Switch to STA mode for audio thread and WASAPI I/O streams.

media/audio/win/audio_low_latency_output_win.cc

Index: media/audio/win/audio_low_latency_output_win.cc
diff --git a/media/audio/win/audio_low_latency_output_win.cc b/media/audio/win/audio_low_latency_output_win.cc
index f7b31a3c00a09d75376a190fa9df926f9b32b6db..a9d637c3edc7e7c705011c4e88f95885866b4255 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -249,6 +249,11 @@ void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) {
   }
   num_written_frames_ = endpoint_buffer_size_frames_;
 
+  if (!MarshalComPointers()) {
+    callback->OnError(this);
+    return;
+  }
+
   // Create and start the thread that will drive the rendering by waiting for
   // render events.
   render_thread_.reset(
@@ -333,7 +338,7 @@ void WASAPIAudioOutputStream::GetVolume(double* volume) {
 }
 
 void WASAPIAudioOutputStream::Run() {
-  ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
+  ScopedCOMInitializer com_init;
 
   // Increase the thread priority.
   render_thread_->SetThreadPriority(base::ThreadPriority::REALTIME_AUDIO);
@@ -352,6 +357,12 @@ void WASAPIAudioOutputStream::Run() {
     LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ").";
   }
 
+  // Retrieve COM pointers from the main thread.
+  ScopedComPtr<IAudioClient> audio_client;
+  ScopedComPtr<IAudioRenderClient> audio_render_client;
+  ScopedComPtr<IAudioClock> audio_clock;
+  UnmarshalComPointers(&audio_client, &audio_render_client, &audio_clock);
+
   HRESULT hr = S_FALSE;
 
   bool playing = true;
@@ -362,7 +373,7 @@ void WASAPIAudioOutputStream::Run() {
 
   // The device frequency is the frequency generated by the hardware clock in
   // the audio device. The GetFrequency() method reports a constant frequency.
-  hr = audio_clock_->GetFrequency(&device_frequency);
+  hr = audio_clock->GetFrequency(&device_frequency);
   error = FAILED(hr);
   PLOG_IF(ERROR, error) << "Failed to acquire IAudioClock interface: "
                         << std::hex << hr;
@@ -383,7 +394,9 @@ void WASAPIAudioOutputStream::Run() {
         break;
       case WAIT_OBJECT_0 + 1:
         // |audio_samples_render_event_| has been set.
-        error = !RenderAudioFromSource(device_frequency);
+        error = !RenderAudioFromSource(device_frequency, audio_client.get(),
+                                       audio_render_client.get(),
+                                       audio_clock.get());
         break;
       default:
         error = true;
@@ -391,11 +404,11 @@ void WASAPIAudioOutputStream::Run() {
     }
   }
 
-  if (playing && error) {
+  if (playing && error && audio_client) {
     // Stop audio rendering since something has gone wrong in our main thread
     // loop. Note that, we are still in a "started" state, hence a Stop() call
     // is required to join the thread properly.
-    audio_client_->Stop();
+    audio_client->Stop();
     PLOG(ERROR) << "WASAPI rendering failed.";
   }
 
@@ -405,7 +418,11 @@ void WASAPIAudioOutputStream::Run() {
   }
 }
 
-bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
+bool WASAPIAudioOutputStream::RenderAudioFromSource(
+    UINT64 device_frequency,
+    IAudioClient* audio_client,
+    IAudioRenderClient* audio_render_client,
+    IAudioClock* audio_clock) {
   TRACE_EVENT0("audio", "RenderAudioFromSource");
 
   HRESULT hr = S_FALSE;
@@ -420,7 +437,7 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
   if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) {
     // Get the padding value which represents the amount of rendering
     // data that is queued up to play in the endpoint buffer.
-    hr = audio_client_->GetCurrentPadding(&num_queued_frames);
+    hr = audio_client->GetCurrentPadding(&num_queued_frames);
     num_available_frames =
         endpoint_buffer_size_frames_ - num_queued_frames;
     if (FAILED(hr)) {
@@ -462,8 +479,7 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
   for (size_t n = 0; n < num_packets; ++n) {
     // Grab all available space in the rendering endpoint buffer
     // into which the client can write a data packet.
-    hr = audio_render_client_->GetBuffer(packet_size_frames_,
-                                         &audio_data);
+    hr = audio_render_client->GetBuffer(packet_size_frames_, &audio_data);
     if (FAILED(hr)) {
       DLOG(ERROR) << "Failed to use rendering audio buffer: "
                  << std::hex << hr;
@@ -477,7 +493,7 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
     // unit at the render side.
     UINT64 position = 0;
     uint32 audio_delay_bytes = 0;
-    hr = audio_clock_->GetPosition(&position, NULL);
+    hr = audio_clock->GetPosition(&position, NULL);
     if (SUCCEEDED(hr)) {
       // Stream position of the sample that is currently playing
       // through the speaker.
@@ -517,7 +533,7 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
     // Render silence if we were not able to fill up the buffer totally.
     DWORD flags = (num_filled_bytes < packet_size_bytes_) ?
         AUDCLNT_BUFFERFLAGS_SILENT : 0;
-    audio_render_client_->ReleaseBuffer(packet_size_frames_, flags);
+    audio_render_client->ReleaseBuffer(packet_size_frames_, flags);
 
     num_written_frames_ += packet_size_frames_;
   }
@@ -622,4 +638,73 @@ void WASAPIAudioOutputStream::StopThread() {
   source_ = NULL;
 }
 
+bool WASAPIAudioOutputStream::MarshalComPointers() {
+  DCHECK_EQ(creating_thread_id_, base::PlatformThread::CurrentId());
+  DCHECK(!com_stream_);
+
+  ScopedComPtr<IStream> com_stream;
+  HRESULT hr = CreateStreamOnHGlobal(NULL, TRUE, com_stream.Receive());
+  if (FAILED(hr)) {
+    DLOG(ERROR) << "Failed to create stream for marshaling COM pointers.";
+    return false;
+  }
+
+  hr = CoMarshalInterface(com_stream.get(), __uuidof(IAudioClient),
+                          audio_client_.get(), MSHCTX_INPROC, NULL,
+                          MSHLFLAGS_NORMAL);
+  if (FAILED(hr)) {
+    DLOG(ERROR) << "Marshal failed for IAudioClient: " << std::hex << hr;
+    return false;
+  }
+
+  hr = CoMarshalInterface(com_stream.get(), __uuidof(IAudioRenderClient),
+                          audio_render_client_.get(), MSHCTX_INPROC, NULL,
+                          MSHLFLAGS_NORMAL);
+  if (FAILED(hr)) {
+    DLOG(ERROR) << "Marshal failed for IAudioRenderClient: " << std::hex << hr;
+    return false;
+  }
+
+  hr = CoMarshalInterface(com_stream.get(), __uuidof(IAudioClock),
+                          audio_clock_.get(), MSHCTX_INPROC, NULL,
+                          MSHLFLAGS_NORMAL);
+  if (FAILED(hr)) {
+    DLOG(ERROR) << "Marshal failed for IAudioClock: " << std::hex << hr;
+    return false;
+  }
+
+  LARGE_INTEGER pos = {0};
+  hr = com_stream->Seek(pos, STREAM_SEEK_SET, NULL);
+  if (FAILED(hr)) {
+    DLOG(ERROR) << "Failed to seek IStream for marshaling: " << std::hex << hr;
+    return false;
+  }
+
+  com_stream_ = com_stream.Pass();
+  return true;
+}
+
+void WASAPIAudioOutputStream::UnmarshalComPointers(
+    ScopedComPtr<IAudioClient>* audio_client,
+    ScopedComPtr<IAudioRenderClient>* audio_render_client,
+    ScopedComPtr<IAudioClock>* audio_clock) {
+  DCHECK_EQ(render_thread_->tid(), base::PlatformThread::CurrentId());
+
+  DCHECK(com_stream_);
+  ScopedComPtr<IStream> com_stream;
+  com_stream = com_stream_.Pass();
+
+  HRESULT hr = CoUnmarshalInterface(com_stream.get(), __uuidof(IAudioClient),
+                                    audio_client->ReceiveVoid());
+  CHECK(SUCCEEDED(hr));
+
+  hr = CoUnmarshalInterface(com_stream.get(), __uuidof(IAudioRenderClient),
+                            audio_render_client->ReceiveVoid());
+  CHECK(SUCCEEDED(hr));
+
+  hr = CoUnmarshalInterface(com_stream.get(), __uuidof(IAudioClock),
+                            audio_clock->ReceiveVoid());
+  CHECK(SUCCEEDED(hr));
+}
+
 }  // namespace media