The IAudioClient implementation from both Windows and winepulse.drv never sets the event more than once per period, which is usually around 10 ms long. Some codecs produce audio samples shorter than 10 ms, so it is critical that the SAR is able to process more than a sample per period.
This is not currently the case: a new sample is requested only in audio_renderer_render, which is executed (at most) once per period. This results in the SAR not being able to keep up with the audio client, and eventually underrunning.
With this patch the SAR keeps a count of how many frames are currently queued, and a new sample is immediately requested if the internal queue has less than a buffer worth of frames.
This patch fixes audio stuttering problems in the logo videos of Borderlands 3, Deep Rock Galactic and Mutant Year Zero.
Signed-off-by: Giovanni Mascellani gmascellani@codeweavers.com --- v2: Remove some changes that didn't really help the solution and update the description accordingly. v3: Reimplement from scratches, using a different strategy and avoiding manually setting the event. v4: Resent with no changes, because it fails to apply without the first patch. v5: * Zero the count when flushing the queue. * Only request more sample when running. v6: Store a buffer worth of frames in the queue, instead of two periods.
dlls/mf/sar.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+)
diff --git a/dlls/mf/sar.c b/dlls/mf/sar.c index eba822ae0fe..e35531acb71 100644 --- a/dlls/mf/sar.c +++ b/dlls/mf/sar.c @@ -101,6 +101,8 @@ struct audio_renderer HANDLE buffer_ready_event; MFWORKITEM_KEY buffer_ready_key; unsigned int frame_size; + unsigned int queued_frames; + unsigned int max_frames; struct list queue; enum stream_state state; unsigned int flags; @@ -234,6 +236,7 @@ static void audio_renderer_release_audio_client(struct audio_renderer *renderer) { release_pending_object(obj); } + renderer->queued_frames = 0; renderer->buffer_ready_key = 0; if (renderer->audio_client) { @@ -1330,6 +1333,13 @@ static HRESULT WINAPI audio_renderer_stream_GetMediaTypeHandler(IMFStreamSink *i static HRESULT stream_queue_sample(struct audio_renderer *renderer, IMFSample *sample) { struct queued_object *object; + DWORD sample_len, sample_frames; + HRESULT hr; + + if (FAILED(hr = IMFSample_GetTotalLength(sample, &sample_len))) + return hr; + + sample_frames = sample_len / renderer->frame_size;
if (!(object = calloc(1, sizeof(*object)))) return E_OUTOFMEMORY; @@ -1339,6 +1349,7 @@ static HRESULT stream_queue_sample(struct audio_renderer *renderer, IMFSample *s IMFSample_AddRef(object->u.sample.sample);
list_add_tail(&renderer->queue, &object->entry); + renderer->queued_frames += sample_frames;
return S_OK; } @@ -1357,9 +1368,17 @@ static HRESULT WINAPI audio_renderer_stream_ProcessSample(IMFStreamSink *iface, return MF_E_STREAMSINK_REMOVED;
EnterCriticalSection(&renderer->cs); + if (renderer->state == STREAM_STATE_RUNNING) hr = stream_queue_sample(renderer, sample); renderer->flags &= ~SAR_SAMPLE_REQUESTED; + + if (renderer->queued_frames < renderer->max_frames && renderer->state == STREAM_STATE_RUNNING) + { + IMFMediaEventQueue_QueueEventParamVar(renderer->stream_event_queue, MEStreamSinkRequestSample, &GUID_NULL, S_OK, NULL); + renderer->flags |= SAR_SAMPLE_REQUESTED; + } + LeaveCriticalSection(&renderer->cs);
return hr; @@ -1428,6 +1447,7 @@ static HRESULT WINAPI audio_renderer_stream_Flush(IMFStreamSink *iface) release_pending_object(obj); } } + renderer->queued_frames = 0; LeaveCriticalSection(&renderer->cs);
return hr; @@ -1576,6 +1596,12 @@ static HRESULT audio_renderer_create_audio_client(struct audio_renderer *rendere return hr; }
+ if (FAILED(hr = IAudioClient_GetBufferSize(renderer->audio_client, &renderer->max_frames))) + { + WARN("Failed to get buffer size, hr %#x.\n", hr); + return hr; + } + if (SUCCEEDED(hr = MFCreateAsyncResult(NULL, &renderer->render_callback, NULL, &result))) { if (FAILED(hr = MFPutWaitingWorkItem(renderer->buffer_ready_event, 0, result, &renderer->buffer_ready_key))) @@ -1789,6 +1815,7 @@ static void audio_renderer_render(struct audio_renderer *renderer, IMFAsyncResul IAudioRenderClient_ReleaseBuffer(renderer->audio_render_client, dst_frames, 0);
obj->u.sample.frame_offset += dst_frames; + renderer->queued_frames -= dst_frames; }
keep_sample = FAILED(hr) || src_frames > max_frames;
-
Giovanni Mascellani -
Nikolay Sivov