static DWORD WINAPI DSOUND_capture_thread(void *user)
{
IDirectSoundCaptureBufferImpl *buffer = user;
HRESULT hr;
DWORD ret, wait_ms;
REFERENCE_TIME period;
hr = IAudioClient_GetDevicePeriod(buffer->device->client, &period, NULL);
if(FAILED(hr)){
WARN("GetDevicePeriod failed: %08x\n", hr);
wait_ms = 5;
}else
wait_ms = MulDiv(5, period, 10000);
while(buffer->ref){
ret = WaitForSingleObject(buffer->sleepev, wait_ms);
if(!buffer->device->ref)
break;
if(ret == WAIT_OBJECT_0){
EnterCriticalSection(&buffer->device->lock);
DSOUND_capture_data(buffer->device);
LeaveCriticalSection(&buffer->device->lock);
}else if(ret != WAIT_TIMEOUT)
WARN("WaitForSingleObject failed: %u\n", GetLastError());
}
return 0;
}
static DWORD DSOUND_fraglen(DirectSoundDevice *device)
{
REFERENCE_TIME period;
HRESULT hr;
DWORD ret;
hr = IAudioClient_GetDevicePeriod(device->client, &period, NULL);
if(FAILED(hr)){
/* just guess at 10ms */
WARN("GetDevicePeriod failed: %08x\n", hr);
ret = MulDiv(device->pwfx->nBlockAlign, device->pwfx->nSamplesPerSec, 100);
}else
ret = MulDiv(device->pwfx->nSamplesPerSec * device->pwfx->nBlockAlign, period, 10000000);
ret -= ret % device->pwfx->nBlockAlign;
return ret;
}
JNIEXPORT jlong JNICALL
Java_org_jitsi_impl_neomedia_jmfext_media_protocol_wasapi_WASAPI_IAudioClient_1GetMinimumDevicePeriod
(JNIEnv *env, jclass clazz, jlong thiz)
{
HRESULT hr;
REFERENCE_TIME hnsDefaultDevicePeriod;
REFERENCE_TIME hnsMinimumDevicePeriod;
hr
= IAudioClient_GetDevicePeriod(
(IAudioClient *) (intptr_t) thiz,
&hnsDefaultDevicePeriod,
&hnsMinimumDevicePeriod);
if (FAILED(hr))
{
hnsMinimumDevicePeriod = 0;
WASAPI_throwNewHResultException(env, hr, __func__, __LINE__);
}
return (jlong) hnsMinimumDevicePeriod;
}
static void test_audioclient(void)
{
IAudioClient *ac;
IUnknown *unk;
HRESULT hr;
ULONG ref;
WAVEFORMATEX *pwfx, *pwfx2;
REFERENCE_TIME t1, t2;
HANDLE handle;
hr = IMMDevice_Activate(dev, &IID_IAudioClient, CLSCTX_INPROC_SERVER,
NULL, (void**)&ac);
ok(hr == S_OK, "Activation failed with %08x\n", hr);
if(hr != S_OK)
return;
handle = CreateEventW(NULL, FALSE, FALSE, NULL);
hr = IAudioClient_QueryInterface(ac, &IID_IUnknown, NULL);
ok(hr == E_POINTER, "QueryInterface(NULL) returned %08x\n", hr);
unk = (void*)(LONG_PTR)0x12345678;
hr = IAudioClient_QueryInterface(ac, &IID_NULL, (void**)&unk);
ok(hr == E_NOINTERFACE, "QueryInterface(IID_NULL) returned %08x\n", hr);
ok(!unk, "QueryInterface(IID_NULL) returned non-null pointer %p\n", unk);
hr = IAudioClient_QueryInterface(ac, &IID_IUnknown, (void**)&unk);
ok(hr == S_OK, "QueryInterface(IID_IUnknown) returned %08x\n", hr);
if (unk)
{
ref = IUnknown_Release(unk);
ok(ref == 1, "Released count is %u\n", ref);
}
hr = IAudioClient_QueryInterface(ac, &IID_IAudioClient, (void**)&unk);
ok(hr == S_OK, "QueryInterface(IID_IAudioClient) returned %08x\n", hr);
if (unk)
{
ref = IUnknown_Release(unk);
ok(ref == 1, "Released count is %u\n", ref);
}
hr = IAudioClient_GetDevicePeriod(ac, NULL, NULL);
ok(hr == E_POINTER, "Invalid GetDevicePeriod call returns %08x\n", hr);
hr = IAudioClient_GetDevicePeriod(ac, &t1, NULL);
ok(hr == S_OK, "Valid GetDevicePeriod call returns %08x\n", hr);
hr = IAudioClient_GetDevicePeriod(ac, NULL, &t2);
ok(hr == S_OK, "Valid GetDevicePeriod call returns %08x\n", hr);
hr = IAudioClient_GetDevicePeriod(ac, &t1, &t2);
ok(hr == S_OK, "Valid GetDevicePeriod call returns %08x\n", hr);
trace("Returned periods: %u.%05u ms %u.%05u ms\n",
(UINT)(t1/10000), (UINT)(t1 % 10000),
(UINT)(t2/10000), (UINT)(t2 % 10000));
hr = IAudioClient_GetMixFormat(ac, NULL);
ok(hr == E_POINTER, "GetMixFormat returns %08x\n", hr);
hr = IAudioClient_GetMixFormat(ac, &pwfx);
ok(hr == S_OK, "Valid GetMixFormat returns %08x\n", hr);
if (hr == S_OK)
{
trace("pwfx: %p\n", pwfx);
trace("Tag: %04x\n", pwfx->wFormatTag);
trace("bits: %u\n", pwfx->wBitsPerSample);
trace("chan: %u\n", pwfx->nChannels);
trace("rate: %u\n", pwfx->nSamplesPerSec);
trace("align: %u\n", pwfx->nBlockAlign);
trace("extra: %u\n", pwfx->cbSize);
ok(pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE, "wFormatTag is %x\n", pwfx->wFormatTag);
if (pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE)
{
WAVEFORMATEXTENSIBLE *pwfxe = (void*)pwfx;
trace("Res: %u\n", pwfxe->Samples.wReserved);
trace("Mask: %x\n", pwfxe->dwChannelMask);
trace("Alg: %s\n",
IsEqualGUID(&pwfxe->SubFormat, &KSDATAFORMAT_SUBTYPE_PCM)?"PCM":
(IsEqualGUID(&pwfxe->SubFormat,
&KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)?"FLOAT":"Other"));
}
hr = IAudioClient_IsFormatSupported(ac, AUDCLNT_SHAREMODE_SHARED, pwfx, &pwfx2);
ok(hr == S_OK, "Valid IsFormatSupported(Shared) call returns %08x\n", hr);
ok(pwfx2 == NULL, "pwfx2 is non-null\n");
CoTaskMemFree(pwfx2);
hr = IAudioClient_IsFormatSupported(ac, AUDCLNT_SHAREMODE_SHARED, NULL, NULL);
ok(hr == E_POINTER, "IsFormatSupported(NULL) call returns %08x\n", hr);
hr = IAudioClient_IsFormatSupported(ac, AUDCLNT_SHAREMODE_SHARED, pwfx, NULL);
ok(hr == E_POINTER, "IsFormatSupported(Shared,NULL) call returns %08x\n", hr);
hr = IAudioClient_IsFormatSupported(ac, AUDCLNT_SHAREMODE_EXCLUSIVE, pwfx, NULL);
ok(hr == S_OK || hr == AUDCLNT_E_UNSUPPORTED_FORMAT, "IsFormatSupported(Exclusive) call returns %08x\n", hr);
hr = IAudioClient_IsFormatSupported(ac, AUDCLNT_SHAREMODE_EXCLUSIVE, pwfx, &pwfx2);
ok(hr == S_OK || hr == AUDCLNT_E_UNSUPPORTED_FORMAT, "IsFormatSupported(Exclusive) call returns %08x\n", hr);
ok(pwfx2 == NULL, "pwfx2 non-null on exclusive IsFormatSupported\n");
hr = IAudioClient_IsFormatSupported(ac, 0xffffffff, pwfx, NULL);
ok(hr == E_INVALIDARG ||
hr == AUDCLNT_E_UNSUPPORTED_FORMAT,
"IsFormatSupported(0xffffffff) call returns %08x\n", hr);
}
test_uninitialized(ac);
hr = IAudioClient_Initialize(ac, 3, 0, 5000000, 0, pwfx, NULL);
ok(hr == AUDCLNT_E_NOT_INITIALIZED, "Initialize with invalid sharemode returns %08x\n", hr);
hr = IAudioClient_Initialize(ac, AUDCLNT_SHAREMODE_SHARED, 0xffffffff, 5000000, 0, pwfx, NULL);
ok(hr == E_INVALIDARG, "Initialize with invalid flags returns %08x\n", hr);
/* It seems that if length > 2s or periodicity != 0 the length is ignored and call succeeds
* Since we can only initialize successfully once, skip those tests.
*/
hr = IAudioClient_Initialize(ac, AUDCLNT_SHAREMODE_SHARED, 0, 5000000, 0, NULL, NULL);
ok(hr == E_POINTER, "Initialize with null format returns %08x\n", hr);
hr = IAudioClient_Initialize(ac, AUDCLNT_SHAREMODE_SHARED, 0, 5000000, 0, pwfx, NULL);
ok(hr == S_OK, "Valid Initialize returns %08x\n", hr);
if (hr != S_OK)
{
skip("Cannot initialize %08x, remainder of tests is useless\n", hr);
CoTaskMemFree(pwfx);
return;
}
hr = IAudioClient_GetStreamLatency(ac, NULL);
ok(hr == E_POINTER, "GetStreamLatency(NULL) call returns %08x\n", hr);
hr = IAudioClient_GetStreamLatency(ac, &t1);
ok(hr == S_OK, "Valid GetStreamLatency call returns %08x\n", hr);
trace("Returned latency: %u.%05u ms\n",
(UINT)(t1/10000), (UINT)(t1 % 10000));
hr = IAudioClient_Initialize(ac, AUDCLNT_SHAREMODE_SHARED, 0, 5000000, 0, pwfx, NULL);
ok(hr == AUDCLNT_E_ALREADY_INITIALIZED, "Calling Initialize twice returns %08x\n", hr);
hr = IAudioClient_SetEventHandle(ac, NULL);
ok(hr == E_INVALIDARG, "SetEventHandle(NULL) returns %08x\n", hr);
hr = IAudioClient_SetEventHandle(ac, handle);
ok(hr == AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED ||
broken(hr == HRESULT_FROM_WIN32(ERROR_INVALID_NAME)) ||
broken(hr == HRESULT_FROM_WIN32(ERROR_FILE_NOT_FOUND)) /* Some 2k8 */ ||
broken(hr == HRESULT_FROM_WIN32(ERROR_BAD_PATHNAME)) /* Some Vista */
, "SetEventHandle returns %08x\n", hr);
hr = IAudioClient_Reset(ac);
ok(hr == S_OK, "Reset on a resetted stream returns %08x\n", hr);
hr = IAudioClient_Stop(ac);
ok(hr == S_FALSE, "Stop on a stopped stream returns %08x\n", hr);
hr = IAudioClient_Start(ac);
ok(hr == S_OK, "Start on a stopped stream returns %08x\n", hr);
IAudioClient_Release(ac);
CloseHandle(handle);
CoTaskMemFree(pwfx);
}
static void test_padding(void)
{
HRESULT hr;
IAudioClient *ac;
IAudioRenderClient *arc;
WAVEFORMATEX *pwfx;
REFERENCE_TIME minp, defp;
BYTE *buf;
UINT32 psize, pad, written;
hr = IMMDevice_Activate(dev, &IID_IAudioClient, CLSCTX_INPROC_SERVER,
NULL, (void**)&ac);
ok(hr == S_OK, "Activation failed with %08x\n", hr);
if(hr != S_OK)
return;
hr = IAudioClient_GetMixFormat(ac, &pwfx);
ok(hr == S_OK, "GetMixFormat failed: %08x\n", hr);
if(hr != S_OK)
return;
hr = IAudioClient_Initialize(ac, AUDCLNT_SHAREMODE_SHARED,
0, 5000000, 0, pwfx, NULL);
ok(hr == S_OK, "Initialize failed: %08x\n", hr);
hr = IAudioClient_GetDevicePeriod(ac, &defp, &minp);
ok(hr == S_OK, "GetDevicePeriod failed: %08x\n", hr);
ok(defp != 0, "Default period is 0\n");
ok(minp != 0, "Minimum period is 0\n");
ok(minp <= defp, "Mininum period is greater than default period\n");
hr = IAudioClient_GetService(ac, &IID_IAudioRenderClient, (void**)&arc);
ok(hr == S_OK, "GetService failed: %08x\n", hr);
psize = (defp / 10000000.) * pwfx->nSamplesPerSec * pwfx->nBlockAlign;
written = 0;
hr = IAudioClient_GetCurrentPadding(ac, &pad);
ok(hr == S_OK, "GetCurrentPadding failed: %08x\n", hr);
ok(pad == written, "GetCurrentPadding returned %u, should be %u\n", pad, written);
hr = IAudioRenderClient_GetBuffer(arc, psize, &buf);
ok(hr == S_OK, "GetBuffer failed: %08x\n", hr);
ok(buf != NULL, "NULL buffer returned\n");
hr = IAudioRenderClient_ReleaseBuffer(arc, psize,
AUDCLNT_BUFFERFLAGS_SILENT);
ok(hr == S_OK, "ReleaseBuffer failed: %08x\n", hr);
written += psize;
hr = IAudioClient_GetCurrentPadding(ac, &pad);
ok(hr == S_OK, "GetCurrentPadding failed: %08x\n", hr);
ok(pad == written, "GetCurrentPadding returned %u, should be %u\n", pad, written);
psize = (minp / 10000000.) * pwfx->nSamplesPerSec * pwfx->nBlockAlign;
hr = IAudioRenderClient_GetBuffer(arc, psize, &buf);
ok(hr == S_OK, "GetBuffer failed: %08x\n", hr);
ok(buf != NULL, "NULL buffer returned\n");
hr = IAudioRenderClient_ReleaseBuffer(arc, psize,
AUDCLNT_BUFFERFLAGS_SILENT);
ok(hr == S_OK, "ReleaseBuffer failed: %08x\n", hr);
written += psize;
hr = IAudioClient_GetCurrentPadding(ac, &pad);
ok(hr == S_OK, "GetCurrentPadding failed: %08x\n", hr);
ok(pad == written, "GetCurrentPadding returned %u, should be %u\n", pad, written);
/* overfull buffer. requested 1/2s buffer size, so try
* to get a 1/2s buffer, which should fail */
psize = pwfx->nSamplesPerSec / 2.;
hr = IAudioRenderClient_GetBuffer(arc, psize, &buf);
ok(hr == AUDCLNT_E_BUFFER_TOO_LARGE, "GetBuffer gave wrong error: %08x\n", hr);
hr = IAudioRenderClient_ReleaseBuffer(arc, psize, 0);
ok(hr == AUDCLNT_E_OUT_OF_ORDER, "ReleaseBuffer gave wrong error: %08x\n", hr);
hr = IAudioClient_GetCurrentPadding(ac, &pad);
ok(hr == S_OK, "GetCurrentPadding failed: %08x\n", hr);
ok(pad == written, "GetCurrentPadding returned %u, should be %u\n", pad, written);
CoTaskMemFree(pwfx);
IAudioRenderClient_Release(arc);
IAudioClient_Release(ac);
}
static gboolean
gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
{
GstWasapiSink *self = GST_WASAPI_SINK (asink);
gboolean res = FALSE;
HRESULT hr;
REFERENCE_TIME latency_rt, def_period, min_period;
WAVEFORMATEXTENSIBLE format;
IAudioRenderClient *render_client = NULL;
hr = IAudioClient_GetDevicePeriod (self->client, &def_period, &min_period);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetDevicePeriod () failed");
goto beach;
}
gst_wasapi_util_audio_info_to_waveformatex (&spec->info, &format);
self->info = spec->info;
hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
spec->buffer_time / 100, 0, (WAVEFORMATEX *) & format, NULL);
if (hr != S_OK) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("IAudioClient::Initialize () failed: %s",
gst_wasapi_util_hresult_to_string (hr)));
goto beach;
}
hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetStreamLatency () failed");
goto beach;
}
GST_INFO_OBJECT (self, "default period: %d (%d ms), "
"minimum period: %d (%d ms), "
"latency: %d (%d ms)",
(guint32) def_period, (guint32) def_period / 10000,
(guint32) min_period, (guint32) min_period / 10000,
(guint32) latency_rt, (guint32) latency_rt / 10000);
/* FIXME: What to do with the latency? */
hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
goto beach;
}
if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client,
&render_client)) {
goto beach;
}
hr = IAudioClient_Start (self->client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
goto beach;
}
self->render_client = render_client;
render_client = NULL;
res = TRUE;
beach:
if (render_client != NULL)
IUnknown_Release (render_client);
return res;
}
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 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);
}