Chris Robinson
1017 lines
34 KiB
C++
1017 lines
34 KiB
C++
/**
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* OpenAL cross platform audio library
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* Copyright (C) 1999-2007 by authors.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include "backends/dsound.h"
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#include <stdlib.h>
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#include <stdio.h>
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#include <memory.h>
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#include <cguid.h>
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#include <mmreg.h>
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#ifndef _WAVEFORMATEXTENSIBLE_
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#include <ks.h>
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#include <ksmedia.h>
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#endif
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#include <atomic>
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#include <thread>
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#include <string>
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#include <vector>
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#include <algorithm>
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#include "alMain.h"
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#include "alu.h"
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#include "ringbuffer.h"
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#include "compat.h"
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/* MinGW-w64 needs this for some unknown reason now. */
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using LPCWAVEFORMATEX = const WAVEFORMATEX*;
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#include <dsound.h>
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#ifndef DSSPEAKER_5POINT1
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# define DSSPEAKER_5POINT1 0x00000006
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#endif
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#ifndef DSSPEAKER_5POINT1_BACK
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# define DSSPEAKER_5POINT1_BACK 0x00000006
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#endif
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#ifndef DSSPEAKER_7POINT1
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# define DSSPEAKER_7POINT1 0x00000007
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#endif
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#ifndef DSSPEAKER_7POINT1_SURROUND
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# define DSSPEAKER_7POINT1_SURROUND 0x00000008
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#endif
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#ifndef DSSPEAKER_5POINT1_SURROUND
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# define DSSPEAKER_5POINT1_SURROUND 0x00000009
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#endif
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/* Some headers seem to define these as macros for __uuidof, which is annoying
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* since some headers don't declare them at all. Hopefully the ifdef is enough
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* to tell if they need to be declared.
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*/
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#ifndef KSDATAFORMAT_SUBTYPE_PCM
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DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
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#endif
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#ifndef KSDATAFORMAT_SUBTYPE_IEEE_FLOAT
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DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
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#endif
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namespace {
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#define DEVNAME_HEAD "OpenAL Soft on "
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#ifdef HAVE_DYNLOAD
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void *ds_handle;
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HRESULT (WINAPI *pDirectSoundCreate)(const GUID *pcGuidDevice, IDirectSound **ppDS, IUnknown *pUnkOuter);
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HRESULT (WINAPI *pDirectSoundEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext);
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HRESULT (WINAPI *pDirectSoundCaptureCreate)(const GUID *pcGuidDevice, IDirectSoundCapture **ppDSC, IUnknown *pUnkOuter);
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HRESULT (WINAPI *pDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext);
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#ifndef IN_IDE_PARSER
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#define DirectSoundCreate pDirectSoundCreate
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#define DirectSoundEnumerateW pDirectSoundEnumerateW
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#define DirectSoundCaptureCreate pDirectSoundCaptureCreate
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#define DirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW
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#endif
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#endif
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bool DSoundLoad(void)
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{
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#ifdef HAVE_DYNLOAD
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if(!ds_handle)
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{
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ds_handle = LoadLib("dsound.dll");
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if(!ds_handle)
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{
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ERR("Failed to load dsound.dll\n");
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return false;
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}
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#define LOAD_FUNC(f) do { \
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p##f = reinterpret_cast<decltype(p##f)>(GetSymbol(ds_handle, #f)); \
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if(!p##f) \
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{ \
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CloseLib(ds_handle); \
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ds_handle = nullptr; \
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return false; \
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} \
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} while(0)
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LOAD_FUNC(DirectSoundCreate);
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LOAD_FUNC(DirectSoundEnumerateW);
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LOAD_FUNC(DirectSoundCaptureCreate);
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LOAD_FUNC(DirectSoundCaptureEnumerateW);
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#undef LOAD_FUNC
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}
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#endif
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return true;
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}
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#define MAX_UPDATES 128
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struct DevMap {
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std::string name;
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GUID guid;
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template<typename T0, typename T1>
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DevMap(T0&& name_, T1&& guid_)
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: name{std::forward<T0>(name_)}, guid{std::forward<T1>(guid_)}
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{ }
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};
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al::vector<DevMap> PlaybackDevices;
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al::vector<DevMap> CaptureDevices;
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bool checkName(const al::vector<DevMap> &list, const std::string &name)
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{
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return std::find_if(list.cbegin(), list.cend(),
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[&name](const DevMap &entry) -> bool
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{ return entry.name == name; }
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) != list.cend();
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}
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BOOL CALLBACK DSoundEnumDevices(GUID *guid, const WCHAR *desc, const WCHAR* UNUSED(drvname), void *data)
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{
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if(!guid)
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return TRUE;
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auto& devices = *reinterpret_cast<al::vector<DevMap>*>(data);
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const std::string basename{DEVNAME_HEAD + wstr_to_utf8(desc)};
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int count{1};
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std::string newname{basename};
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while(checkName(devices, newname))
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{
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newname = basename;
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newname += " #";
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newname += std::to_string(++count);
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}
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devices.emplace_back(std::move(newname), *guid);
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const DevMap &newentry = devices.back();
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OLECHAR *guidstr{nullptr};
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HRESULT hr{StringFromCLSID(*guid, &guidstr)};
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if(SUCCEEDED(hr))
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{
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TRACE("Got device \"%s\", GUID \"%ls\"\n", newentry.name.c_str(), guidstr);
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CoTaskMemFree(guidstr);
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}
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return TRUE;
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}
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struct ALCdsoundPlayback final : public ALCbackend {
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ALCdsoundPlayback(ALCdevice *device) noexcept : ALCbackend{device} { }
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~ALCdsoundPlayback() override;
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int mixerProc();
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IDirectSound *mDS{nullptr};
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IDirectSoundBuffer *mPrimaryBuffer{nullptr};
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IDirectSoundBuffer *mBuffer{nullptr};
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IDirectSoundNotify *mNotifies{nullptr};
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HANDLE mNotifyEvent{nullptr};
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std::atomic<ALenum> mKillNow{AL_TRUE};
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std::thread mThread;
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};
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void ALCdsoundPlayback_Construct(ALCdsoundPlayback *self, ALCdevice *device);
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void ALCdsoundPlayback_Destruct(ALCdsoundPlayback *self);
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ALCenum ALCdsoundPlayback_open(ALCdsoundPlayback *self, const ALCchar *name);
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ALCboolean ALCdsoundPlayback_reset(ALCdsoundPlayback *self);
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ALCboolean ALCdsoundPlayback_start(ALCdsoundPlayback *self);
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void ALCdsoundPlayback_stop(ALCdsoundPlayback *self);
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DECLARE_FORWARD2(ALCdsoundPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
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DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, ALCuint, availableSamples)
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DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, ClockLatency, getClockLatency)
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DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, void, lock)
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DECLARE_FORWARD(ALCdsoundPlayback, ALCbackend, void, unlock)
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DECLARE_DEFAULT_ALLOCATORS(ALCdsoundPlayback)
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DEFINE_ALCBACKEND_VTABLE(ALCdsoundPlayback);
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void ALCdsoundPlayback_Construct(ALCdsoundPlayback *self, ALCdevice *device)
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{
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new (self) ALCdsoundPlayback{device};
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SET_VTABLE2(ALCdsoundPlayback, ALCbackend, self);
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}
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void ALCdsoundPlayback_Destruct(ALCdsoundPlayback *self)
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{ self->~ALCdsoundPlayback(); }
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ALCdsoundPlayback::~ALCdsoundPlayback()
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{
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if(mNotifies)
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mNotifies->Release();
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mNotifies = nullptr;
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if(mBuffer)
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mBuffer->Release();
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mBuffer = nullptr;
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if(mPrimaryBuffer)
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mPrimaryBuffer->Release();
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mPrimaryBuffer = nullptr;
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if(mDS)
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mDS->Release();
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mDS = nullptr;
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if(mNotifyEvent)
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CloseHandle(mNotifyEvent);
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mNotifyEvent = nullptr;
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}
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FORCE_ALIGN int ALCdsoundPlayback::mixerProc()
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{
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SetRTPriority();
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althrd_setname(MIXER_THREAD_NAME);
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DSBCAPS DSBCaps{};
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DSBCaps.dwSize = sizeof(DSBCaps);
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HRESULT err{mBuffer->GetCaps(&DSBCaps)};
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if(FAILED(err))
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{
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ERR("Failed to get buffer caps: 0x%lx\n", err);
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ALCdsoundPlayback_lock(this);
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aluHandleDisconnect(mDevice, "Failure retrieving playback buffer info: 0x%lx", err);
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ALCdsoundPlayback_unlock(this);
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return 1;
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}
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ALsizei FrameSize{mDevice->frameSizeFromFmt()};
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DWORD FragSize{mDevice->UpdateSize * FrameSize};
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bool Playing{false};
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DWORD LastCursor{0u};
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mBuffer->GetCurrentPosition(&LastCursor, nullptr);
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while(!mKillNow.load(std::memory_order_acquire) &&
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mDevice->Connected.load(std::memory_order_acquire))
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{
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// Get current play cursor
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DWORD PlayCursor;
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mBuffer->GetCurrentPosition(&PlayCursor, nullptr);
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DWORD avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes;
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if(avail < FragSize)
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{
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if(!Playing)
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{
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err = mBuffer->Play(0, 0, DSBPLAY_LOOPING);
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if(FAILED(err))
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{
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ERR("Failed to play buffer: 0x%lx\n", err);
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ALCdsoundPlayback_lock(this);
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aluHandleDisconnect(mDevice, "Failure starting playback: 0x%lx", err);
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ALCdsoundPlayback_unlock(this);
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return 1;
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}
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Playing = true;
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}
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avail = WaitForSingleObjectEx(mNotifyEvent, 2000, FALSE);
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if(avail != WAIT_OBJECT_0)
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ERR("WaitForSingleObjectEx error: 0x%lx\n", avail);
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continue;
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}
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avail -= avail%FragSize;
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// Lock output buffer
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void *WritePtr1, *WritePtr2;
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DWORD WriteCnt1{0u}, WriteCnt2{0u};
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err = mBuffer->Lock(LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
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// If the buffer is lost, restore it and lock
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if(err == DSERR_BUFFERLOST)
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{
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WARN("Buffer lost, restoring...\n");
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err = mBuffer->Restore();
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if(SUCCEEDED(err))
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{
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Playing = false;
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LastCursor = 0;
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err = mBuffer->Lock(0, DSBCaps.dwBufferBytes, &WritePtr1, &WriteCnt1,
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&WritePtr2, &WriteCnt2, 0);
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}
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}
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// Successfully locked the output buffer
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if(SUCCEEDED(err))
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{
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// If we have an active context, mix data directly into output buffer otherwise fill with silence
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ALCdsoundPlayback_lock(this);
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aluMixData(mDevice, WritePtr1, WriteCnt1/FrameSize);
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if(WriteCnt2 > 0)
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aluMixData(mDevice, WritePtr2, WriteCnt2/FrameSize);
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ALCdsoundPlayback_unlock(this);
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// Unlock output buffer only when successfully locked
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mBuffer->Unlock(WritePtr1, WriteCnt1, WritePtr2, WriteCnt2);
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}
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else
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{
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ERR("Buffer lock error: %#lx\n", err);
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ALCdsoundPlayback_lock(this);
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aluHandleDisconnect(mDevice, "Failed to lock output buffer: 0x%lx", err);
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ALCdsoundPlayback_unlock(this);
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return 1;
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}
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// Update old write cursor location
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LastCursor += WriteCnt1+WriteCnt2;
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LastCursor %= DSBCaps.dwBufferBytes;
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}
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return 0;
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}
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ALCenum ALCdsoundPlayback_open(ALCdsoundPlayback *self, const ALCchar *deviceName)
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{
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ALCdevice *device{self->mDevice};
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HRESULT hr;
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if(PlaybackDevices.empty())
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{
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/* Initialize COM to prevent name truncation */
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HRESULT hrcom{CoInitialize(nullptr)};
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hr = DirectSoundEnumerateW(DSoundEnumDevices, &PlaybackDevices);
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if(FAILED(hr))
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ERR("Error enumerating DirectSound devices (0x%lx)!\n", hr);
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if(SUCCEEDED(hrcom))
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CoUninitialize();
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}
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const GUID *guid{nullptr};
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if(!deviceName && !PlaybackDevices.empty())
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{
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deviceName = PlaybackDevices[0].name.c_str();
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guid = &PlaybackDevices[0].guid;
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}
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else
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{
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auto iter = std::find_if(PlaybackDevices.cbegin(), PlaybackDevices.cend(),
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[deviceName](const DevMap &entry) -> bool
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{ return entry.name == deviceName; }
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);
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if(iter == PlaybackDevices.cend())
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return ALC_INVALID_VALUE;
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guid = &iter->guid;
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}
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hr = DS_OK;
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self->mNotifyEvent = CreateEventW(nullptr, FALSE, FALSE, nullptr);
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if(!self->mNotifyEvent) hr = E_FAIL;
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//DirectSound Init code
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if(SUCCEEDED(hr))
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hr = DirectSoundCreate(guid, &self->mDS, nullptr);
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if(SUCCEEDED(hr))
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hr = self->mDS->SetCooperativeLevel(GetForegroundWindow(), DSSCL_PRIORITY);
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if(FAILED(hr))
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{
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ERR("Device init failed: 0x%08lx\n", hr);
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return ALC_INVALID_VALUE;
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}
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device->DeviceName = deviceName;
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return ALC_NO_ERROR;
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}
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ALCboolean ALCdsoundPlayback_reset(ALCdsoundPlayback *self)
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{
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ALCdevice *device{self->mDevice};
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if(self->mNotifies)
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self->mNotifies->Release();
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self->mNotifies = nullptr;
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if(self->mBuffer)
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self->mBuffer->Release();
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self->mBuffer = nullptr;
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if(self->mPrimaryBuffer)
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self->mPrimaryBuffer->Release();
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self->mPrimaryBuffer = nullptr;
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switch(device->FmtType)
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{
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case DevFmtByte:
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device->FmtType = DevFmtUByte;
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break;
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case DevFmtFloat:
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if((device->Flags&DEVICE_SAMPLE_TYPE_REQUEST))
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break;
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/* fall-through */
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case DevFmtUShort:
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device->FmtType = DevFmtShort;
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break;
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case DevFmtUInt:
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device->FmtType = DevFmtInt;
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break;
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case DevFmtUByte:
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case DevFmtShort:
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case DevFmtInt:
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break;
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}
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WAVEFORMATEXTENSIBLE OutputType{};
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DWORD speakers;
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HRESULT hr{self->mDS->GetSpeakerConfig(&speakers)};
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if(SUCCEEDED(hr))
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{
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speakers = DSSPEAKER_CONFIG(speakers);
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if(!(device->Flags&DEVICE_CHANNELS_REQUEST))
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{
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if(speakers == DSSPEAKER_MONO)
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device->FmtChans = DevFmtMono;
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else if(speakers == DSSPEAKER_STEREO || speakers == DSSPEAKER_HEADPHONE)
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device->FmtChans = DevFmtStereo;
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else if(speakers == DSSPEAKER_QUAD)
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device->FmtChans = DevFmtQuad;
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else if(speakers == DSSPEAKER_5POINT1_SURROUND)
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device->FmtChans = DevFmtX51;
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else if(speakers == DSSPEAKER_5POINT1_BACK)
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device->FmtChans = DevFmtX51Rear;
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else if(speakers == DSSPEAKER_7POINT1 || speakers == DSSPEAKER_7POINT1_SURROUND)
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device->FmtChans = DevFmtX71;
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else
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ERR("Unknown system speaker config: 0x%lx\n", speakers);
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}
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device->IsHeadphones = (device->FmtChans == DevFmtStereo &&
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speakers == DSSPEAKER_HEADPHONE);
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switch(device->FmtChans)
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{
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case DevFmtMono:
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OutputType.dwChannelMask = SPEAKER_FRONT_CENTER;
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break;
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case DevFmtAmbi3D:
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device->FmtChans = DevFmtStereo;
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/*fall-through*/
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case DevFmtStereo:
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OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
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SPEAKER_FRONT_RIGHT;
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break;
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case DevFmtQuad:
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OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
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SPEAKER_FRONT_RIGHT |
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SPEAKER_BACK_LEFT |
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SPEAKER_BACK_RIGHT;
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break;
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case DevFmtX51:
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OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
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SPEAKER_FRONT_RIGHT |
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SPEAKER_FRONT_CENTER |
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SPEAKER_LOW_FREQUENCY |
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SPEAKER_SIDE_LEFT |
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SPEAKER_SIDE_RIGHT;
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break;
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case DevFmtX51Rear:
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OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
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SPEAKER_FRONT_RIGHT |
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SPEAKER_FRONT_CENTER |
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SPEAKER_LOW_FREQUENCY |
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SPEAKER_BACK_LEFT |
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SPEAKER_BACK_RIGHT;
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break;
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case DevFmtX61:
|
|
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT |
|
|
SPEAKER_FRONT_CENTER |
|
|
SPEAKER_LOW_FREQUENCY |
|
|
SPEAKER_BACK_CENTER |
|
|
SPEAKER_SIDE_LEFT |
|
|
SPEAKER_SIDE_RIGHT;
|
|
break;
|
|
case DevFmtX71:
|
|
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT |
|
|
SPEAKER_FRONT_CENTER |
|
|
SPEAKER_LOW_FREQUENCY |
|
|
SPEAKER_BACK_LEFT |
|
|
SPEAKER_BACK_RIGHT |
|
|
SPEAKER_SIDE_LEFT |
|
|
SPEAKER_SIDE_RIGHT;
|
|
break;
|
|
}
|
|
|
|
retry_open:
|
|
hr = S_OK;
|
|
OutputType.Format.wFormatTag = WAVE_FORMAT_PCM;
|
|
OutputType.Format.nChannels = device->channelsFromFmt();
|
|
OutputType.Format.wBitsPerSample = device->bytesFromFmt() * 8;
|
|
OutputType.Format.nBlockAlign = OutputType.Format.nChannels*OutputType.Format.wBitsPerSample/8;
|
|
OutputType.Format.nSamplesPerSec = device->Frequency;
|
|
OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec*OutputType.Format.nBlockAlign;
|
|
OutputType.Format.cbSize = 0;
|
|
}
|
|
|
|
if(OutputType.Format.nChannels > 2 || device->FmtType == DevFmtFloat)
|
|
{
|
|
OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
|
|
OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample;
|
|
OutputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
|
|
if(device->FmtType == DevFmtFloat)
|
|
OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
|
|
else
|
|
OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
|
|
|
|
if(self->mPrimaryBuffer)
|
|
self->mPrimaryBuffer->Release();
|
|
self->mPrimaryBuffer = nullptr;
|
|
}
|
|
else
|
|
{
|
|
if(SUCCEEDED(hr) && !self->mPrimaryBuffer)
|
|
{
|
|
DSBUFFERDESC DSBDescription{};
|
|
DSBDescription.dwSize = sizeof(DSBDescription);
|
|
DSBDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
|
|
hr = self->mDS->CreateSoundBuffer(&DSBDescription, &self->mPrimaryBuffer, nullptr);
|
|
}
|
|
if(SUCCEEDED(hr))
|
|
hr = self->mPrimaryBuffer->SetFormat(&OutputType.Format);
|
|
}
|
|
|
|
if(SUCCEEDED(hr))
|
|
{
|
|
if(device->NumUpdates > MAX_UPDATES)
|
|
{
|
|
device->UpdateSize = (device->UpdateSize*device->NumUpdates +
|
|
MAX_UPDATES-1) / MAX_UPDATES;
|
|
device->NumUpdates = MAX_UPDATES;
|
|
}
|
|
|
|
DSBUFFERDESC DSBDescription{};
|
|
DSBDescription.dwSize = sizeof(DSBDescription);
|
|
DSBDescription.dwFlags = DSBCAPS_CTRLPOSITIONNOTIFY | DSBCAPS_GETCURRENTPOSITION2 |
|
|
DSBCAPS_GLOBALFOCUS;
|
|
DSBDescription.dwBufferBytes = device->UpdateSize * device->NumUpdates *
|
|
OutputType.Format.nBlockAlign;
|
|
DSBDescription.lpwfxFormat = &OutputType.Format;
|
|
|
|
hr = self->mDS->CreateSoundBuffer(&DSBDescription, &self->mBuffer, nullptr);
|
|
if(FAILED(hr) && device->FmtType == DevFmtFloat)
|
|
{
|
|
device->FmtType = DevFmtShort;
|
|
goto retry_open;
|
|
}
|
|
}
|
|
|
|
if(SUCCEEDED(hr))
|
|
{
|
|
void *ptr;
|
|
hr = self->mBuffer->QueryInterface(IID_IDirectSoundNotify, &ptr);
|
|
if(SUCCEEDED(hr))
|
|
{
|
|
auto Notifies = static_cast<IDirectSoundNotify*>(ptr);
|
|
self->mNotifies = Notifies;
|
|
|
|
device->NumUpdates = minu(device->NumUpdates, MAX_UPDATES);
|
|
|
|
std::array<DSBPOSITIONNOTIFY,MAX_UPDATES> nots;
|
|
for(ALuint i{0};i < device->NumUpdates;++i)
|
|
{
|
|
nots[i].dwOffset = i * device->UpdateSize * OutputType.Format.nBlockAlign;
|
|
nots[i].hEventNotify = self->mNotifyEvent;
|
|
}
|
|
if(Notifies->SetNotificationPositions(device->NumUpdates, nots.data()) != DS_OK)
|
|
hr = E_FAIL;
|
|
}
|
|
}
|
|
|
|
if(FAILED(hr))
|
|
{
|
|
if(self->mNotifies)
|
|
self->mNotifies->Release();
|
|
self->mNotifies = nullptr;
|
|
if(self->mBuffer)
|
|
self->mBuffer->Release();
|
|
self->mBuffer = nullptr;
|
|
if(self->mPrimaryBuffer)
|
|
self->mPrimaryBuffer->Release();
|
|
self->mPrimaryBuffer = nullptr;
|
|
return ALC_FALSE;
|
|
}
|
|
|
|
ResetEvent(self->mNotifyEvent);
|
|
SetDefaultWFXChannelOrder(device);
|
|
|
|
return ALC_TRUE;
|
|
}
|
|
|
|
ALCboolean ALCdsoundPlayback_start(ALCdsoundPlayback *self)
|
|
{
|
|
try {
|
|
self->mKillNow.store(AL_FALSE, std::memory_order_release);
|
|
self->mThread = std::thread(std::mem_fn(&ALCdsoundPlayback::mixerProc), self);
|
|
return ALC_TRUE;
|
|
}
|
|
catch(std::exception& e) {
|
|
ERR("Failed to start mixing thread: %s\n", e.what());
|
|
}
|
|
catch(...) {
|
|
}
|
|
return ALC_FALSE;
|
|
}
|
|
|
|
void ALCdsoundPlayback_stop(ALCdsoundPlayback *self)
|
|
{
|
|
if(self->mKillNow.exchange(AL_TRUE, std::memory_order_acq_rel) || !self->mThread.joinable())
|
|
return;
|
|
|
|
self->mThread.join();
|
|
|
|
self->mBuffer->Stop();
|
|
}
|
|
|
|
|
|
struct ALCdsoundCapture final : public ALCbackend {
|
|
ALCdsoundCapture(ALCdevice *device) noexcept : ALCbackend{device} { }
|
|
~ALCdsoundCapture() override;
|
|
|
|
IDirectSoundCapture *mDSC{nullptr};
|
|
IDirectSoundCaptureBuffer *mDSCbuffer{nullptr};
|
|
DWORD mBufferBytes{0u};
|
|
DWORD mCursor{0u};
|
|
|
|
RingBufferPtr mRing;
|
|
};
|
|
|
|
void ALCdsoundCapture_Construct(ALCdsoundCapture *self, ALCdevice *device);
|
|
void ALCdsoundCapture_Destruct(ALCdsoundCapture *self);
|
|
ALCenum ALCdsoundCapture_open(ALCdsoundCapture *self, const ALCchar *name);
|
|
DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, ALCboolean, reset)
|
|
ALCboolean ALCdsoundCapture_start(ALCdsoundCapture *self);
|
|
void ALCdsoundCapture_stop(ALCdsoundCapture *self);
|
|
ALCenum ALCdsoundCapture_captureSamples(ALCdsoundCapture *self, ALCvoid *buffer, ALCuint samples);
|
|
ALCuint ALCdsoundCapture_availableSamples(ALCdsoundCapture *self);
|
|
DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, ClockLatency, getClockLatency)
|
|
DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, void, lock)
|
|
DECLARE_FORWARD(ALCdsoundCapture, ALCbackend, void, unlock)
|
|
DECLARE_DEFAULT_ALLOCATORS(ALCdsoundCapture)
|
|
DEFINE_ALCBACKEND_VTABLE(ALCdsoundCapture);
|
|
|
|
void ALCdsoundCapture_Construct(ALCdsoundCapture *self, ALCdevice *device)
|
|
{
|
|
new (self) ALCdsoundCapture{device};
|
|
SET_VTABLE2(ALCdsoundCapture, ALCbackend, self);
|
|
}
|
|
|
|
void ALCdsoundCapture_Destruct(ALCdsoundCapture *self)
|
|
{ self->~ALCdsoundCapture(); }
|
|
|
|
ALCdsoundCapture::~ALCdsoundCapture()
|
|
{
|
|
if(mDSCbuffer)
|
|
{
|
|
mDSCbuffer->Stop();
|
|
mDSCbuffer->Release();
|
|
mDSCbuffer = nullptr;
|
|
}
|
|
|
|
if(mDSC)
|
|
mDSC->Release();
|
|
mDSC = nullptr;
|
|
}
|
|
|
|
|
|
ALCenum ALCdsoundCapture_open(ALCdsoundCapture *self, const ALCchar *deviceName)
|
|
{
|
|
ALCdevice *device{self->mDevice};
|
|
|
|
HRESULT hr;
|
|
if(CaptureDevices.empty())
|
|
{
|
|
/* Initialize COM to prevent name truncation */
|
|
HRESULT hrcom{CoInitialize(nullptr)};
|
|
hr = DirectSoundCaptureEnumerateW(DSoundEnumDevices, &CaptureDevices);
|
|
if(FAILED(hr))
|
|
ERR("Error enumerating DirectSound devices (0x%lx)!\n", hr);
|
|
if(SUCCEEDED(hrcom))
|
|
CoUninitialize();
|
|
}
|
|
|
|
const GUID *guid{nullptr};
|
|
if(!deviceName && !CaptureDevices.empty())
|
|
{
|
|
deviceName = CaptureDevices[0].name.c_str();
|
|
guid = &CaptureDevices[0].guid;
|
|
}
|
|
else
|
|
{
|
|
auto iter = std::find_if(CaptureDevices.cbegin(), CaptureDevices.cend(),
|
|
[deviceName](const DevMap &entry) -> bool
|
|
{ return entry.name == deviceName; }
|
|
);
|
|
if(iter == CaptureDevices.cend())
|
|
return ALC_INVALID_VALUE;
|
|
guid = &iter->guid;
|
|
}
|
|
|
|
switch(device->FmtType)
|
|
{
|
|
case DevFmtByte:
|
|
case DevFmtUShort:
|
|
case DevFmtUInt:
|
|
WARN("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
|
|
return ALC_INVALID_ENUM;
|
|
|
|
case DevFmtUByte:
|
|
case DevFmtShort:
|
|
case DevFmtInt:
|
|
case DevFmtFloat:
|
|
break;
|
|
}
|
|
|
|
WAVEFORMATEXTENSIBLE InputType{};
|
|
switch(device->FmtChans)
|
|
{
|
|
case DevFmtMono:
|
|
InputType.dwChannelMask = SPEAKER_FRONT_CENTER;
|
|
break;
|
|
case DevFmtStereo:
|
|
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT;
|
|
break;
|
|
case DevFmtQuad:
|
|
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT |
|
|
SPEAKER_BACK_LEFT |
|
|
SPEAKER_BACK_RIGHT;
|
|
break;
|
|
case DevFmtX51:
|
|
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT |
|
|
SPEAKER_FRONT_CENTER |
|
|
SPEAKER_LOW_FREQUENCY |
|
|
SPEAKER_SIDE_LEFT |
|
|
SPEAKER_SIDE_RIGHT;
|
|
break;
|
|
case DevFmtX51Rear:
|
|
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT |
|
|
SPEAKER_FRONT_CENTER |
|
|
SPEAKER_LOW_FREQUENCY |
|
|
SPEAKER_BACK_LEFT |
|
|
SPEAKER_BACK_RIGHT;
|
|
break;
|
|
case DevFmtX61:
|
|
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT |
|
|
SPEAKER_FRONT_CENTER |
|
|
SPEAKER_LOW_FREQUENCY |
|
|
SPEAKER_BACK_CENTER |
|
|
SPEAKER_SIDE_LEFT |
|
|
SPEAKER_SIDE_RIGHT;
|
|
break;
|
|
case DevFmtX71:
|
|
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
|
|
SPEAKER_FRONT_RIGHT |
|
|
SPEAKER_FRONT_CENTER |
|
|
SPEAKER_LOW_FREQUENCY |
|
|
SPEAKER_BACK_LEFT |
|
|
SPEAKER_BACK_RIGHT |
|
|
SPEAKER_SIDE_LEFT |
|
|
SPEAKER_SIDE_RIGHT;
|
|
break;
|
|
case DevFmtAmbi3D:
|
|
WARN("%s capture not supported\n", DevFmtChannelsString(device->FmtChans));
|
|
return ALC_INVALID_ENUM;
|
|
}
|
|
|
|
InputType.Format.wFormatTag = WAVE_FORMAT_PCM;
|
|
InputType.Format.nChannels = device->channelsFromFmt();
|
|
InputType.Format.wBitsPerSample = device->bytesFromFmt() * 8;
|
|
InputType.Format.nBlockAlign = InputType.Format.nChannels*InputType.Format.wBitsPerSample/8;
|
|
InputType.Format.nSamplesPerSec = device->Frequency;
|
|
InputType.Format.nAvgBytesPerSec = InputType.Format.nSamplesPerSec*InputType.Format.nBlockAlign;
|
|
InputType.Format.cbSize = 0;
|
|
InputType.Samples.wValidBitsPerSample = InputType.Format.wBitsPerSample;
|
|
if(device->FmtType == DevFmtFloat)
|
|
InputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
|
|
else
|
|
InputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
|
|
|
|
if(InputType.Format.nChannels > 2 || device->FmtType == DevFmtFloat)
|
|
{
|
|
InputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
|
|
InputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
|
|
}
|
|
|
|
ALuint samples{device->UpdateSize * device->NumUpdates};
|
|
samples = maxu(samples, 100 * device->Frequency / 1000);
|
|
|
|
DSCBUFFERDESC DSCBDescription{};
|
|
DSCBDescription.dwSize = sizeof(DSCBDescription);
|
|
DSCBDescription.dwFlags = 0;
|
|
DSCBDescription.dwBufferBytes = samples * InputType.Format.nBlockAlign;
|
|
DSCBDescription.lpwfxFormat = &InputType.Format;
|
|
|
|
//DirectSoundCapture Init code
|
|
hr = DirectSoundCaptureCreate(guid, &self->mDSC, nullptr);
|
|
if(SUCCEEDED(hr))
|
|
self->mDSC->CreateCaptureBuffer(&DSCBDescription, &self->mDSCbuffer, nullptr);
|
|
if(SUCCEEDED(hr))
|
|
{
|
|
self->mRing = CreateRingBuffer(device->UpdateSize*device->NumUpdates,
|
|
InputType.Format.nBlockAlign, false);
|
|
if(!self->mRing) hr = DSERR_OUTOFMEMORY;
|
|
}
|
|
|
|
if(FAILED(hr))
|
|
{
|
|
ERR("Device init failed: 0x%08lx\n", hr);
|
|
|
|
self->mRing = nullptr;
|
|
if(self->mDSCbuffer)
|
|
self->mDSCbuffer->Release();
|
|
self->mDSCbuffer = nullptr;
|
|
if(self->mDSC)
|
|
self->mDSC->Release();
|
|
self->mDSC = nullptr;
|
|
|
|
return ALC_INVALID_VALUE;
|
|
}
|
|
|
|
self->mBufferBytes = DSCBDescription.dwBufferBytes;
|
|
SetDefaultWFXChannelOrder(device);
|
|
|
|
device->DeviceName = deviceName;
|
|
return ALC_NO_ERROR;
|
|
}
|
|
|
|
ALCboolean ALCdsoundCapture_start(ALCdsoundCapture *self)
|
|
{
|
|
HRESULT hr{self->mDSCbuffer->Start(DSCBSTART_LOOPING)};
|
|
if(FAILED(hr))
|
|
{
|
|
ERR("start failed: 0x%08lx\n", hr);
|
|
aluHandleDisconnect(self->mDevice, "Failure starting capture: 0x%lx", hr);
|
|
return ALC_FALSE;
|
|
}
|
|
|
|
return ALC_TRUE;
|
|
}
|
|
|
|
void ALCdsoundCapture_stop(ALCdsoundCapture *self)
|
|
{
|
|
HRESULT hr{self->mDSCbuffer->Stop()};
|
|
if(FAILED(hr))
|
|
{
|
|
ERR("stop failed: 0x%08lx\n", hr);
|
|
aluHandleDisconnect(self->mDevice, "Failure stopping capture: 0x%lx", hr);
|
|
}
|
|
}
|
|
|
|
ALCenum ALCdsoundCapture_captureSamples(ALCdsoundCapture *self, ALCvoid *buffer, ALCuint samples)
|
|
{
|
|
RingBuffer *ring{self->mRing.get()};
|
|
ring->read(buffer, samples);
|
|
return ALC_NO_ERROR;
|
|
}
|
|
|
|
ALCuint ALCdsoundCapture_availableSamples(ALCdsoundCapture *self)
|
|
{
|
|
ALCdevice *device{self->mDevice};
|
|
RingBuffer *ring{self->mRing.get()};
|
|
|
|
if(!device->Connected.load(std::memory_order_acquire))
|
|
return static_cast<ALCuint>(ring->readSpace());
|
|
|
|
ALsizei FrameSize{device->frameSizeFromFmt()};
|
|
DWORD BufferBytes{self->mBufferBytes};
|
|
DWORD LastCursor{self->mCursor};
|
|
|
|
DWORD ReadCursor;
|
|
void *ReadPtr1, *ReadPtr2;
|
|
DWORD ReadCnt1, ReadCnt2;
|
|
HRESULT hr{self->mDSCbuffer->GetCurrentPosition(nullptr, &ReadCursor)};
|
|
if(SUCCEEDED(hr))
|
|
{
|
|
DWORD NumBytes{(ReadCursor-LastCursor + BufferBytes) % BufferBytes};
|
|
if(!NumBytes) return static_cast<ALCubyte>(ring->readSpace());
|
|
hr = self->mDSCbuffer->Lock(LastCursor, NumBytes, &ReadPtr1, &ReadCnt1,
|
|
&ReadPtr2, &ReadCnt2, 0);
|
|
}
|
|
if(SUCCEEDED(hr))
|
|
{
|
|
ring->write(ReadPtr1, ReadCnt1/FrameSize);
|
|
if(ReadPtr2 != nullptr && ReadCnt2 > 0)
|
|
ring->write(ReadPtr2, ReadCnt2/FrameSize);
|
|
hr = self->mDSCbuffer->Unlock(ReadPtr1, ReadCnt1, ReadPtr2, ReadCnt2);
|
|
self->mCursor = (LastCursor+ReadCnt1+ReadCnt2) % BufferBytes;
|
|
}
|
|
|
|
if(FAILED(hr))
|
|
{
|
|
ERR("update failed: 0x%08lx\n", hr);
|
|
aluHandleDisconnect(device, "Failure retrieving capture data: 0x%lx", hr);
|
|
}
|
|
|
|
return static_cast<ALCuint>(ring->readSpace());
|
|
}
|
|
|
|
} // namespace
|
|
|
|
|
|
BackendFactory &DSoundBackendFactory::getFactory()
|
|
{
|
|
static DSoundBackendFactory factory{};
|
|
return factory;
|
|
}
|
|
|
|
bool DSoundBackendFactory::init()
|
|
{ return DSoundLoad(); }
|
|
|
|
void DSoundBackendFactory::deinit()
|
|
{
|
|
PlaybackDevices.clear();
|
|
CaptureDevices.clear();
|
|
|
|
#ifdef HAVE_DYNLOAD
|
|
if(ds_handle)
|
|
CloseLib(ds_handle);
|
|
ds_handle = nullptr;
|
|
#endif
|
|
}
|
|
|
|
bool DSoundBackendFactory::querySupport(ALCbackend_Type type)
|
|
{ return (type == ALCbackend_Playback || type == ALCbackend_Capture); }
|
|
|
|
void DSoundBackendFactory::probe(DevProbe type, std::string *outnames)
|
|
{
|
|
auto add_device = [outnames](const DevMap &entry) -> void
|
|
{
|
|
/* +1 to also append the null char (to ensure a null-separated list and
|
|
* double-null terminated list).
|
|
*/
|
|
outnames->append(entry.name.c_str(), entry.name.length()+1);
|
|
};
|
|
|
|
/* Initialize COM to prevent name truncation */
|
|
HRESULT hr;
|
|
HRESULT hrcom{CoInitialize(nullptr)};
|
|
switch(type)
|
|
{
|
|
case ALL_DEVICE_PROBE:
|
|
PlaybackDevices.clear();
|
|
hr = DirectSoundEnumerateW(DSoundEnumDevices, &PlaybackDevices);
|
|
if(FAILED(hr))
|
|
ERR("Error enumerating DirectSound playback devices (0x%lx)!\n", hr);
|
|
std::for_each(PlaybackDevices.cbegin(), PlaybackDevices.cend(), add_device);
|
|
break;
|
|
|
|
case CAPTURE_DEVICE_PROBE:
|
|
CaptureDevices.clear();
|
|
hr = DirectSoundCaptureEnumerateW(DSoundEnumDevices, &CaptureDevices);
|
|
if(FAILED(hr))
|
|
ERR("Error enumerating DirectSound capture devices (0x%lx)!\n", hr);
|
|
std::for_each(CaptureDevices.cbegin(), CaptureDevices.cend(), add_device);
|
|
break;
|
|
}
|
|
if(SUCCEEDED(hrcom))
|
|
CoUninitialize();
|
|
}
|
|
|
|
ALCbackend *DSoundBackendFactory::createBackend(ALCdevice *device, ALCbackend_Type type)
|
|
{
|
|
if(type == ALCbackend_Playback)
|
|
{
|
|
ALCdsoundPlayback *backend;
|
|
NEW_OBJ(backend, ALCdsoundPlayback)(device);
|
|
return backend;
|
|
}
|
|
|
|
if(type == ALCbackend_Capture)
|
|
{
|
|
ALCdsoundCapture *backend;
|
|
NEW_OBJ(backend, ALCdsoundCapture)(device);
|
|
return backend;
|
|
}
|
|
|
|
return nullptr;
|
|
}
|