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authorChris Robinson <[email protected]>2019-07-28 18:56:04 -0700
committerChris Robinson <[email protected]>2019-07-28 18:56:04 -0700
commitcb3e96e75640730b9391f0d2d922eecd9ee2ce79 (patch)
tree23520551bddb2a80354e44da47f54201fdc084f0 /alc/backends/dsound.cpp
parent93e60919c8f387c36c267ca9faa1ac653254aea6 (diff)
Rename Alc to alc
Diffstat (limited to 'alc/backends/dsound.cpp')
-rw-r--r--alc/backends/dsound.cpp938
1 files changed, 938 insertions, 0 deletions
diff --git a/alc/backends/dsound.cpp b/alc/backends/dsound.cpp
new file mode 100644
index 00000000..5a156d54
--- /dev/null
+++ b/alc/backends/dsound.cpp
@@ -0,0 +1,938 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include "backends/dsound.h"
+
+#define WIN32_LEAN_AND_MEAN
+#include <windows.h>
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <memory.h>
+
+#include <cguid.h>
+#include <mmreg.h>
+#ifndef _WAVEFORMATEXTENSIBLE_
+#include <ks.h>
+#include <ksmedia.h>
+#endif
+
+#include <atomic>
+#include <cassert>
+#include <thread>
+#include <string>
+#include <vector>
+#include <algorithm>
+#include <functional>
+
+#include "alcmain.h"
+#include "alu.h"
+#include "ringbuffer.h"
+#include "compat.h"
+#include "threads.h"
+
+/* MinGW-w64 needs this for some unknown reason now. */
+using LPCWAVEFORMATEX = const WAVEFORMATEX*;
+#include <dsound.h>
+
+
+#ifndef DSSPEAKER_5POINT1
+# define DSSPEAKER_5POINT1 0x00000006
+#endif
+#ifndef DSSPEAKER_5POINT1_BACK
+# define DSSPEAKER_5POINT1_BACK 0x00000006
+#endif
+#ifndef DSSPEAKER_7POINT1
+# define DSSPEAKER_7POINT1 0x00000007
+#endif
+#ifndef DSSPEAKER_7POINT1_SURROUND
+# define DSSPEAKER_7POINT1_SURROUND 0x00000008
+#endif
+#ifndef DSSPEAKER_5POINT1_SURROUND
+# define DSSPEAKER_5POINT1_SURROUND 0x00000009
+#endif
+
+
+/* Some headers seem to define these as macros for __uuidof, which is annoying
+ * since some headers don't declare them at all. Hopefully the ifdef is enough
+ * to tell if they need to be declared.
+ */
+#ifndef KSDATAFORMAT_SUBTYPE_PCM
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
+#endif
+#ifndef KSDATAFORMAT_SUBTYPE_IEEE_FLOAT
+DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
+#endif
+
+namespace {
+
+#define DEVNAME_HEAD "OpenAL Soft on "
+
+
+#ifdef HAVE_DYNLOAD
+void *ds_handle;
+HRESULT (WINAPI *pDirectSoundCreate)(const GUID *pcGuidDevice, IDirectSound **ppDS, IUnknown *pUnkOuter);
+HRESULT (WINAPI *pDirectSoundEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext);
+HRESULT (WINAPI *pDirectSoundCaptureCreate)(const GUID *pcGuidDevice, IDirectSoundCapture **ppDSC, IUnknown *pUnkOuter);
+HRESULT (WINAPI *pDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext);
+
+#ifndef IN_IDE_PARSER
+#define DirectSoundCreate pDirectSoundCreate
+#define DirectSoundEnumerateW pDirectSoundEnumerateW
+#define DirectSoundCaptureCreate pDirectSoundCaptureCreate
+#define DirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW
+#endif
+#endif
+
+
+#define MAX_UPDATES 128
+
+struct DevMap {
+ std::string name;
+ GUID guid;
+
+ template<typename T0, typename T1>
+ DevMap(T0&& name_, T1&& guid_)
+ : name{std::forward<T0>(name_)}, guid{std::forward<T1>(guid_)}
+ { }
+};
+
+al::vector<DevMap> PlaybackDevices;
+al::vector<DevMap> CaptureDevices;
+
+bool checkName(const al::vector<DevMap> &list, const std::string &name)
+{
+ return std::find_if(list.cbegin(), list.cend(),
+ [&name](const DevMap &entry) -> bool
+ { return entry.name == name; }
+ ) != list.cend();
+}
+
+BOOL CALLBACK DSoundEnumDevices(GUID *guid, const WCHAR *desc, const WCHAR*, void *data)
+{
+ if(!guid)
+ return TRUE;
+
+ auto& devices = *static_cast<al::vector<DevMap>*>(data);
+ const std::string basename{DEVNAME_HEAD + wstr_to_utf8(desc)};
+
+ int count{1};
+ std::string newname{basename};
+ while(checkName(devices, newname))
+ {
+ newname = basename;
+ newname += " #";
+ newname += std::to_string(++count);
+ }
+ devices.emplace_back(std::move(newname), *guid);
+ const DevMap &newentry = devices.back();
+
+ OLECHAR *guidstr{nullptr};
+ HRESULT hr{StringFromCLSID(*guid, &guidstr)};
+ if(SUCCEEDED(hr))
+ {
+ TRACE("Got device \"%s\", GUID \"%ls\"\n", newentry.name.c_str(), guidstr);
+ CoTaskMemFree(guidstr);
+ }
+
+ return TRUE;
+}
+
+
+struct DSoundPlayback final : public BackendBase {
+ DSoundPlayback(ALCdevice *device) noexcept : BackendBase{device} { }
+ ~DSoundPlayback() override;
+
+ int mixerProc();
+
+ ALCenum open(const ALCchar *name) override;
+ ALCboolean reset() override;
+ ALCboolean start() override;
+ void stop() override;
+
+ IDirectSound *mDS{nullptr};
+ IDirectSoundBuffer *mPrimaryBuffer{nullptr};
+ IDirectSoundBuffer *mBuffer{nullptr};
+ IDirectSoundNotify *mNotifies{nullptr};
+ HANDLE mNotifyEvent{nullptr};
+
+ std::atomic<bool> mKillNow{true};
+ std::thread mThread;
+
+ DEF_NEWDEL(DSoundPlayback)
+};
+
+DSoundPlayback::~DSoundPlayback()
+{
+ if(mNotifies)
+ mNotifies->Release();
+ mNotifies = nullptr;
+ if(mBuffer)
+ mBuffer->Release();
+ mBuffer = nullptr;
+ if(mPrimaryBuffer)
+ mPrimaryBuffer->Release();
+ mPrimaryBuffer = nullptr;
+
+ if(mDS)
+ mDS->Release();
+ mDS = nullptr;
+ if(mNotifyEvent)
+ CloseHandle(mNotifyEvent);
+ mNotifyEvent = nullptr;
+}
+
+
+FORCE_ALIGN int DSoundPlayback::mixerProc()
+{
+ SetRTPriority();
+ althrd_setname(MIXER_THREAD_NAME);
+
+ DSBCAPS DSBCaps{};
+ DSBCaps.dwSize = sizeof(DSBCaps);
+ HRESULT err{mBuffer->GetCaps(&DSBCaps)};
+ if(FAILED(err))
+ {
+ ERR("Failed to get buffer caps: 0x%lx\n", err);
+ aluHandleDisconnect(mDevice, "Failure retrieving playback buffer info: 0x%lx", err);
+ return 1;
+ }
+
+ ALsizei FrameSize{mDevice->frameSizeFromFmt()};
+ DWORD FragSize{mDevice->UpdateSize * FrameSize};
+
+ bool Playing{false};
+ DWORD LastCursor{0u};
+ mBuffer->GetCurrentPosition(&LastCursor, nullptr);
+ while(!mKillNow.load(std::memory_order_acquire) &&
+ mDevice->Connected.load(std::memory_order_acquire))
+ {
+ // Get current play cursor
+ DWORD PlayCursor;
+ mBuffer->GetCurrentPosition(&PlayCursor, nullptr);
+ DWORD avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes;
+
+ if(avail < FragSize)
+ {
+ if(!Playing)
+ {
+ err = mBuffer->Play(0, 0, DSBPLAY_LOOPING);
+ if(FAILED(err))
+ {
+ ERR("Failed to play buffer: 0x%lx\n", err);
+ aluHandleDisconnect(mDevice, "Failure starting playback: 0x%lx", err);
+ return 1;
+ }
+ Playing = true;
+ }
+
+ avail = WaitForSingleObjectEx(mNotifyEvent, 2000, FALSE);
+ if(avail != WAIT_OBJECT_0)
+ ERR("WaitForSingleObjectEx error: 0x%lx\n", avail);
+ continue;
+ }
+ avail -= avail%FragSize;
+
+ // Lock output buffer
+ void *WritePtr1, *WritePtr2;
+ DWORD WriteCnt1{0u}, WriteCnt2{0u};
+ err = mBuffer->Lock(LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
+
+ // If the buffer is lost, restore it and lock
+ if(err == DSERR_BUFFERLOST)
+ {
+ WARN("Buffer lost, restoring...\n");
+ err = mBuffer->Restore();
+ if(SUCCEEDED(err))
+ {
+ Playing = false;
+ LastCursor = 0;
+ err = mBuffer->Lock(0, DSBCaps.dwBufferBytes, &WritePtr1, &WriteCnt1,
+ &WritePtr2, &WriteCnt2, 0);
+ }
+ }
+
+ if(SUCCEEDED(err))
+ {
+ lock();
+ aluMixData(mDevice, WritePtr1, WriteCnt1/FrameSize);
+ if(WriteCnt2 > 0)
+ aluMixData(mDevice, WritePtr2, WriteCnt2/FrameSize);
+ unlock();
+
+ mBuffer->Unlock(WritePtr1, WriteCnt1, WritePtr2, WriteCnt2);
+ }
+ else
+ {
+ ERR("Buffer lock error: %#lx\n", err);
+ aluHandleDisconnect(mDevice, "Failed to lock output buffer: 0x%lx", err);
+ return 1;
+ }
+
+ // Update old write cursor location
+ LastCursor += WriteCnt1+WriteCnt2;
+ LastCursor %= DSBCaps.dwBufferBytes;
+ }
+
+ return 0;
+}
+
+ALCenum DSoundPlayback::open(const ALCchar *name)
+{
+ HRESULT hr;
+ if(PlaybackDevices.empty())
+ {
+ /* Initialize COM to prevent name truncation */
+ HRESULT hrcom{CoInitialize(nullptr)};
+ hr = DirectSoundEnumerateW(DSoundEnumDevices, &PlaybackDevices);
+ if(FAILED(hr))
+ ERR("Error enumerating DirectSound devices (0x%lx)!\n", hr);
+ if(SUCCEEDED(hrcom))
+ CoUninitialize();
+ }
+
+ const GUID *guid{nullptr};
+ if(!name && !PlaybackDevices.empty())
+ {
+ name = PlaybackDevices[0].name.c_str();
+ guid = &PlaybackDevices[0].guid;
+ }
+ else
+ {
+ auto iter = std::find_if(PlaybackDevices.cbegin(), PlaybackDevices.cend(),
+ [name](const DevMap &entry) -> bool
+ { return entry.name == name; }
+ );
+ if(iter == PlaybackDevices.cend())
+ return ALC_INVALID_VALUE;
+ guid = &iter->guid;
+ }
+
+ hr = DS_OK;
+ mNotifyEvent = CreateEventW(nullptr, FALSE, FALSE, nullptr);
+ if(!mNotifyEvent) hr = E_FAIL;
+
+ //DirectSound Init code
+ if(SUCCEEDED(hr))
+ hr = DirectSoundCreate(guid, &mDS, nullptr);
+ if(SUCCEEDED(hr))
+ hr = mDS->SetCooperativeLevel(GetForegroundWindow(), DSSCL_PRIORITY);
+ if(FAILED(hr))
+ {
+ ERR("Device init failed: 0x%08lx\n", hr);
+ return ALC_INVALID_VALUE;
+ }
+
+ mDevice->DeviceName = name;
+ return ALC_NO_ERROR;
+}
+
+ALCboolean DSoundPlayback::reset()
+{
+ if(mNotifies)
+ mNotifies->Release();
+ mNotifies = nullptr;
+ if(mBuffer)
+ mBuffer->Release();
+ mBuffer = nullptr;
+ if(mPrimaryBuffer)
+ mPrimaryBuffer->Release();
+ mPrimaryBuffer = nullptr;
+
+ switch(mDevice->FmtType)
+ {
+ case DevFmtByte:
+ mDevice->FmtType = DevFmtUByte;
+ break;
+ case DevFmtFloat:
+ if(mDevice->Flags.get<SampleTypeRequest>())
+ break;
+ /* fall-through */
+ case DevFmtUShort:
+ mDevice->FmtType = DevFmtShort;
+ break;
+ case DevFmtUInt:
+ mDevice->FmtType = DevFmtInt;
+ break;
+ case DevFmtUByte:
+ case DevFmtShort:
+ case DevFmtInt:
+ break;
+ }
+
+ WAVEFORMATEXTENSIBLE OutputType{};
+ DWORD speakers;
+ HRESULT hr{mDS->GetSpeakerConfig(&speakers)};
+ if(SUCCEEDED(hr))
+ {
+ speakers = DSSPEAKER_CONFIG(speakers);
+ if(!mDevice->Flags.get<ChannelsRequest>())
+ {
+ if(speakers == DSSPEAKER_MONO)
+ mDevice->FmtChans = DevFmtMono;
+ else if(speakers == DSSPEAKER_STEREO || speakers == DSSPEAKER_HEADPHONE)
+ mDevice->FmtChans = DevFmtStereo;
+ else if(speakers == DSSPEAKER_QUAD)
+ mDevice->FmtChans = DevFmtQuad;
+ else if(speakers == DSSPEAKER_5POINT1_SURROUND)
+ mDevice->FmtChans = DevFmtX51;
+ else if(speakers == DSSPEAKER_5POINT1_BACK)
+ mDevice->FmtChans = DevFmtX51Rear;
+ else if(speakers == DSSPEAKER_7POINT1 || speakers == DSSPEAKER_7POINT1_SURROUND)
+ mDevice->FmtChans = DevFmtX71;
+ else
+ ERR("Unknown system speaker config: 0x%lx\n", speakers);
+ }
+ mDevice->IsHeadphones = (mDevice->FmtChans == DevFmtStereo &&
+ speakers == DSSPEAKER_HEADPHONE);
+
+ switch(mDevice->FmtChans)
+ {
+ case DevFmtMono:
+ OutputType.dwChannelMask = SPEAKER_FRONT_CENTER;
+ break;
+ case DevFmtAmbi3D:
+ mDevice->FmtChans = DevFmtStereo;
+ /*fall-through*/
+ case DevFmtStereo:
+ OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
+ SPEAKER_FRONT_RIGHT;
+ break;
+ case DevFmtQuad:
+ OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
+ SPEAKER_FRONT_RIGHT |
+ SPEAKER_BACK_LEFT |
+ SPEAKER_BACK_RIGHT;
+ break;
+ case DevFmtX51:
+ OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
+ SPEAKER_FRONT_RIGHT |
+ SPEAKER_FRONT_CENTER |
+ SPEAKER_LOW_FREQUENCY |
+ SPEAKER_SIDE_LEFT |
+ SPEAKER_SIDE_RIGHT;
+ break;
+ case DevFmtX51Rear:
+ OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
+ SPEAKER_FRONT_RIGHT |
+ SPEAKER_FRONT_CENTER |
+ SPEAKER_LOW_FREQUENCY |
+ SPEAKER_BACK_LEFT |
+ SPEAKER_BACK_RIGHT;
+ break;
+ 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 = mDevice->channelsFromFmt();
+ OutputType.Format.wBitsPerSample = mDevice->bytesFromFmt() * 8;
+ OutputType.Format.nBlockAlign = OutputType.Format.nChannels*OutputType.Format.wBitsPerSample/8;
+ OutputType.Format.nSamplesPerSec = mDevice->Frequency;
+ OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec*OutputType.Format.nBlockAlign;
+ OutputType.Format.cbSize = 0;
+ }
+
+ if(OutputType.Format.nChannels > 2 || mDevice->FmtType == DevFmtFloat)
+ {
+ OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+ OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample;
+ OutputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
+ if(mDevice->FmtType == DevFmtFloat)
+ OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
+ else
+ OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+
+ if(mPrimaryBuffer)
+ mPrimaryBuffer->Release();
+ mPrimaryBuffer = nullptr;
+ }
+ else
+ {
+ if(SUCCEEDED(hr) && !mPrimaryBuffer)
+ {
+ DSBUFFERDESC DSBDescription{};
+ DSBDescription.dwSize = sizeof(DSBDescription);
+ DSBDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+ hr = mDS->CreateSoundBuffer(&DSBDescription, &mPrimaryBuffer, nullptr);
+ }
+ if(SUCCEEDED(hr))
+ hr = mPrimaryBuffer->SetFormat(&OutputType.Format);
+ }
+
+ if(SUCCEEDED(hr))
+ {
+ ALuint num_updates{mDevice->BufferSize / mDevice->UpdateSize};
+ if(num_updates > MAX_UPDATES)
+ num_updates = MAX_UPDATES;
+ mDevice->BufferSize = mDevice->UpdateSize * num_updates;
+
+ DSBUFFERDESC DSBDescription{};
+ DSBDescription.dwSize = sizeof(DSBDescription);
+ DSBDescription.dwFlags = DSBCAPS_CTRLPOSITIONNOTIFY | DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_GLOBALFOCUS;
+ DSBDescription.dwBufferBytes = mDevice->BufferSize * OutputType.Format.nBlockAlign;
+ DSBDescription.lpwfxFormat = &OutputType.Format;
+
+ hr = mDS->CreateSoundBuffer(&DSBDescription, &mBuffer, nullptr);
+ if(FAILED(hr) && mDevice->FmtType == DevFmtFloat)
+ {
+ mDevice->FmtType = DevFmtShort;
+ goto retry_open;
+ }
+ }
+
+ if(SUCCEEDED(hr))
+ {
+ void *ptr;
+ hr = mBuffer->QueryInterface(IID_IDirectSoundNotify, &ptr);
+ if(SUCCEEDED(hr))
+ {
+ auto Notifies = static_cast<IDirectSoundNotify*>(ptr);
+ mNotifies = Notifies;
+
+ ALuint num_updates{mDevice->BufferSize / mDevice->UpdateSize};
+ assert(num_updates <= MAX_UPDATES);
+
+ std::array<DSBPOSITIONNOTIFY,MAX_UPDATES> nots;
+ for(ALuint i{0};i < num_updates;++i)
+ {
+ nots[i].dwOffset = i * mDevice->UpdateSize * OutputType.Format.nBlockAlign;
+ nots[i].hEventNotify = mNotifyEvent;
+ }
+ if(Notifies->SetNotificationPositions(num_updates, nots.data()) != DS_OK)
+ hr = E_FAIL;
+ }
+ }
+
+ if(FAILED(hr))
+ {
+ if(mNotifies)
+ mNotifies->Release();
+ mNotifies = nullptr;
+ if(mBuffer)
+ mBuffer->Release();
+ mBuffer = nullptr;
+ if(mPrimaryBuffer)
+ mPrimaryBuffer->Release();
+ mPrimaryBuffer = nullptr;
+ return ALC_FALSE;
+ }
+
+ ResetEvent(mNotifyEvent);
+ SetDefaultWFXChannelOrder(mDevice);
+
+ return ALC_TRUE;
+}
+
+ALCboolean DSoundPlayback::start()
+{
+ try {
+ mKillNow.store(false, std::memory_order_release);
+ mThread = std::thread{std::mem_fn(&DSoundPlayback::mixerProc), this};
+ return ALC_TRUE;
+ }
+ catch(std::exception& e) {
+ ERR("Failed to start mixing thread: %s\n", e.what());
+ }
+ catch(...) {
+ }
+ return ALC_FALSE;
+}
+
+void DSoundPlayback::stop()
+{
+ if(mKillNow.exchange(true, std::memory_order_acq_rel) || !mThread.joinable())
+ return;
+ mThread.join();
+
+ mBuffer->Stop();
+}
+
+
+struct DSoundCapture final : public BackendBase {
+ DSoundCapture(ALCdevice *device) noexcept : BackendBase{device} { }
+ ~DSoundCapture() override;
+
+ ALCenum open(const ALCchar *name) override;
+ ALCboolean start() override;
+ void stop() override;
+ ALCenum captureSamples(void *buffer, ALCuint samples) override;
+ ALCuint availableSamples() override;
+
+ IDirectSoundCapture *mDSC{nullptr};
+ IDirectSoundCaptureBuffer *mDSCbuffer{nullptr};
+ DWORD mBufferBytes{0u};
+ DWORD mCursor{0u};
+
+ RingBufferPtr mRing;
+
+ DEF_NEWDEL(DSoundCapture)
+};
+
+DSoundCapture::~DSoundCapture()
+{
+ if(mDSCbuffer)
+ {
+ mDSCbuffer->Stop();
+ mDSCbuffer->Release();
+ mDSCbuffer = nullptr;
+ }
+
+ if(mDSC)
+ mDSC->Release();
+ mDSC = nullptr;
+}
+
+
+ALCenum DSoundCapture::open(const ALCchar *name)
+{
+ 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(!name && !CaptureDevices.empty())
+ {
+ name = CaptureDevices[0].name.c_str();
+ guid = &CaptureDevices[0].guid;
+ }
+ else
+ {
+ auto iter = std::find_if(CaptureDevices.cbegin(), CaptureDevices.cend(),
+ [name](const DevMap &entry) -> bool
+ { return entry.name == name; }
+ );
+ if(iter == CaptureDevices.cend())
+ return ALC_INVALID_VALUE;
+ guid = &iter->guid;
+ }
+
+ switch(mDevice->FmtType)
+ {
+ case DevFmtByte:
+ case DevFmtUShort:
+ case DevFmtUInt:
+ WARN("%s capture samples not supported\n", DevFmtTypeString(mDevice->FmtType));
+ return ALC_INVALID_ENUM;
+
+ case DevFmtUByte:
+ case DevFmtShort:
+ case DevFmtInt:
+ case DevFmtFloat:
+ break;
+ }
+
+ WAVEFORMATEXTENSIBLE InputType{};
+ switch(mDevice->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(mDevice->FmtChans));
+ return ALC_INVALID_ENUM;
+ }
+
+ InputType.Format.wFormatTag = WAVE_FORMAT_PCM;
+ InputType.Format.nChannels = mDevice->channelsFromFmt();
+ InputType.Format.wBitsPerSample = mDevice->bytesFromFmt() * 8;
+ InputType.Format.nBlockAlign = InputType.Format.nChannels*InputType.Format.wBitsPerSample/8;
+ InputType.Format.nSamplesPerSec = mDevice->Frequency;
+ InputType.Format.nAvgBytesPerSec = InputType.Format.nSamplesPerSec*InputType.Format.nBlockAlign;
+ InputType.Format.cbSize = 0;
+ InputType.Samples.wValidBitsPerSample = InputType.Format.wBitsPerSample;
+ if(mDevice->FmtType == DevFmtFloat)
+ InputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
+ else
+ InputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+
+ if(InputType.Format.nChannels > 2 || mDevice->FmtType == DevFmtFloat)
+ {
+ InputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+ InputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
+ }
+
+ ALuint samples{mDevice->BufferSize};
+ samples = maxu(samples, 100 * mDevice->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, &mDSC, nullptr);
+ if(SUCCEEDED(hr))
+ mDSC->CreateCaptureBuffer(&DSCBDescription, &mDSCbuffer, nullptr);
+ if(SUCCEEDED(hr))
+ {
+ mRing = CreateRingBuffer(mDevice->BufferSize, InputType.Format.nBlockAlign, false);
+ if(!mRing) hr = DSERR_OUTOFMEMORY;
+ }
+
+ if(FAILED(hr))
+ {
+ ERR("Device init failed: 0x%08lx\n", hr);
+
+ mRing = nullptr;
+ if(mDSCbuffer)
+ mDSCbuffer->Release();
+ mDSCbuffer = nullptr;
+ if(mDSC)
+ mDSC->Release();
+ mDSC = nullptr;
+
+ return ALC_INVALID_VALUE;
+ }
+
+ mBufferBytes = DSCBDescription.dwBufferBytes;
+ SetDefaultWFXChannelOrder(mDevice);
+
+ mDevice->DeviceName = name;
+ return ALC_NO_ERROR;
+}
+
+ALCboolean DSoundCapture::start()
+{
+ HRESULT hr{mDSCbuffer->Start(DSCBSTART_LOOPING)};
+ if(FAILED(hr))
+ {
+ ERR("start failed: 0x%08lx\n", hr);
+ aluHandleDisconnect(mDevice, "Failure starting capture: 0x%lx", hr);
+ return ALC_FALSE;
+ }
+ return ALC_TRUE;
+}
+
+void DSoundCapture::stop()
+{
+ HRESULT hr{mDSCbuffer->Stop()};
+ if(FAILED(hr))
+ {
+ ERR("stop failed: 0x%08lx\n", hr);
+ aluHandleDisconnect(mDevice, "Failure stopping capture: 0x%lx", hr);
+ }
+}
+
+ALCenum DSoundCapture::captureSamples(void *buffer, ALCuint samples)
+{
+ mRing->read(buffer, samples);
+ return ALC_NO_ERROR;
+}
+
+ALCuint DSoundCapture::availableSamples()
+{
+ if(!mDevice->Connected.load(std::memory_order_acquire))
+ return static_cast<ALCuint>(mRing->readSpace());
+
+ ALsizei FrameSize{mDevice->frameSizeFromFmt()};
+ DWORD BufferBytes{mBufferBytes};
+ DWORD LastCursor{mCursor};
+
+ DWORD ReadCursor;
+ void *ReadPtr1, *ReadPtr2;
+ DWORD ReadCnt1, ReadCnt2;
+ HRESULT hr{mDSCbuffer->GetCurrentPosition(nullptr, &ReadCursor)};
+ if(SUCCEEDED(hr))
+ {
+ DWORD NumBytes{(ReadCursor-LastCursor + BufferBytes) % BufferBytes};
+ if(!NumBytes) return static_cast<ALCubyte>(mRing->readSpace());
+ hr = mDSCbuffer->Lock(LastCursor, NumBytes, &ReadPtr1, &ReadCnt1, &ReadPtr2, &ReadCnt2, 0);
+ }
+ if(SUCCEEDED(hr))
+ {
+ mRing->write(ReadPtr1, ReadCnt1/FrameSize);
+ if(ReadPtr2 != nullptr && ReadCnt2 > 0)
+ mRing->write(ReadPtr2, ReadCnt2/FrameSize);
+ hr = mDSCbuffer->Unlock(ReadPtr1, ReadCnt1, ReadPtr2, ReadCnt2);
+ mCursor = (LastCursor+ReadCnt1+ReadCnt2) % BufferBytes;
+ }
+
+ if(FAILED(hr))
+ {
+ ERR("update failed: 0x%08lx\n", hr);
+ aluHandleDisconnect(mDevice, "Failure retrieving capture data: 0x%lx", hr);
+ }
+
+ return static_cast<ALCuint>(mRing->readSpace());
+}
+
+} // namespace
+
+
+BackendFactory &DSoundBackendFactory::getFactory()
+{
+ static DSoundBackendFactory factory{};
+ return factory;
+}
+
+bool DSoundBackendFactory::init()
+{
+#ifdef HAVE_DYNLOAD
+ if(!ds_handle)
+ {
+ ds_handle = LoadLib("dsound.dll");
+ if(!ds_handle)
+ {
+ ERR("Failed to load dsound.dll\n");
+ return false;
+ }
+
+#define LOAD_FUNC(f) do { \
+ p##f = reinterpret_cast<decltype(p##f)>(GetSymbol(ds_handle, #f)); \
+ if(!p##f) \
+ { \
+ CloseLib(ds_handle); \
+ ds_handle = nullptr; \
+ return false; \
+ } \
+} while(0)
+ LOAD_FUNC(DirectSoundCreate);
+ LOAD_FUNC(DirectSoundEnumerateW);
+ LOAD_FUNC(DirectSoundCaptureCreate);
+ LOAD_FUNC(DirectSoundCaptureEnumerateW);
+#undef LOAD_FUNC
+ }
+#endif
+ return true;
+}
+
+bool DSoundBackendFactory::querySupport(BackendType type)
+{ return (type == BackendType::Playback || type == BackendType::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 DevProbe::Playback:
+ 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 DevProbe::Capture:
+ 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();
+}
+
+BackendPtr DSoundBackendFactory::createBackend(ALCdevice *device, BackendType type)
+{
+ if(type == BackendType::Playback)
+ return BackendPtr{new DSoundPlayback{device}};
+ if(type == BackendType::Capture)
+ return BackendPtr{new DSoundCapture{device}};
+ return nullptr;
+}