static gboolean
gst_wasapi_src_start (GstBaseSrc * src)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
gboolean res = FALSE;
IAudioClient *client = NULL;
IAudioClock *client_clock = NULL;
guint64 client_clock_freq = 0;
IAudioCaptureClient *capture_client = NULL;
HRESULT hr;
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
&self->latency))
goto beach;
hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
"failed");
goto beach;
}
hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed");
goto beach;
}
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
&capture_client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioCaptureClient) failed");
goto beach;
}
hr = IAudioClient_Start (client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
goto beach;
}
self->client = client;
self->client_clock = client_clock;
self->client_clock_freq = client_clock_freq;
self->capture_client = capture_client;
res = TRUE;
beach:
if (!res) {
if (capture_client != NULL)
IUnknown_Release (capture_client);
if (client_clock != NULL)
IUnknown_Release (client_clock);
if (client != NULL)
IUnknown_Release (client);
}
return res;
}
static void test_clock(void)
{
HRESULT hr;
IAudioClient *ac;
IAudioClock *acl;
IAudioRenderClient *arc;
UINT64 freq, pos, pcpos, last;
BYTE *data;
WAVEFORMATEX *pwfx;
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_GetService(ac, &IID_IAudioClock, (void**)&acl);
ok(hr == S_OK, "GetService(IAudioClock) failed: %08x\n", hr);
hr = IAudioClock_GetFrequency(acl, &freq);
ok(hr == S_OK, "GetFrequency failed: %08x\n", hr);
hr = IAudioClock_GetPosition(acl, NULL, NULL);
ok(hr == E_POINTER, "GetPosition wrong error: %08x\n", hr);
pcpos = 0;
hr = IAudioClock_GetPosition(acl, &pos, &pcpos);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos == 0, "GetPosition returned non-zero pos before being started\n");
ok(pcpos != 0, "GetPosition returned zero pcpos\n");
last = pos;
hr = IAudioClient_GetService(ac, &IID_IAudioRenderClient, (void**)&arc);
ok(hr == S_OK, "GetService(IAudioRenderClient) failed: %08x\n", hr);
hr = IAudioRenderClient_GetBuffer(arc, pwfx->nSamplesPerSec / 2., &data);
ok(hr == S_OK, "GetBuffer failed: %08x\n", hr);
hr = IAudioRenderClient_ReleaseBuffer(arc, pwfx->nSamplesPerSec / 2., AUDCLNT_BUFFERFLAGS_SILENT);
ok(hr == S_OK, "ReleaseBuffer failed: %08x\n", hr);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos == 0, "GetPosition returned non-zero pos before being started\n");
hr = IAudioClient_Start(ac);
ok(hr == S_OK, "Start failed: %08x\n", hr);
Sleep(100);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos > 0, "Position should have been further along...\n");
last = pos;
hr = IAudioClient_Stop(ac);
ok(hr == S_OK, "Stop failed: %08x\n", hr);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos >= last, "Position should have been further along...\n");
last = pos;
hr = IAudioClient_Start(ac);
ok(hr == S_OK, "Start failed: %08x\n", hr);
Sleep(100);
hr = IAudioClient_Stop(ac);
ok(hr == S_OK, "Stop failed: %08x\n", hr);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos >= last, "Position should have been further along...\n");
last = pos;
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos == last, "Position should have been further along...\n");
hr = IAudioClient_Reset(ac);
ok(hr == S_OK, "Reset failed: %08x\n", hr);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos == 0, "GetPosition returned non-zero pos after Reset\n");
last = pos;
hr = IAudioRenderClient_GetBuffer(arc, pwfx->nSamplesPerSec / 2., &data);
ok(hr == S_OK, "GetBuffer failed: %08x\n", hr);
hr = IAudioRenderClient_ReleaseBuffer(arc, pwfx->nSamplesPerSec / 2., AUDCLNT_BUFFERFLAGS_SILENT);
ok(hr == S_OK, "ReleaseBuffer failed: %08x\n", hr);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos == 0, "GetPosition returned non-zero pos after Reset\n");
last = pos;
hr = IAudioClient_Start(ac);
ok(hr == S_OK, "Start failed: %08x\n", hr);
Sleep(100);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos > last, "Position should have been further along...\n");
hr = IAudioClient_Stop(ac);
ok(hr == S_OK, "Stop failed: %08x\n", hr);
hr = IAudioClock_GetPosition(acl, &pos, NULL);
ok(hr == S_OK, "GetPosition failed: %08x\n", hr);
ok(pos >= last, "Position should have been further along...\n");
IAudioClock_Release(acl);
IAudioClient_Release(ac);
}
/**
* Perform mixing for a Direct Sound device. That is, go through all the
* secondary buffers (the sound bites currently playing) and mix them in
* to the primary buffer (the device buffer).
*/
static void DSOUND_PerformMix(DirectSoundDevice *device)
{
UINT64 clock_pos, clock_freq, pos_bytes;
UINT delta_frags;
HRESULT hr;
TRACE("(%p)\n", device);
/* **** */
EnterCriticalSection(&device->mixlock);
hr = IAudioClock_GetFrequency(device->clock, &clock_freq);
if(FAILED(hr)){
WARN("GetFrequency failed: %08x\n", hr);
LeaveCriticalSection(&device->mixlock);
return;
}
hr = IAudioClock_GetPosition(device->clock, &clock_pos, NULL);
if(FAILED(hr)){
WARN("GetCurrentPadding failed: %08x\n", hr);
LeaveCriticalSection(&device->mixlock);
return;
}
pos_bytes = (clock_pos * device->pwfx->nSamplesPerSec * device->pwfx->nBlockAlign) / clock_freq;
delta_frags = (pos_bytes - device->last_pos_bytes) / device->fraglen;
if(delta_frags > 0){
device->pwplay += delta_frags;
device->pwplay %= device->helfrags;
device->pwqueue -= delta_frags;
device->last_pos_bytes = pos_bytes - (pos_bytes % device->fraglen);
}
if (device->priolevel != DSSCL_WRITEPRIMARY) {
BOOL recover = FALSE, all_stopped = FALSE;
DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2, mixplaypos, mixplaypos2;
LPVOID buf1, buf2;
int nfiller;
/* the sound of silence */
nfiller = device->pwfx->wBitsPerSample == 8 ? 128 : 0;
/* get the position in the primary buffer */
if (DSOUND_PrimaryGetPosition(device, &playpos, &writepos) != 0){
LeaveCriticalSection(&(device->mixlock));
return;
}
TRACE("primary playpos=%d, writepos=%d, clrpos=%d, mixpos=%d, buflen=%d\n",
playpos,writepos,device->playpos,device->mixpos,device->buflen);
assert(device->playpos < device->buflen);
mixplaypos = DSOUND_bufpos_to_mixpos(device, device->playpos);
mixplaypos2 = DSOUND_bufpos_to_mixpos(device, playpos);
/* calc maximum prebuff */
prebuff_max = (device->prebuf * device->fraglen);
if (playpos + prebuff_max >= device->helfrags * device->fraglen)
prebuff_max += device->buflen - device->helfrags * device->fraglen;
/* check how close we are to an underrun. It occurs when the writepos overtakes the mixpos */
prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
writelead = DSOUND_BufPtrDiff(device->buflen, writepos, playpos);
/* check for underrun. underrun occurs when the write position passes the mix position
* also wipe out just-played sound data */
if((prebuff_left > prebuff_max) || (device->state == STATE_STOPPED) || (device->state == STATE_STARTING)){
if (device->state == STATE_STOPPING || device->state == STATE_PLAYING)
WARN("Probable buffer underrun\n");
else TRACE("Buffer starting or buffer underrun\n");
/* recover mixing for all buffers */
recover = TRUE;
/* reset mix position to write position */
device->mixpos = writepos;
ZeroMemory(device->mix_buffer, device->mix_buffer_len);
ZeroMemory(device->buffer, device->buflen);
} else if (playpos < device->playpos) {
buf1 = device->buffer + device->playpos;
buf2 = device->buffer;
size1 = device->buflen - device->playpos;
size2 = playpos;
FillMemory(device->mix_buffer + mixplaypos, device->mix_buffer_len - mixplaypos, 0);
FillMemory(device->mix_buffer, mixplaypos2, 0);
FillMemory(buf1, size1, nfiller);
if (playpos && (!buf2 || !size2))
FIXME("%d: (%d, %d)=>(%d, %d) There should be an additional buffer here!!\n", __LINE__, device->playpos, device->mixpos, playpos, writepos);
FillMemory(buf2, size2, nfiller);
} else {
buf1 = device->buffer + device->playpos;
buf2 = NULL;
size1 = playpos - device->playpos;
size2 = 0;
FillMemory(device->mix_buffer + mixplaypos, mixplaypos2 - mixplaypos, 0);
FillMemory(buf1, size1, nfiller);
}
device->playpos = playpos;
/* find the maximum we can prebuffer from current write position */
maxq = (writelead < prebuff_max) ? (prebuff_max - writelead) : 0;
TRACE("prebuff_left = %d, prebuff_max = %dx%d=%d, writelead=%d\n",
prebuff_left, device->prebuf, device->fraglen, prebuff_max, writelead);
/* do the mixing */
frag = DSOUND_MixToPrimary(device, writepos, maxq, recover, &all_stopped);
if (frag + writepos > device->buflen)
{
DWORD todo = device->buflen - writepos;
device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, todo);
device->normfunction(device->mix_buffer, device->buffer, frag - todo);
}
else
device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, frag);
/* update the mix position, taking wrap-around into account */
device->mixpos = writepos + frag;
device->mixpos %= device->buflen;
/* update prebuff left */
prebuff_left = DSOUND_BufPtrDiff(device->buflen, device->mixpos, playpos);
/* check if have a whole fragment */
if (prebuff_left >= device->fraglen){
/* update the wave queue */
DSOUND_WaveQueue(device, FALSE);
/* buffers are full. start playing if applicable */
if(device->state == STATE_STARTING){
TRACE("started primary buffer\n");
if(DSOUND_PrimaryPlay(device) != DS_OK){
WARN("DSOUND_PrimaryPlay failed\n");
}
else{
/* we are playing now */
device->state = STATE_PLAYING;
}
}
/* buffers are full. start stopping if applicable */
if(device->state == STATE_STOPPED){
TRACE("restarting primary buffer\n");
if(DSOUND_PrimaryPlay(device) != DS_OK){
WARN("DSOUND_PrimaryPlay failed\n");
}
else{
/* start stopping again. as soon as there is no more data, it will stop */
device->state = STATE_STOPPING;
}
}
}
/* if device was stopping, its for sure stopped when all buffers have stopped */
else if((all_stopped == TRUE) && (device->state == STATE_STOPPING)){
TRACE("All buffers have stopped. Stopping primary buffer\n");
device->state = STATE_STOPPED;
/* stop the primary buffer now */
DSOUND_PrimaryStop(device);
}
} else {
DSOUND_WaveQueue(device, TRUE);
/* in the DSSCL_WRITEPRIMARY mode, the app is totally in charge... */
if (device->state == STATE_STARTING) {
if (DSOUND_PrimaryPlay(device) != DS_OK)
WARN("DSOUND_PrimaryPlay failed\n");
else
device->state = STATE_PLAYING;
}
else if (device->state == STATE_STOPPING) {
if (DSOUND_PrimaryStop(device) != DS_OK)
WARN("DSOUND_PrimaryStop failed\n");
else
device->state = STATE_STOPPED;
}
}
LeaveCriticalSection(&(device->mixlock));
/* **** */
}
