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

media/audio/win/audio_low_latency_output_win.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/audio/win/audio_low_latency_output_win.h"
 
 #include <Functiondiscoverykeys_devpkey.h>
 
 #include "base/command_line.h"
 #include "base/logging.h"
@@ skipping 231 matching lines @@
   if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) {
     if (!CoreAudioUtil::FillRenderEndpointBufferWithSilence(
             audio_client_.get(), audio_render_client_.get())) {
       LOG(ERROR) << "Failed to prepare endpoint buffers with silence.";
       callback->OnError(this);
       return;
     }
   }
   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(
       new base::DelegateSimpleThread(this, "wasapi_render_thread"));
   render_thread_->Start();
   if (!render_thread_->HasBeenStarted()) {
     LOG(ERROR) << "Failed to start WASAPI render thread.";
     StopThread();
     callback->OnError(this);
     return;
@@ skipping 64 matching lines @@
   }
   volume_ = volume_float;
 }
 
 void WASAPIAudioOutputStream::GetVolume(double* volume) {
   DVLOG(1) << "GetVolume()";
   *volume = static_cast<double>(volume_);
 }
 
 void WASAPIAudioOutputStream::Run() {
-  ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
+  ScopedCOMInitializer com_init;
 
   // Increase the thread priority.
   render_thread_->SetThreadPriority(base::ThreadPriority::REALTIME_AUDIO);
 
   // Enable MMCSS to ensure that this thread receives prioritized access to
   // CPU resources.
   DWORD task_index = 0;
   HANDLE mm_task = avrt::AvSetMmThreadCharacteristics(L"Pro Audio",
                                                       &task_index);
   bool mmcss_is_ok =
       (mm_task && avrt::AvSetMmThreadPriority(mm_task, AVRT_PRIORITY_CRITICAL));
   if (!mmcss_is_ok) {
     // Failed to enable MMCSS on this thread. It is not fatal but can lead
     // to reduced QoS at high load.
     DWORD err = GetLastError();
     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;
   bool error = false;
   HANDLE wait_array[] = { stop_render_event_.Get(),
                           audio_samples_render_event_.Get() };
   UINT64 device_frequency = 0;
 
   // 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;
 
   // Keep rendering audio until the stop event or the stream-switch event
   // is signaled. An error event can also break the main thread loop.
   while (playing && !error) {
     // Wait for a close-down event, stream-switch event or a new render event.
     DWORD wait_result = WaitForMultipleObjects(arraysize(wait_array),
                                                wait_array,
                                                FALSE,
                                                INFINITE);
 
     switch (wait_result) {
       case WAIT_OBJECT_0 + 0:
         // |stop_render_event_| has been set.
         playing = false;
         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;
         break;
     }
   }
 
-  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.";
   }
 
   // Disable MMCSS.
   if (mm_task && !avrt::AvRevertMmThreadCharacteristics(mm_task)) {
     PLOG(WARNING) << "Failed to disable MMCSS";
   }
 }
 
-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;
   UINT32 num_queued_frames = 0;
   uint8* audio_data = NULL;
 
   // Contains how much new data we can write to the buffer without
   // the risk of overwriting previously written data that the audio
   // engine has not yet read from the buffer.
   size_t num_available_frames = 0;
 
   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)) {
       DLOG(ERROR) << "Failed to retrieve amount of available space: "
                   << std::hex << hr;
       return false;
     }
   } else {
     // While the stream is running, the system alternately sends one
     // buffer or the other to the client. This form of double buffering
@@ skipping 21 matching lines @@
   // fill up the available area in the endpoint buffer.
   // |num_packets| will always be one for exclusive-mode streams and
   // will be one in most cases for shared mode streams as well.
   // However, we have found that two packets can sometimes be
   // required.
   size_t num_packets = (num_available_frames / packet_size_frames_);
 
   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;
       return false;
     }
 
     // Derive the audio delay which corresponds to the delay between
     // a render event and the time when the first audio sample in a
     // packet is played out through the speaker. This delay value
     // can typically be utilized by an acoustic echo-control (AEC)
     // 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.
       double pos_sample_playing_frames = format_.Format.nSamplesPerSec *
           (static_cast<double>(position) / device_frequency);
 
       // Stream position of the last sample written to the endpoint
       // buffer. Note that, the packet we are about to receive in
       // the upcoming callback is also included.
       size_t pos_last_sample_written_frames =
@@ skipping 19 matching lines @@
     const int bytes_per_sample = format_.Format.wBitsPerSample >> 3;
     audio_bus_->Scale(volume_);
     audio_bus_->ToInterleaved(
         frames_filled, bytes_per_sample, audio_data);
 
 
     // Release the buffer space acquired in the GetBuffer() call.
     // 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_;
   }
 
   return true;
 }
 
 HRESULT WASAPIAudioOutputStream::ExclusiveModeInitialization(
     IAudioClient* client, HANDLE event_handle, uint32* endpoint_buffer_size) {
   DCHECK_EQ(share_mode_, AUDCLNT_SHAREMODE_EXCLUSIVE);
@@ skipping 84 matching lines @@
     render_thread_.reset();
 
     // Ensure that we don't quit the main thread loop immediately next
     // time Start() is called.
     ResetEvent(stop_render_event_.Get());
   }
 
   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