static void test_uninitialized(IAudioClient *ac) { HRESULT hr; UINT32 num; REFERENCE_TIME t1; HANDLE handle = CreateEventW(NULL, FALSE, FALSE, NULL); IUnknown *unk; hr = IAudioClient_GetBufferSize(ac, &num); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized GetBufferSize call returns %08x\n", hr); hr = IAudioClient_GetStreamLatency(ac, &t1); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized GetStreamLatency call returns %08x\n", hr); hr = IAudioClient_GetCurrentPadding(ac, &num); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized GetCurrentPadding call returns %08x\n", hr); hr = IAudioClient_Start(ac); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized Start call returns %08x\n", hr); hr = IAudioClient_Stop(ac); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized Stop call returns %08x\n", hr); hr = IAudioClient_Reset(ac); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized Reset call returns %08x\n", hr); hr = IAudioClient_SetEventHandle(ac, handle); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized SetEventHandle call returns %08x\n", hr); hr = IAudioClient_GetService(ac, &IID_IAudioStreamVolume, (void**)&unk); ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Uninitialized GetService call returns %08x\n", hr); CloseHandle(handle); }
JNIEXPORT jint JNICALL Java_org_jitsi_impl_neomedia_jmfext_media_protocol_wasapi_WASAPI_IAudioClient_1GetBufferSize (JNIEnv *env, jclass clazz, jlong thiz) { HRESULT hr; UINT32 numBufferFrames; hr = IAudioClient_GetBufferSize( (IAudioClient *) (intptr_t) thiz, &numBufferFrames); if (FAILED(hr)) { numBufferFrames = 0; WASAPI_throwNewHResultException(env, hr, __func__, __LINE__); } return (jint) numBufferFrames; }
HRESULT DSOUND_ReopenDevice(DirectSoundDevice *device, BOOL forcewave) { HRESULT hres; REFERENCE_TIME period; UINT32 frames; DWORD period_ms; IAudioClient *client = NULL; IAudioRenderClient *render = NULL; IAudioStreamVolume *volume = NULL; DWORD fraglen; WAVEFORMATEX *wfx = NULL; DWORD oldspeakerconfig = device->speaker_config; TRACE("(%p, %d)\n", device, forcewave); hres = IMMDevice_Activate(device->mmdevice, &IID_IAudioClient, CLSCTX_INPROC_SERVER, NULL, (void **)&client); if(FAILED(hres)){ WARN("Activate failed: %08x\n", hres); return hres; } hres = DSOUND_WaveFormat(device, client, forcewave, &wfx); if (FAILED(hres)) { IAudioClient_Release(client); return hres; } hres = IAudioClient_Initialize(client, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_NOPERSIST | AUDCLNT_STREAMFLAGS_EVENTCALLBACK, 800000, 0, wfx, NULL); if(FAILED(hres)){ IAudioClient_Release(client); ERR("Initialize failed: %08x\n", hres); return hres; } IAudioClient_SetEventHandle(client, device->sleepev); hres = IAudioClient_GetService(client, &IID_IAudioRenderClient, (void**)&render); if(FAILED(hres)) goto err_service; hres = IAudioClient_GetService(client, &IID_IAudioStreamVolume, (void**)&volume); if(FAILED(hres)) goto err_service; /* Now kick off the timer so the event fires periodically */ hres = IAudioClient_Start(client); if (FAILED(hres)) { WARN("Start failed with %08x\n", hres); goto err; } hres = IAudioClient_GetStreamLatency(client, &period); if (FAILED(hres)) { WARN("GetStreamLatency failed with %08x\n", hres); goto err; } hres = IAudioClient_GetBufferSize(client, &frames); if (FAILED(hres)) { WARN("GetBufferSize failed with %08x\n", hres); goto err; } period_ms = (period + 9999) / 10000; fraglen = MulDiv(wfx->nSamplesPerSec, period, 10000000) * wfx->nBlockAlign; TRACE("period %u ms fraglen %u buflen %u\n", period_ms, fraglen, frames * wfx->nBlockAlign); hres = DSOUND_PrimaryOpen(device, wfx, frames, forcewave); if(FAILED(hres)) goto err; DSOUND_ReleaseDevice(device); device->client = client; device->render = render; device->volume = volume; device->fraglen = fraglen; device->aclen = frames * wfx->nBlockAlign; if (period_ms < 3) device->sleeptime = 5; else device->sleeptime = period_ms * 5 / 2; return S_OK; err_service: WARN("GetService failed: %08x\n", hres); err: device->speaker_config = oldspeakerconfig; DSOUND_ParseSpeakerConfig(device); if (volume) IAudioStreamVolume_Release(volume); if (render) IAudioRenderClient_Release(render); IAudioClient_Release(client); HeapFree(GetProcessHeap(), 0, wfx); return hres; }
static HRESULT DoReset(ALCdevice *device) { MMDevApiData *data = device->ExtraData; WAVEFORMATEXTENSIBLE OutputType; WAVEFORMATEX *wfx = NULL; REFERENCE_TIME min_per, buf_time; UINT32 buffer_len, min_len; HRESULT hr; hr = IAudioClient_GetMixFormat(data->client, &wfx); if(FAILED(hr)) { ERR("Failed to get mix format: 0x%08lx\n", hr); return hr; } if(!MakeExtensible(&OutputType, wfx)) { CoTaskMemFree(wfx); return E_FAIL; } CoTaskMemFree(wfx); wfx = NULL; buf_time = ((REFERENCE_TIME)device->UpdateSize*device->NumUpdates*10000000 + device->Frequency-1) / device->Frequency; if(!(device->Flags&DEVICE_FREQUENCY_REQUEST)) device->Frequency = OutputType.Format.nSamplesPerSec; if(!(device->Flags&DEVICE_CHANNELS_REQUEST)) { if(OutputType.Format.nChannels == 1 && OutputType.dwChannelMask == MONO) device->FmtChans = DevFmtMono; else if(OutputType.Format.nChannels == 2 && OutputType.dwChannelMask == STEREO) device->FmtChans = DevFmtStereo; else if(OutputType.Format.nChannels == 4 && OutputType.dwChannelMask == QUAD) device->FmtChans = DevFmtQuad; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1) device->FmtChans = DevFmtX51; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1SIDE) device->FmtChans = DevFmtX51Side; else if(OutputType.Format.nChannels == 7 && OutputType.dwChannelMask == X6DOT1) device->FmtChans = DevFmtX61; else if(OutputType.Format.nChannels == 8 && OutputType.dwChannelMask == X7DOT1) device->FmtChans = DevFmtX71; else ERR("Unhandled channel config: %d -- 0x%08lx\n", OutputType.Format.nChannels, OutputType.dwChannelMask); } switch(device->FmtChans) { case DevFmtMono: OutputType.Format.nChannels = 1; OutputType.dwChannelMask = MONO; break; case DevFmtStereo: OutputType.Format.nChannels = 2; OutputType.dwChannelMask = STEREO; break; case DevFmtQuad: OutputType.Format.nChannels = 4; OutputType.dwChannelMask = QUAD; break; case DevFmtX51: OutputType.Format.nChannels = 6; OutputType.dwChannelMask = X5DOT1; break; case DevFmtX51Side: OutputType.Format.nChannels = 6; OutputType.dwChannelMask = X5DOT1SIDE; break; case DevFmtX61: OutputType.Format.nChannels = 7; OutputType.dwChannelMask = X6DOT1; break; case DevFmtX71: OutputType.Format.nChannels = 8; OutputType.dwChannelMask = X7DOT1; break; } switch(device->FmtType) { case DevFmtByte: device->FmtType = DevFmtUByte; /* fall-through */ case DevFmtUByte: OutputType.Format.wBitsPerSample = 8; OutputType.Samples.wValidBitsPerSample = 8; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtUShort: device->FmtType = DevFmtShort; /* fall-through */ case DevFmtShort: OutputType.Format.wBitsPerSample = 16; OutputType.Samples.wValidBitsPerSample = 16; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtUInt: device->FmtType = DevFmtInt; /* fall-through */ case DevFmtInt: OutputType.Format.wBitsPerSample = 32; OutputType.Samples.wValidBitsPerSample = 32; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; break; case DevFmtFloat: OutputType.Format.wBitsPerSample = 32; OutputType.Samples.wValidBitsPerSample = 32; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT; break; } OutputType.Format.nSamplesPerSec = device->Frequency; OutputType.Format.nBlockAlign = OutputType.Format.nChannels * OutputType.Format.wBitsPerSample / 8; OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec * OutputType.Format.nBlockAlign; hr = IAudioClient_IsFormatSupported(data->client, AUDCLNT_SHAREMODE_SHARED, &OutputType.Format, &wfx); if(FAILED(hr)) { ERR("Failed to check format support: 0x%08lx\n", hr); hr = IAudioClient_GetMixFormat(data->client, &wfx); } if(FAILED(hr)) { ERR("Failed to find a supported format: 0x%08lx\n", hr); return hr; } if(wfx != NULL) { if(!MakeExtensible(&OutputType, wfx)) { CoTaskMemFree(wfx); return E_FAIL; } CoTaskMemFree(wfx); wfx = NULL; device->Frequency = OutputType.Format.nSamplesPerSec; if(OutputType.Format.nChannels == 1 && OutputType.dwChannelMask == MONO) device->FmtChans = DevFmtMono; else if(OutputType.Format.nChannels == 2 && OutputType.dwChannelMask == STEREO) device->FmtChans = DevFmtStereo; else if(OutputType.Format.nChannels == 4 && OutputType.dwChannelMask == QUAD) device->FmtChans = DevFmtQuad; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1) device->FmtChans = DevFmtX51; else if(OutputType.Format.nChannels == 6 && OutputType.dwChannelMask == X5DOT1SIDE) device->FmtChans = DevFmtX51Side; else if(OutputType.Format.nChannels == 7 && OutputType.dwChannelMask == X6DOT1) device->FmtChans = DevFmtX61; else if(OutputType.Format.nChannels == 8 && OutputType.dwChannelMask == X7DOT1) device->FmtChans = DevFmtX71; else { ERR("Unhandled extensible channels: %d -- 0x%08lx\n", OutputType.Format.nChannels, OutputType.dwChannelMask); device->FmtChans = DevFmtStereo; OutputType.Format.nChannels = 2; OutputType.dwChannelMask = STEREO; } if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_PCM)) { if(OutputType.Format.wBitsPerSample == 8) device->FmtType = DevFmtUByte; else if(OutputType.Format.wBitsPerSample == 16) device->FmtType = DevFmtShort; else if(OutputType.Format.wBitsPerSample == 32) device->FmtType = DevFmtInt; else { device->FmtType = DevFmtShort; OutputType.Format.wBitsPerSample = 16; } } else if(IsEqualGUID(&OutputType.SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) { device->FmtType = DevFmtFloat; OutputType.Format.wBitsPerSample = 32; } else { ERR("Unhandled format sub-type\n"); device->FmtType = DevFmtShort; OutputType.Format.wBitsPerSample = 16; OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM; } OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample; } SetDefaultWFXChannelOrder(device); hr = IAudioClient_Initialize(data->client, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, buf_time, 0, &OutputType.Format, NULL); if(FAILED(hr)) { ERR("Failed to initialize audio client: 0x%08lx\n", hr); return hr; } hr = IAudioClient_GetDevicePeriod(data->client, &min_per, NULL); if(SUCCEEDED(hr)) { min_len = (UINT32)((min_per*device->Frequency + 10000000-1) / 10000000); /* Find the nearest multiple of the period size to the update size */ if(min_len < device->UpdateSize) min_len *= (device->UpdateSize + min_len/2)/min_len; hr = IAudioClient_GetBufferSize(data->client, &buffer_len); } if(FAILED(hr)) { ERR("Failed to get audio buffer info: 0x%08lx\n", hr); return hr; } device->UpdateSize = min_len; device->NumUpdates = buffer_len / device->UpdateSize; if(device->NumUpdates <= 1) { ERR("Audio client returned buffer_len < period*2; expect break up\n"); device->NumUpdates = 2; device->UpdateSize = buffer_len / device->NumUpdates; } return hr; }
static ALuint MMDevApiProc(ALvoid *ptr) { ALCdevice *device = ptr; MMDevApiData *data = device->ExtraData; UINT32 buffer_len, written; ALuint update_size, len; BYTE *buffer; HRESULT hr; hr = CoInitialize(NULL); if(FAILED(hr)) { ERR("CoInitialize(NULL) failed: 0x%08lx\n", hr); aluHandleDisconnect(device); return 0; } hr = IAudioClient_GetBufferSize(data->client, &buffer_len); if(FAILED(hr)) { ERR("Failed to get audio buffer size: 0x%08lx\n", hr); aluHandleDisconnect(device); CoUninitialize(); return 0; } SetRTPriority(); update_size = device->UpdateSize; while(!data->killNow) { hr = IAudioClient_GetCurrentPadding(data->client, &written); if(FAILED(hr)) { ERR("Failed to get padding: 0x%08lx\n", hr); aluHandleDisconnect(device); break; } len = buffer_len - written; if(len < update_size) { DWORD res; res = WaitForSingleObjectEx(data->NotifyEvent, 2000, FALSE); if(res != WAIT_OBJECT_0) ERR("WaitForSingleObjectEx error: 0x%lx\n", res); continue; } len -= len%update_size; hr = IAudioRenderClient_GetBuffer(data->render, len, &buffer); if(SUCCEEDED(hr)) { aluMixData(device, buffer, len); hr = IAudioRenderClient_ReleaseBuffer(data->render, len, 0); } if(FAILED(hr)) { ERR("Failed to buffer data: 0x%08lx\n", hr); aluHandleDisconnect(device); break; } } CoUninitialize(); return 0; }
static void test_capture(IAudioClient *ac, HANDLE handle, WAVEFORMATEX *wfx) { IAudioCaptureClient *acc; HRESULT hr; UINT32 frames, next, pad, sum = 0; BYTE *data; DWORD flags; UINT64 pos, qpc; REFERENCE_TIME period; hr = IAudioClient_GetService(ac, &IID_IAudioCaptureClient, (void**)&acc); ok(hr == S_OK, "IAudioClient_GetService(IID_IAudioCaptureClient) returns %08x\n", hr); if (hr != S_OK) return; frames = 0xabadcafe; data = (void*)0xdeadf00d; flags = 0xabadcafe; pos = qpc = 0xdeadbeef; hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == AUDCLNT_S_BUFFER_EMPTY, "Initial IAudioCaptureClient_GetBuffer returns %08x\n", hr); /* should be empty right after start. Otherwise consume one packet */ if(hr == S_OK){ hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); sum += frames; frames = 0xabadcafe; data = (void*)0xdeadf00d; flags = 0xabadcafe; pos = qpc = 0xdeadbeef; hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == AUDCLNT_S_BUFFER_EMPTY, "Initial IAudioCaptureClient_GetBuffer returns %08x\n", hr); } if(hr == AUDCLNT_S_BUFFER_EMPTY){ ok(!frames, "frames changed to %u\n", frames); ok(data == (void*)0xdeadf00d, "data changed to %p\n", data); ok(flags == 0xabadcafe, "flags changed to %x\n", flags); ok(pos == 0xdeadbeef, "position changed to %u\n", (UINT)pos); ok(qpc == 0xdeadbeef, "timer changed to %u\n", (UINT)qpc); /* GetNextPacketSize yields 0 if no data is yet available * it is not constantly period_size * SamplesPerSec */ hr = IAudioCaptureClient_GetNextPacketSize(acc, &next); ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr); ok(!next, "GetNextPacketSize %u\n", next); } hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); sum += frames; ok(ResetEvent(handle), "ResetEvent\n"); hr = IAudioCaptureClient_GetNextPacketSize(acc, &next); ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr); hr = IAudioClient_GetCurrentPadding(ac, &pad); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); ok(next == pad, "GetNextPacketSize %u vs. GCP %u\n", next, pad); /* later GCP will grow, while GNPS is 0 or period size */ hr = IAudioCaptureClient_GetNextPacketSize(acc, NULL); ok(hr == E_POINTER, "IAudioCaptureClient_GetNextPacketSize(NULL) returns %08x\n", hr); data = (void*)0xdeadf00d; frames = 0xdeadbeef; flags = 0xabadcafe; hr = IAudioCaptureClient_GetBuffer(acc, &data, NULL, NULL, NULL, NULL); ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(data, NULL, NULL) returns %08x\n", hr); hr = IAudioCaptureClient_GetBuffer(acc, NULL, &frames, NULL, NULL, NULL); ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, &frames, NULL) returns %08x\n", hr); hr = IAudioCaptureClient_GetBuffer(acc, NULL, NULL, &flags, NULL, NULL); ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, NULL, &flags) returns %08x\n", hr); hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, NULL, NULL, NULL); ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(&ata, &frames, NULL) returns %08x\n", hr); ok((DWORD_PTR)data == 0xdeadf00d, "data is reset to %p\n", data); ok(frames == 0xdeadbeef, "frames is reset to %08x\n", frames); ok(flags == 0xabadcafe, "flags is reset to %08x\n", flags); hr = IAudioClient_GetDevicePeriod(ac, &period, NULL); ok(hr == S_OK, "GetDevicePeriod failed: %08x\n", hr); period = MulDiv(period, wfx->nSamplesPerSec, 10000000); /* as in render.c */ ok(WaitForSingleObject(handle, 1000) == WAIT_OBJECT_0, "Waiting on event handle failed!\n"); data = (void*)0xdeadf00d; hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == S_OK || hr == AUDCLNT_S_BUFFER_EMPTY, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr); if (hr == S_OK){ ok(frames, "Amount of frames locked is 0!\n"); /* broken: some w7 machines return pad == 0 and DATA_DISCONTINUITY here, * AUDCLNT_S_BUFFER_EMPTY above, yet pos == 1-2 * period rather than 0 */ ok(pos == sum || broken(pos == period || pos == 2*period), "Position %u expected %u\n", (UINT)pos, sum); sum = pos; }else if (hr == AUDCLNT_S_BUFFER_EMPTY){ ok(!frames, "Amount of frames locked with empty buffer is %u!\n", frames); ok(data == (void*)0xdeadf00d, "No data changed to %p\n", data); } trace("Wait'ed position %d pad %u flags %x, amount of frames locked: %u\n", hr==S_OK ? (UINT)pos : -1, pad, flags, frames); hr = IAudioCaptureClient_GetNextPacketSize(acc, &next); ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr); ok(next == frames, "GetNextPacketSize %u vs. GetBuffer %u\n", next, frames); hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); hr = IAudioCaptureClient_ReleaseBuffer(acc, 0); ok(hr == S_OK, "Releasing 0 returns %08x\n", hr); hr = IAudioCaptureClient_GetNextPacketSize(acc, &next); ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr); if (frames) { hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Releasing buffer twice returns %08x\n", hr); sum += frames; } Sleep(350); /* for sure there's data now */ hr = IAudioClient_GetCurrentPadding(ac, &pad); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); /** GetNextPacketSize * returns either 0 or one period worth of frames * whereas GetCurrentPadding grows when input is not consumed. */ hr = IAudioCaptureClient_GetNextPacketSize(acc, &next); ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr); ok(next < pad, "GetNextPacketSize %u vs. GCP %u\n", next, pad); hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr); ok(next == frames, "GetNextPacketSize %u vs. GetBuffer %u\n", next, frames); if(hr == S_OK){ UINT32 frames2 = frames; UINT64 pos2, qpc2; ok(frames, "Amount of frames locked is 0!\n"); ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum); hr = IAudioCaptureClient_ReleaseBuffer(acc, 0); ok(hr == S_OK, "Releasing 0 returns %08x\n", hr); /* GCP did not decrement, no data consumed */ hr = IAudioClient_GetCurrentPadding(ac, &frames); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); ok(frames == pad || frames == pad + next /* concurrent feeder */, "GCP %u past ReleaseBuffer(0) initially %u\n", frames, pad); /* should re-get the same data */ hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos2, &qpc2); ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr); ok(frames2 == frames, "GetBuffer after ReleaseBuffer(0) %u/%u\n", frames2, frames); ok(pos2 == pos, "Position after ReleaseBuffer(0) %u/%u\n", (UINT)pos2, (UINT)pos); todo_wine ok(qpc2 == qpc, "HPC after ReleaseBuffer(0) %u vs. %u\n", (UINT)qpc2, (UINT)qpc); } /* trace after the GCP test because log output to MS-DOS console disturbs timing */ trace("Sleep.1 position %d pad %u flags %x, amount of frames locked: %u\n", hr==S_OK ? (UINT)pos : -1, pad, flags, frames); if(hr == S_OK){ UINT32 frames2 = 0xabadcafe; BYTE *data2 = (void*)0xdeadf00d; flags = 0xabadcafe; ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum); pos = qpc = 0xdeadbeef; hr = IAudioCaptureClient_GetBuffer(acc, &data2, &frames2, &flags, &pos, &qpc); ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Out of order IAudioCaptureClient_GetBuffer returns %08x\n", hr); ok(frames2 == 0xabadcafe, "Out of order frames changed to %x\n", frames2); ok(data2 == (void*)0xdeadf00d, "Out of order data changed to %p\n", data2); ok(flags == 0xabadcafe, "Out of order flags changed to %x\n", flags); ok(pos == 0xdeadbeef, "Out of order position changed to %x\n", (UINT)pos); ok(qpc == 0xdeadbeef, "Out of order timer changed to %x\n", (UINT)qpc); hr = IAudioCaptureClient_ReleaseBuffer(acc, frames+1); ok(hr == AUDCLNT_E_INVALID_SIZE, "Releasing buffer+1 returns %08x\n", hr); hr = IAudioCaptureClient_ReleaseBuffer(acc, 1); ok(hr == AUDCLNT_E_INVALID_SIZE, "Releasing 1 returns %08x\n", hr); hr = IAudioClient_Reset(ac); ok(hr == AUDCLNT_E_NOT_STOPPED, "Reset failed: %08x\n", hr); } hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); if (frames) { sum += frames; hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Releasing buffer twice returns %08x\n", hr); } frames = period; ok(next == frames, "GetNextPacketSize %u vs. GetDevicePeriod %u\n", next, frames); /* GetBufferSize is not a multiple of the period size! */ hr = IAudioClient_GetBufferSize(ac, &next); ok(hr == S_OK, "GetBufferSize failed: %08x\n", hr); trace("GetBufferSize %u period size %u\n", next, frames); Sleep(400); /* overrun */ hr = IAudioClient_GetCurrentPadding(ac, &pad); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr); trace("Overrun position %d pad %u flags %x, amount of frames locked: %u\n", hr==S_OK ? (UINT)pos : -1, pad, flags, frames); if(hr == S_OK){ /* The discontinuity is reported here, but is this an old or new packet? */ todo_wine ok(flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY, "expect DISCONTINUITY %x\n", flags); ok(pad == next, "GCP %u vs. BufferSize %u\n", (UINT32)pad, next); /* Native's position is one period further than what we read. * Perhaps that's precisely the meaning of DATA_DISCONTINUITY: * signal when the position jump left a gap. */ todo_wine ok(pos == sum + frames, "Position %u gap %d\n", (UINT)pos, (UINT)pos - sum); if(flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) sum = pos; } hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); sum += frames; hr = IAudioClient_GetCurrentPadding(ac, &pad); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr); trace("Cont'ed position %d pad %u flags %x, amount of frames locked: %u\n", hr==S_OK ? (UINT)pos : -1, pad, flags, frames); if(hr == S_OK){ ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum); ok(!flags, "flags %u\n", flags); hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); sum += frames; } hr = IAudioClient_Stop(ac); ok(hr == S_OK, "Stop on a started stream returns %08x\n", hr); hr = IAudioClient_Start(ac); ok(hr == S_OK, "Start on a stopped stream returns %08x\n", hr); hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr); hr = IAudioClient_GetCurrentPadding(ac, &pad); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); trace("Restart position %d pad %u flags %x, amount of frames locked: %u\n", hr==S_OK ? (UINT)pos : -1, pad, flags, frames); ok(pad > sum, "restarted GCP %u\n", pad); /* GCP is still near buffer size */ if(frames){ ok(pos == sum, "Position %u expected %u\n", (UINT)pos, sum); ok(!flags, "flags %u\n", flags); hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); sum += frames; } hr = IAudioClient_Stop(ac); ok(hr == S_OK, "Stop on a started stream returns %08x\n", hr); hr = IAudioClient_Reset(ac); ok(hr == S_OK, "Reset on a stopped stream returns %08x\n", hr); sum += pad - frames; hr = IAudioClient_Start(ac); ok(hr == S_OK, "Start on a stopped stream returns %08x\n", hr); hr = IAudioClient_GetCurrentPadding(ac, &pad); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); flags = 0xabadcafe; hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == AUDCLNT_S_BUFFER_EMPTY || /*PulseAudio*/hr == S_OK, "Initial IAudioCaptureClient_GetBuffer returns %08x\n", hr); trace("Reset position %d pad %u flags %x, amount of frames locked: %u\n", hr==S_OK ? (UINT)pos : -1, pad, flags, frames); if(hr == S_OK){ /* Only PulseAudio goes here; despite snd_pcm_drop it manages * to fill GetBufferSize with a single snd_pcm_read */ trace("Test marked todo: only PulseAudio gets here\n"); todo_wine ok(flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY, "expect DISCONTINUITY %x\n", flags); /* Reset zeroes padding, not the position */ ok(pos >= sum, "Position %u last %u\n", (UINT)pos, sum); /*sum = pos; check after next GetBuffer */ hr = IAudioCaptureClient_ReleaseBuffer(acc, frames); ok(hr == S_OK, "Releasing buffer returns %08x\n", hr); sum += frames; } else if(hr == AUDCLNT_S_BUFFER_EMPTY){ ok(!pad, "resetted GCP %u\n", pad); Sleep(180); } hr = IAudioClient_GetCurrentPadding(ac, &pad); ok(hr == S_OK, "GetCurrentPadding call returns %08x\n", hr); hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &pos, &qpc); ok(hr == S_OK, "Valid IAudioCaptureClient_GetBuffer returns %08x\n", hr); trace("Running position %d pad %u flags %x, amount of frames locked: %u\n", hr==S_OK ? (UINT)pos : -1, pad, flags, frames); if(hr == S_OK){ /* Some w7 machines signal DATA_DISCONTINUITY here following the * previous AUDCLNT_S_BUFFER_EMPTY, others not. What logic? */ ok(pos >= sum, "Position %u gap %d\n", (UINT)pos, (UINT)pos - sum); IAudioCaptureClient_ReleaseBuffer(acc, frames); } IAudioCaptureClient_Release(acc); }