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);
}
