mf/sar: Query for current padding before requesting sample buffer.

According to both MSDN and our impementation, GetBufferSize returns the
size of the buffer, but it doesn't guarantee that all of it is available.
In order to know how much of it is available, the caller must also call
GetCurrentPadding and subtract that number to the buffer size. Failing
to do so might result in GetBuffer returning an error.

Signed-off-by: Giovanni Mascellani <gmascellani@codeweavers.com>
Signed-off-by: Nikolay Sivov <nsivov@codeweavers.com>
Signed-off-by: Alexandre Julliard <julliard@winehq.org>
This commit is contained in:
Giovanni Mascellani 2021-06-21 17:52:53 +02:00 committed by Alexandre Julliard
parent 95817b6386
commit da171b8f90

View File

@ -1751,7 +1751,7 @@ static HRESULT WINAPI audio_renderer_render_callback_GetParameters(IMFAsyncCallb
static void audio_renderer_render(struct audio_renderer *renderer, IMFAsyncResult *result)
{
unsigned int src_frames, dst_frames, max_frames, src_len;
unsigned int src_frames, dst_frames, max_frames, pad_frames, src_len;
struct queued_object *obj, *obj2;
BOOL keep_sample = FALSE;
IMFMediaBuffer *buffer;
@ -1775,20 +1775,24 @@ static void audio_renderer_render(struct audio_renderer *renderer, IMFAsyncResul
{
if (SUCCEEDED(IAudioClient_GetBufferSize(renderer->audio_client, &max_frames)))
{
src_frames -= obj->u.sample.frame_offset;
dst_frames = min(src_frames, max_frames);
if (SUCCEEDED(hr = IAudioRenderClient_GetBuffer(renderer->audio_render_client, dst_frames, &dst)))
if (SUCCEEDED(IAudioClient_GetCurrentPadding(renderer->audio_client, &pad_frames)))
{
memcpy(dst, src + obj->u.sample.frame_offset * renderer->frame_size,
dst_frames * renderer->frame_size);
max_frames -= pad_frames;
src_frames -= obj->u.sample.frame_offset;
dst_frames = min(src_frames, max_frames);
IAudioRenderClient_ReleaseBuffer(renderer->audio_render_client, dst_frames, 0);
if (SUCCEEDED(hr = IAudioRenderClient_GetBuffer(renderer->audio_render_client, dst_frames, &dst)))
{
memcpy(dst, src + obj->u.sample.frame_offset * renderer->frame_size,
dst_frames * renderer->frame_size);
obj->u.sample.frame_offset += dst_frames;
IAudioRenderClient_ReleaseBuffer(renderer->audio_render_client, dst_frames, 0);
obj->u.sample.frame_offset += dst_frames;
}
keep_sample = FAILED(hr) || src_frames > max_frames;
}
keep_sample = FAILED(hr) || src_frames > max_frames;
}
}
IMFMediaBuffer_Unlock(buffer);