CALLBACK_USER_PARAM_SOFT 0x19A1
|
||||
typedef ALsizei (AL_APIENTRY*LPALBUFFERCALLBACKTYPESOFT)(ALvoid *userptr, ALvoid *sampledata, ALsizei numsamples);
|
||||
typedef void (AL_APIENTRY*LPALBUFFERCALLBACKSOFT)(ALuint buffer, ALenum format, ALsizei freq, LPALBUFFERCALLBACKTYPESOFT callback, ALvoid *userptr, ALbitfieldSOFT flags);
|
||||
typedef void (AL_APIENTRY*LPALGETBUFFERPTRSOFT)(ALuint buffer, ALenum param, ALvoid **value);
|
||||
typedef void (AL_APIENTRY*LPALGETBUFFER3PTRSOFT)(ALuint buffer, ALenum param, ALvoid **value1, ALvoid **value2, ALvoid **value3);
|
||||
typedef void (AL_APIENTRY*LPALGETBUFFERPTRVSOFT)(ALuint buffer, ALenum param, ALvoid **values);
|
||||
#endif
|
||||
|
||||
#ifndef AL_SOFT_convolution_reverb
|
||||
#define AL_SOFT_convolution_reverb
|
||||
#define AL_EFFECT_CONVOLUTION_REVERB_SOFT 0xA000
|
||||
#endif
|
||||
|
||||
|
||||
namespace {
|
||||
|
||||
/* Effect object functions */
|
||||
LPALGENEFFECTS alGenEffects;
|
||||
LPALDELETEEFFECTS alDeleteEffects;
|
||||
LPALISEFFECT alIsEffect;
|
||||
LPALEFFECTI alEffecti;
|
||||
LPALEFFECTIV alEffectiv;
|
||||
LPALEFFECTF alEffectf;
|
||||
LPALEFFECTFV alEffectfv;
|
||||
LPALGETEFFECTI alGetEffecti;
|
||||
LPALGETEFFECTIV alGetEffectiv;
|
||||
LPALGETEFFECTF alGetEffectf;
|
||||
LPALGETEFFECTFV alGetEffectfv;
|
||||
|
||||
/* Auxiliary Effect Slot object functions */
|
||||
LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
|
||||
LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
|
||||
LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
|
||||
LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
|
||||
LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
|
||||
LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
|
||||
LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
|
||||
LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
|
||||
LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
|
||||
LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
|
||||
LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
|
||||
|
||||
|
||||
ALuint CreateEffect()
|
||||
{
|
||||
/* Create the effect object and try to set convolution reverb. */
|
||||
ALuint effect{0};
|
||||
alGenEffects(1, &effect);
|
||||
|
||||
printf("Using Convolution Reverb\n");
|
||||
|
||||
alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
|
||||
|
||||
/* Check if an error occured, and clean up if so. */
|
||||
if(ALenum err{alGetError()})
|
||||
{
|
||||
fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
|
||||
if(alIsEffect(effect))
|
||||
alDeleteEffects(1, &effect);
|
||||
return 0;
|
||||
}
|
||||
|
||||
return effect;
|
||||
}
|
||||
|
||||
|
||||
ALuint LoadSound(const char *filename)
|
||||
{
|
||||
/* Open the audio file and check that it's usable. */
|
||||
SF_INFO sfinfo{};
|
||||
SNDFILE *sndfile{sf_open(filename, SFM_READ, &sfinfo)};
|
||||
if(!sndfile)
|
||||
{
|
||||
fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
|
||||
return 0;
|
||||
}
|
||||
constexpr sf_count_t max_samples{std::numeric_limits<int>::max() / sizeof(float)};
|
||||
if(sfinfo.frames < 1 || sfinfo.frames > max_samples/sfinfo.channels)
|
||||
{
|
||||
fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
|
||||
sf_close(sndfile);
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* Get the sound format, and figure out the OpenAL format. Use a float
|
||||
* format since impulse responses are keen on having a low noise floor.
|
||||
*/
|
||||
ALenum format{};
|
||||
if(sfinfo.channels == 1)
|
||||
format = AL_FORMAT_MONO_FLOAT32;
|
||||
else if(sfinfo.channels == 2)
|
||||
format = AL_FORMAT_STEREO_FLOAT32;
|
||||
else
|
||||
{
|
||||
fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
|
||||
sf_close(sndfile);
|
||||
return 0;
|
||||
}
|
||||
|
||||
auto membuf = std::make_unique<float[]>(static_cast<size_t>(sfinfo.frames * sfinfo.channels));
|
||||
|
||||
sf_count_t num_frames{sf_readf_float(sndfile, membuf.get(), sfinfo.frames)};
|
||||
if(num_frames < 1)
|
||||
{
|
||||
membuf = nullptr;
|
||||
sf_close(sndfile);
|
||||
fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
|
||||
return 0;
|
||||
}
|
||||
const auto num_bytes = static_cast<ALsizei>(num_frames * sfinfo.channels) *
|
||||
ALsizei{sizeof(float)};
|
||||
|
||||
ALuint buffer{0};
|
||||
alGenBuffers(1, &buffer);
|
||||
alBufferData(buffer, format, membuf.get(), num_bytes, sfinfo.samplerate);
|
||||
|
||||
membuf = nullptr;
|
||||
sf_close(sndfile);
|
||||
|
||||
if(ALenum err{alGetError()})
|
||||
{
|
||||
fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
|
||||
if(buffer && alIsBuffer(buffer))
|
||||
alDeleteBuffers(1, &buffer);
|
||||
return 0;
|
||||
}
|
||||
|
||||
return buffer;
|
||||
}
|
||||
|
||||
|
||||
/* This is largely the same as in alstreamcb.cpp. Comments removed for brevity,
|
||||
* see the aforementioned source for more details.
|
||||
*/
|
||||
using std::chrono::seconds;
|
||||
using std::chrono::nanoseconds;
|
||||
|
||||
LPALBUFFERCALLBACKSOFT alBufferCallbackSOFT;
|
||||
|
||||
struct StreamPlayer {
|
||||
std::unique_ptr<ALbyte[]> mBufferData;
|
||||
size_t mBufferDataSize{0};
|
||||
std::atomic<size_t> mReadPos{0};
|
||||
std::atomic<size_t> mWritePos{0};
|
||||
|
||||
ALuint mBuffer{0}, mSource{0};
|
||||
size_t mStartOffset{0};
|
||||
|
||||
SNDFILE *mSndfile{nullptr};
|
||||
SF_INFO mSfInfo{};
|
||||
size_t mDecoderOffset{0};
|
||||
|
||||
ALenum mFormat;
|
||||
|
||||
StreamPlayer()
|
||||
{
|
||||
alGenBuffers(1, &mBuffer);
|
||||
if(ALenum err{alGetError()})
|
||||
throw std::runtime_error{"alGenBuffers failed"};
|
||||
alGenSources(1, &mSource);
|
||||
if(ALenum err{alGetError()})
|
||||
{
|
||||
alDeleteBuffers(1, &mBuffer);
|
||||
throw std::runtime_error{"alGenSources failed"};
|
||||
}
|
||||
}
|
||||
~StreamPlayer()
|
||||
{
|
||||
alDeleteSources(1, &mSource);
|
||||
alDeleteBuffers(1, &mBuffer);
|
||||
if(mSndfile)
|
||||
sf_close(mSndfile);
|
||||
}
|
||||
|
||||
void close()
|
||||
{
|
||||
if(mSndfile)
|
||||
{
|
||||
alSourceRewind(mSource);
|
||||
alSourcei(mSource, AL_BUFFER, 0);
|
||||
sf_close(mSndfile);
|
||||
mSndfile = nullptr;
|
||||
}
|
||||
}
|
||||
|
||||
bool open(const char *filename)
|
||||
{
|
||||
close();
|
||||
|
||||
mSndfile = sf_open(filename, SFM_READ, &mSfInfo);
|
||||
if(!mSndfile)
|
||||
{
|
||||
fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(mSndfile));
|
||||
return false;
|
||||
}
|
||||
|
||||
mFormat = AL_NONE;
|
||||
if(mSfInfo.channels == 1)
|
||||
mFormat = AL_FORMAT_MONO_FLOAT32;
|
||||
else if(mSfInfo.channels == 2)
|
||||
mFormat = AL_FORMAT_STEREO_FLOAT32;
|
||||
else if(mSfInfo.channels == 6)
|
||||
mFormat = AL_FORMAT_51CHN32;
|
||||
else
|
||||
{
|
||||
fprintf(stderr, "Unsupported channel count: %d\n", mSfInfo.channels);
|
||||
sf_close(mSndfile);
|
||||
mSndfile = nullptr;
|
||||
|
||||
return false;
|
||||
}
|
||||
|
||||
mBufferDataSize = static_cast<ALuint>(mSfInfo.samplerate*mSfInfo.channels) * sizeof(float);
|
||||
mBufferData.reset(new ALbyte[mBufferDataSize]);
|
||||
mReadPos.store(0, std::memory_order_relaxed);
|
||||
mWritePos.store(0, std::memory_order_relaxed);
|
||||
mDecoderOffset = 0;
|
||||
|
||||
return true;
|
||||
}
|
||||
|
||||
static ALsizei AL_APIENTRY bufferCallbackC(void *userptr, void *data, ALsizei size)
|
||||
{ return static_cast<StreamPlayer*>(userptr)->bufferCallback(data, size); }
|
||||
ALsizei bufferCallback(void *data, ALsizei size)
|
||||
{
|
||||
ALsizei got{0};
|
||||
|
||||
size_t roffset{mReadPos.load(std::memory_order_acquire)};
|
||||
while(got < size)
|
||||
{
|
||||
const size_t woffset{mWritePos.load(std::memory_order_relaxed)};
|
||||
if(woffset == roffset) break;
|
||||
|
||||
size_t todo{((woffset < roffset) ? mBufferDataSize : woffset) - roffset};
|
||||
todo = std::min<size_t>(todo, static_cast<ALuint>(size-got));
|
||||
|
||||
memcpy(data, &mBufferData[roffset], todo);
|
||||
data = static_cast<ALbyte*>(data) + todo;
|
||||
got += static_cast<ALsizei>(todo);
|
||||
|
||||
roffset += todo;
|
||||
if(roffset == mBufferDataSize)
|
||||
roffset = 0;
|
||||
}
|
||||
mReadPos.store(roffset, std::memory_order_release);
|
||||
|
||||
return got;
|
||||
}
|
||||
|
||||
bool prepare()
|
||||
{
|
||||
alBufferCallbackSOFT(mBuffer, mFormat, mSfInfo.samplerate, bufferCallbackC, this, 0);
|
||||
alSourcei(mSource, AL_BUFFER, static_cast<ALint>(mBuffer));
|
||||
if(ALenum err{alGetError()})
|
||||
{
|
||||
fprintf(stderr, "Failed to set callback: %s (0x%04x)\n", alGetString(err), err);
|
||||
return false;
|
||||
}
|
||||
return true;
|
||||
}
|
||||
|
||||
bool update()
|
||||
{
|
||||
ALenum state;
|
||||
ALint pos;
|
||||
alGetSourcei(mSource, AL_SAMPLE_OFFSET, &pos);
|
||||
alGetSourcei(mSource, AL_SOURCE_STATE, &state);
|
||||
|
||||
const size_t frame_size{static_cast<ALuint>(mSfInfo.channels) * sizeof(float)};
|
||||
size_t woffset{mWritePos.load(std::memory_order_acquire)};
|
||||
if(state != AL_INITIAL)
|
||||
{
|
||||
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
|
||||
const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
|
||||
roffset};
|
||||
const size_t curtime{((state==AL_STOPPED) ? (mDecoderOffset-readable) / frame_size
|
||||
: (static_cast<ALuint>(pos) + mStartOffset/frame_size))
|
||||
/ static_cast<ALuint>(mSfInfo.samplerate)};
|
||||
printf("\r%3zus (%3zu%% full)", curtime, readable * 100 / mBufferDataSize);
|
||||
}
|
||||
else
|
||||
fputs("Starting...", stdout);
|
||||
fflush(stdout);
|
||||
|
||||
while(!sf_error(mSndfile))
|
||||
{
|
||||
size_t read_bytes;
|
||||
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
|
||||
if(roffset > woffset)
|
||||
{
|
||||
const size_t writable{roffset-woffset-1};
|
||||
if(writable < frame_size) break;
|
||||
|
||||
sf_count_t num_frames{sf_readf_float(mSndfile,
|
||||
reinterpret_cast<float*>(&mBufferData[woffset]),
|
||||
static_cast<sf_count_t>(writable/frame_size))};
|
||||
if(num_frames < 1) break;
|
||||
|
||||
read_bytes = static_cast<size_t>(num_frames) * frame_size;
|
||||
woffset += read_bytes;
|
||||
}
|
||||
else
|
||||
{
|
||||
const size_t writable{!roffset ? mBufferDataSize-woffset-1 :
|
||||
(mBufferDataSize-woffset)};
|
||||
if(writable < frame_size) break;
|
||||
|
||||
sf_count_t num_frames{sf_readf_float(mSndfile,
|
||||
reinterpret_cast<float*>(&mBufferData[woffset]),
|
||||
static_cast<sf_count_t>(writable/frame_size))};
|
||||
if(num_frames < 1) break;
|
||||
|
||||
read_bytes = static_cast<size_t>(num_frames) * frame_size;
|
||||
woffset += read_bytes;
|
||||
if(woffset == mBufferDataSize)
|
||||
woffset = 0;
|
||||
}
|
||||
mWritePos.store(woffset, std::memory_order_release);
|
||||
mDecoderOffset += read_bytes;
|
||||
}
|
||||
|
||||
if(state != AL_PLAYING && state != AL_PAUSED)
|
||||
{
|
||||
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
|
||||
const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
|
||||
roffset};
|
||||
if(readable == 0)
|
||||
return false;
|
||||
|
||||
mStartOffset = mDecoderOffset - readable;
|
||||
alSourcePlay(mSource);
|
||||
if(alGetError() != AL_NO_ERROR)
|
||||
return false;
|
||||
}
|
||||
return true;
|
||||
}
|
||||
};
|
||||
|
||||
} // namespace
|
||||
|
||||
int main(int argc, char **argv)
|
||||
{
|
||||
/* A simple RAII container for OpenAL startup and shutdown. */
|
||||
struct AudioManager {
|
||||
AudioManager(char ***argv_, int *argc_)
|
||||
{
|
||||
if(InitAL(argv_, argc_) != 0)
|
||||
throw std::runtime_error{"Failed to initialize OpenAL"};
|
||||
}
|
||||
~AudioManager() { CloseAL(); }
|
||||
};
|
||||
|
||||
/* Print out usage if no arguments were specified */
|
||||
if(argc < 2)
|
||||
{
|
||||
fprintf(stderr, "Usage: %s [-device <name>] <impulse response sound> [sound files...]\n",
|
||||
argv[0]);
|
||||
return 1;
|
||||
}
|
||||
|
||||
argv++; argc--;
|
||||
AudioManager almgr{&argv, &argc};
|
||||
|
||||
if(!alIsExtensionPresent("AL_SOFTX_callback_buffer"))
|
||||
{
|
||||
fprintf(stderr, "AL_SOFT_callback_buffer extension not available\n");
|
||||
return 1;
|
||||
}
|
||||
|
||||
/* Define a macro to help load the function pointers. */
|
||||
#define LOAD_PROC(T, x) ((x) = reinterpret_cast<T>(alGetProcAddress(#x)))
|
||||
LOAD_PROC(LPALBUFFERCALLBACKSOFT, alBufferCallbackSOFT);
|
||||
|
||||
LOAD_PROC(LPALGENEFFECTS, alGenEffects);
|
||||
LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
|
||||
LOAD_PROC(LPALISEFFECT, alIsEffect);
|
||||
LOAD_PROC(LPALEFFECTI, alEffecti);
|
||||
LOAD_PROC(LPALEFFECTIV, alEffectiv);
|
||||
LOAD_PROC(LPALEFFECTF, alEffectf);
|
||||
LOAD_PROC(LPALEFFECTFV, alEffectfv);
|
||||
LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
|
||||
LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
|
||||
LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
|
||||
LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
|
||||
|
||||
LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
|
||||
LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
|
||||
LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
|
||||
LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
|
||||
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
|
||||
#undef LOAD_PROC
|
||||
|
||||
/* Load the impulse response sound file into a buffer. */
|
||||
ALuint buffer{LoadSound(argv[0])};
|
||||
if(!buffer) return 1;
|
||||
|
||||
/* Create the convolution reverb effect. */
|
||||
ALuint effect{CreateEffect()};
|
||||
if(!effect)
|
||||
{
|
||||
alDeleteBuffers(1, &buffer);
|
||||
return 1;
|
||||
}
|
||||
|
||||
/* Create the effect slot object. This is what "plays" an effect on sources
|
||||
* that connect to it. */
|
||||
ALuint slot{0};
|
||||
alGenAuxiliaryEffectSlots(1, &slot);
|
||||
|
||||
/* Set the impulse response sound buffer on the effect slot. This allows
|
||||
* effects to access it as needed. In this case, convolution reverb uses it
|
||||
* as the filter source. NOTE: Unlike the effect object, the buffer *is*
|
||||
* kept referenced and may not be changed or deleted as long as it's set,
|
||||
* just like with a source. When another buffer is set, or the effect slot
|
||||
* is deleted, the buffer reference is released.
|
||||
*
|
||||
* The effect slot's gain is reduced because the impulse responses I've
|
||||
* tested with result in excessively loud reverb. Is that normal? Even with
|
||||
* this, it seems a bit on the loud side.
|
||||
*
|
||||
* Also note: unlike standard or EAX reverb, there is no automatic
|
||||
* attenuation of a source's reverb response with distance, so the reverb
|
||||
* will remain full volume regardless of a given sound's distance from the
|
||||
* listener. You can use a send filter to alter a given source's
|
||||
* contribution to reverb.
|
||||
*/
|
||||
alAuxiliaryEffectSloti(slot, AL_BUFFER, static_cast<ALint>(buffer));
|
||||
alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
|
||||
alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, static_cast<ALint>(effect));
|
||||
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
|
||||
|
||||
ALCint refresh{25};
|
||||
alcGetIntegerv(alcGetContextsDevice(alcGetCurrentContext()), ALC_REFRESH, 1, &refresh);
|
||||
|
||||
std::unique_ptr<StreamPlayer> player{new StreamPlayer{}};
|
||||
alSource3i(player->mSource, AL_AUXILIARY_SEND_FILTER, static_cast<ALint>(slot), 0,
|
||||
AL_FILTER_NULL);
|
||||
|
||||
for(int i{1};i < argc;++i)
|
||||
{
|
||||
if(!player->open(argv[i]))
|
||||
continue;
|
||||
|
||||
const char *namepart{strrchr(argv[i], '/')};
|
||||
if(namepart || (namepart=strrchr(argv[i], '\\')))
|
||||
++namepart;
|
||||
else
|
||||
namepart = argv[i];
|
||||
|
||||
printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->mFormat),
|
||||
player->mSfInfo.samplerate);
|
||||
fflush(stdout);
|
||||
|
||||
if(!player->prepare())
|
||||
{
|
||||
player->close();
|
||||
continue;
|
||||
}
|
||||
|
||||
while(player->update())
|
||||
std::this_thread::sleep_for(nanoseconds{seconds{1}} / refresh);
|
||||
putc('\n', stdout);
|
||||
|
||||
player->close();
|
||||
}
|
||||
/* All done. */
|
||||
printf("Done.\n");
|
||||
|
||||
player = nullptr;
|
||||
alDeleteAuxiliaryEffectSlots(1, &slot);
|
||||
alDeleteEffects(1, &effect);
|
||||
alDeleteBuffers(1, &buffer);
|
||||
|
||||
return 0;
|
||||
}
|
||||
Reference in New Issue
Block a user