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

media/audio/win/audio_low_latency_input_win.cc

Index: media/audio/win/audio_low_latency_input_win.cc
diff --git a/media/audio/win/audio_low_latency_input_win.cc b/media/audio/win/audio_low_latency_input_win.cc
index 72d1d725f5bede99fcea3668212ba63d61c4e020..7003c9d68e65757d039cd16f9a238044c7e0d0d8 100644
--- a/media/audio/win/audio_low_latency_input_win.cc
+++ b/media/audio/win/audio_low_latency_input_win.cc
@@ -158,6 +158,11 @@ void WASAPIAudioInputStream::Start(AudioInputCallback* callback) {
   // using SetAutomaticGainControl().
   StartAgc();
 
+  if (!MarshalComPointers()) {
+    HandleError(S_FALSE);
+    return;
+  }
+
   // Create and start the thread that will drive the capturing by waiting for
   // capture events.
   capture_thread_ =
@@ -172,6 +177,8 @@ void WASAPIAudioInputStream::Start(AudioInputCallback* callback) {
     hr = audio_render_client_for_loopback_->Start();
 
   started_ = SUCCEEDED(hr);
+  if (!started_)
+    HandleError(hr);
 }
 
 void WASAPIAudioInputStream::Stop() {
@@ -350,7 +357,7 @@ HRESULT WASAPIAudioInputStream::GetMixFormat(const std::string& device_id,
 }
 
 void WASAPIAudioInputStream::Run() {
-  ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
+  ScopedCOMInitializer com_init;
 
   // Increase the thread priority.
   capture_thread_->SetThreadPriority(base::ThreadPriority::REALTIME_AUDIO);
@@ -369,6 +376,11 @@ void WASAPIAudioInputStream::Run() {
     LOG(WARNING) << "Failed to enable MMCSS (error code=" << err << ").";
   }
 
+  // Retrieve COM pointers from the main thread.
+  ScopedComPtr<IAudioCaptureClient> thread_audio_capture_client;

[tommi (sloooow) - chröme, 2015/04/22 18:10:08]

nit: audio_capture_client

[DaleCurtis, 2015/04/22 18:31:47]

Done.

+
+  bool error = !UnmarshalComPointers(&thread_audio_capture_client);
+
   // Allocate a buffer with a size that enables us to take care of cases like:
   // 1) The recorded buffer size is smaller, or does not match exactly with,
   //    the selected packet size used in each callback.
@@ -382,8 +394,7 @@ void WASAPIAudioInputStream::Run() {
 
   LARGE_INTEGER now_count;
   bool recording = true;
-  bool error = false;
-  double volume = GetVolume();
+  double volume = 0;
   HANDLE wait_array[2] =
       { stop_capture_event_.Get(), audio_samples_ready_event_.Get() };
 
@@ -412,11 +423,9 @@ void WASAPIAudioInputStream::Run() {
           // Retrieve the amount of data in the capture endpoint buffer,
           // replace it with silence if required, create callbacks for each
           // packet and store non-delivered data for the next event.
-          hr = audio_capture_client_->GetBuffer(&data_ptr,
-                                                &num_frames_to_read,
-                                                &flags,
-                                                &device_position,
-                                                &first_audio_frame_timestamp);
+          hr = thread_audio_capture_client->GetBuffer(
+              &data_ptr, &num_frames_to_read, &flags, &device_position,
+              &first_audio_frame_timestamp);
           if (FAILED(hr)) {
             DLOG(ERROR) << "Failed to get data from the capture buffer";
             continue;
@@ -438,7 +447,7 @@ void WASAPIAudioInputStream::Run() {
             buffer_frame_index += num_frames_to_read;
           }
 
-          hr = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
+          hr = thread_audio_capture_client->ReleaseBuffer(num_frames_to_read);
           DLOG_IF(ERROR, FAILED(hr)) << "Failed to release capture buffer";
 
           // Derive a delay estimate for the captured audio packet.
@@ -676,8 +685,11 @@ HRESULT WASAPIAudioInputStream::InitializeAudioEngine() {
       &format_,
       (effects_ & AudioParameters::DUCKING) ? &kCommunicationsSessionId : NULL);
 
-  if (FAILED(hr))
+  if (FAILED(hr)) {
+    PLOG(ERROR) << "Failed to initalize IAudioClient: " << std::hex << hr
+                << " : ";
     return hr;
+  }
 
   // Retrieve the length of the endpoint buffer shared between the client
   // and the audio engine. The buffer length determines the maximum amount
@@ -768,4 +780,28 @@ HRESULT WASAPIAudioInputStream::InitializeAudioEngine() {
   return hr;
 }
 
+bool WASAPIAudioInputStream::MarshalComPointers() {
+  DCHECK(CalledOnValidThread());
+  DCHECK(!com_stream_);
+  HRESULT hr = CoMarshalInterThreadInterfaceInStream(
+      __uuidof(IAudioCaptureClient), audio_capture_client_.get(),
+      com_stream_.Receive());
+  if (FAILED(hr))
+    DLOG(ERROR) << "Marshal failed for IAudioCaptureClient: " << std::hex << hr;
+  DCHECK_EQ(SUCCEEDED(hr), !!com_stream_);
+  return SUCCEEDED(hr);
+}
+
+bool WASAPIAudioInputStream::UnmarshalComPointers(
+    ScopedComPtr<IAudioCaptureClient>* audio_capture_client) {
+  DCHECK_EQ(capture_thread_->tid(), base::PlatformThread::CurrentId());
+  DCHECK(com_stream_);
+  HRESULT hr = CoGetInterfaceAndReleaseStream(
+      com_stream_.Detach(), __uuidof(IAudioCaptureClient),
+      audio_capture_client->ReceiveVoid());
+  if (FAILED(hr))

[tommi (sloooow) - chröme, 2015/04/22 18:10:08]

DCHECK(SUCCEEDED(hr)) ?

I think this should only fail on non-COM threads and that would probably be a
programmer error.

[DaleCurtis, 2015/04/22 18:31:47]

Done.

+    DLOG(ERROR) << "Unmarshal failed IAudioCaptureClient: " << std::hex << hr;
+  return SUCCEEDED(hr);
+}
+
 }  // namespace media