static void CALLBACK DSOUND_capture_timer(UINT timerID, UINT msg, DWORD_PTR user,
DWORD_PTR dw1, DWORD_PTR dw2)
{
DirectSoundCaptureDevice *device = (DirectSoundCaptureDevice*)user;
UINT32 packet_frames, packet_bytes, avail_bytes, skip_bytes = 0;
DWORD flags;
BYTE *buf;
HRESULT hr;
if(!device->ref)
return;
EnterCriticalSection(&device->lock);
if(!device->capture_buffer || device->state == STATE_STOPPED){
LeaveCriticalSection(&device->lock);
return;
}
if(device->state == STATE_STOPPING){
device->state = STATE_STOPPED;
LeaveCriticalSection(&device->lock);
return;
}
if(device->state == STATE_STARTING)
device->state = STATE_CAPTURING;
hr = IAudioCaptureClient_GetBuffer(device->capture, &buf, &packet_frames,
&flags, NULL, NULL);
if(FAILED(hr)){
LeaveCriticalSection(&device->lock);
WARN("GetBuffer failed: %08x\n", hr);
return;
}
packet_bytes = packet_frames * device->pwfx->nBlockAlign;
if(packet_bytes > device->buflen){
TRACE("audio glitch: dsound buffer too small for data\n");
skip_bytes = packet_bytes - device->buflen;
packet_bytes = device->buflen;
}
avail_bytes = device->buflen - device->write_pos_bytes;
if(avail_bytes > packet_bytes)
avail_bytes = packet_bytes;
memcpy(device->buffer + device->write_pos_bytes, buf + skip_bytes, avail_bytes);
capture_CheckNotify(device->capture_buffer, device->write_pos_bytes, avail_bytes);
packet_bytes -= avail_bytes;
if(packet_bytes > 0){
if(device->capture_buffer->flags & DSCBSTART_LOOPING){
memcpy(device->buffer, buf + skip_bytes + avail_bytes, packet_bytes);
capture_CheckNotify(device->capture_buffer, 0, packet_bytes);
}else{
device->state = STATE_STOPPED;
capture_CheckNotify(device->capture_buffer, 0, 0);
}
}
device->write_pos_bytes += avail_bytes + packet_bytes;
device->write_pos_bytes %= device->buflen;
hr = IAudioCaptureClient_ReleaseBuffer(device->capture, packet_frames);
if(FAILED(hr)){
LeaveCriticalSection(&device->lock);
WARN("ReleaseBuffer failed: %08x\n", hr);
return;
}
LeaveCriticalSection(&device->lock);
}
static void test_capture(IAudioClient *ac, HANDLE handle, WAVEFORMATEX *wfx)
{
IAudioCaptureClient *acc;
HRESULT hr;
UINT32 frames = 0;
BYTE *data = NULL;
DWORD flags;
UINT64 devpos, qpcpos;
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;
hr = IAudioCaptureClient_GetNextPacketSize(acc, NULL);
ok(hr == E_POINTER, "IAudioCaptureClient_GetNextPacketSize(NULL) returns %08x\n", hr);
ok(WaitForSingleObject(handle, 2000) == WAIT_OBJECT_0, "Waiting on event handle failed!\n");
/* frames can be 0 if no data is available yet.. */
hr = IAudioCaptureClient_GetNextPacketSize(acc, &frames);
ok(hr == S_OK, "IAudioCaptureClient_GetNextPacketSize returns %08x\n", hr);
data = (BYTE*)(DWORD_PTR)0xdeadbeef;
hr = IAudioCaptureClient_GetBuffer(acc, &data, NULL, NULL, NULL, NULL);
ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(data, NULL, NULL) returns %08x\n", hr);
ok((DWORD_PTR)data == 0xdeadbeef, "data is reset to %p\n", data);
frames = 0xdeadbeef;
hr = IAudioCaptureClient_GetBuffer(acc, NULL, &frames, NULL, NULL, NULL);
ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, &frames, NULL) returns %08x\n", hr);
ok(frames == 0xdeadbeef, "frames is reset to %08x\n", frames);
flags = 0xdeadbeef;
hr = IAudioCaptureClient_GetBuffer(acc, NULL, NULL, &flags, NULL, NULL);
ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(NULL, NULL, &flags) returns %08x\n", hr);
ok(flags == 0xdeadbeef, "flags is reset to %08x\n", flags);
hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, NULL, NULL, NULL);
ok(hr == E_POINTER, "IAudioCaptureClient_GetBuffer(&ata, &frames, NULL) returns %08x\n", hr);
hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &devpos, &qpcpos);
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");
else if (hr == AUDCLNT_S_BUFFER_EMPTY)
ok(!frames, "Amount of frames locked with empty buffer is %u!\n", frames);
else
ok(0, "GetBuffer returned %08x\n", hr);
trace("Device position is at %u, amount of frames locked: %u\n", (DWORD)devpos, frames);
if (frames) {
hr = IAudioCaptureClient_GetBuffer(acc, &data, &frames, &flags, &devpos, &qpcpos);
ok(hr == AUDCLNT_E_OUT_OF_ORDER, "Out of order IAudioCaptureClient_GetBuffer returns %08x\n", hr);
}
hr = IAudioCaptureClient_ReleaseBuffer(acc, frames);
ok(hr == S_OK, "Releasing buffer 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);
}
IUnknown_Release(acc);
}
static GstFlowReturn
gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
GstFlowReturn ret = GST_FLOW_OK;
GstClock *clock;
GstClockTime timestamp, duration = self->period_time;
HRESULT hr;
gint16 *samples = NULL;
guint32 nsamples_read = 0, nsamples;
DWORD flags = 0;
guint64 devpos;
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL)
gst_object_ref (clock);
GST_OBJECT_UNLOCK (self);
if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
GstClockID id;
id = gst_clock_new_single_shot_id (clock, self->next_time);
gst_clock_id_wait (id, NULL);
gst_clock_id_unref (id);
}
do {
hr = IAudioCaptureClient_GetBuffer (self->capture_client,
(BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
}
while (hr == AUDCLNT_S_BUFFER_EMPTY);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
goto beach;
}
if (flags != 0) {
GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
devpos, flags);
}
/* FIXME: Why do we get 1024 sometimes and not a multiple of
* samples_per_buffer? Shouldn't WASAPI provide a DISCONT
* flag if we read too slow?
*/
nsamples = nsamples_read;
g_assert (nsamples >= self->samples_per_buffer);
if (nsamples > self->samples_per_buffer) {
GST_WARNING_OBJECT (self,
"devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
devpos, nsamples, self->samples_per_buffer);
nsamples = self->samples_per_buffer;
}
if (clock == NULL || clock == self->clock) {
timestamp =
gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
} else {
GstClockTime base_time;
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (self)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
if (timestamp > duration)
timestamp -= duration;
else
timestamp = 0;
}
ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
devpos,
nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
if (ret == GST_FLOW_OK) {
guint i;
gint16 *dst;
GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
GST_BUFFER_DURATION (*buf) = duration;
dst = (gint16 *) GST_BUFFER_DATA (*buf);
for (i = 0; i < nsamples; i++) {
*dst = *samples;
samples += 2;
dst++;
}
}
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
goto beach;
}
beach:
if (clock != NULL)
gst_object_unref (clock);
return ret;
}
static HRESULT DSOUND_capture_data(DirectSoundCaptureDevice *device)
{
HRESULT hr;
UINT32 packet_frames, packet_bytes, avail_bytes, skip_bytes = 0;
DWORD flags;
BYTE *buf;
if(!device->capture_buffer || device->state == STATE_STOPPED)
return S_FALSE;
if(device->state == STATE_STOPPING){
device->state = STATE_STOPPED;
return S_FALSE;
}
if(device->state == STATE_STARTING)
device->state = STATE_CAPTURING;
hr = IAudioCaptureClient_GetBuffer(device->capture, &buf, &packet_frames,
&flags, NULL, NULL);
if(FAILED(hr)){
WARN("GetBuffer failed: %08x\n", hr);
return hr;
}
packet_bytes = packet_frames * device->pwfx->nBlockAlign;
if(packet_bytes > device->buflen){
TRACE("audio glitch: dsound buffer too small for data\n");
skip_bytes = packet_bytes - device->buflen;
packet_bytes = device->buflen;
}
avail_bytes = device->buflen - device->write_pos_bytes;
if(avail_bytes > packet_bytes)
avail_bytes = packet_bytes;
memcpy(device->buffer + device->write_pos_bytes, buf + skip_bytes, avail_bytes);
capture_CheckNotify(device->capture_buffer, device->write_pos_bytes, avail_bytes);
packet_bytes -= avail_bytes;
if(packet_bytes > 0){
if(device->capture_buffer->flags & DSCBSTART_LOOPING){
memcpy(device->buffer, buf + skip_bytes + avail_bytes, packet_bytes);
capture_CheckNotify(device->capture_buffer, 0, packet_bytes);
}else{
device->state = STATE_STOPPED;
capture_CheckNotify(device->capture_buffer, 0, 0);
}
}
device->write_pos_bytes += avail_bytes + packet_bytes;
device->write_pos_bytes %= device->buflen;
hr = IAudioCaptureClient_ReleaseBuffer(device->capture, packet_frames);
if(FAILED(hr)){
WARN("ReleaseBuffer failed: %08x\n", hr);
return hr;
}
return S_OK;
}
JNIEXPORT jint JNICALL
Java_org_jitsi_impl_neomedia_jmfext_media_protocol_wasapi_WASAPI_IAudioCaptureClient_1Read
(JNIEnv *env, jclass clazz, jlong thiz, jbyteArray data, jint offset,
jint length, jint srcSampleSize, jint srcChannels, jint dstSampleSize,
jint dstChannels)
{
HRESULT hr;
IAudioCaptureClient *iAudioCaptureClient
= (IAudioCaptureClient *) (intptr_t) thiz;
BYTE *pData;
UINT32 numFramesToRead;
DWORD dwFlags;
jint read;
hr
= IAudioCaptureClient_GetBuffer(
iAudioCaptureClient,
&pData,
&numFramesToRead,
&dwFlags,
NULL,
NULL);
if (SUCCEEDED(hr))
{
UINT32 numFramesRead;
jint dstFrameSize = dstSampleSize * dstChannels;
if ((numFramesToRead == 0) || (hr == AUDCLNT_S_BUFFER_EMPTY))
numFramesRead = 0;
else
{
if (length < numFramesToRead * dstFrameSize)
{
numFramesRead = 0;
WASAPI_throwNewHResultException(
env,
E_INVALIDARG,
__func__, __LINE__);
}
else
{
jbyte *data_
= (*env)->GetPrimitiveArrayCritical(env, data, NULL);
if (data_)
{
numFramesRead
= WASAPI_audiocopy(
pData, srcSampleSize, srcChannels,
data_ + offset, dstSampleSize, dstChannels,
numFramesToRead);
(*env)->ReleasePrimitiveArrayCritical(env, data, data_, 0);
}
else
numFramesRead = 0; /* An OutOfMemoryError has been thrown. */
}
}
hr
= IAudioCaptureClient_ReleaseBuffer(
iAudioCaptureClient,
numFramesRead);
read = numFramesRead * dstFrameSize;
if (FAILED(hr))
WASAPI_throwNewHResultException(env, hr, __func__, __LINE__);
}
else
{
read = 0;
WASAPI_throwNewHResultException(env, hr, __func__, __LINE__);
}
return read;
}
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);
}
