Don't start WASAPI render thread until setup completes.

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 8ba88217ecc44a2454db1d6341376d4d5b14db9e..a10e67a46cbf2f9b541e68a24a163c5a2d691a24 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -71,9 +71,12 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
                                                  ERole device_role)
     : creating_thread_id_(base::PlatformThread::CurrentId()),
       manager_(manager),
+      format_(),
       opened_(false),
       audio_parameters_are_valid_(false),
       volume_(1.0),
+      packet_size_frames_(0),
+      packet_size_bytes_(0),
       endpoint_buffer_size_frames_(0),
       device_id_(device_id),
       device_role_(device_role),
@@ -130,11 +133,11 @@ WASAPIAudioOutputStream::WASAPIAudioOutputStream(AudioManagerWin* manager,
   // get from the audio endpoint device in each render event.
   packet_size_frames_ = params.frames_per_buffer();
   packet_size_bytes_ = params.GetBytesPerBuffer();
-  packet_size_ms_ = (1000.0 * packet_size_frames_) / params.sample_rate();
   VLOG(1) << "Number of bytes per audio frame  : " << format->nBlockAlign;
   VLOG(1) << "Number of audio frames per packet: " << packet_size_frames_;
   VLOG(1) << "Number of bytes per packet       : " << packet_size_bytes_;
-  VLOG(1) << "Number of milliseconds per packet: " << packet_size_ms_;
+  VLOG(1) << "Number of milliseconds per packet: "
+          << params.GetBufferDuration().InMillisecondsF();
 
   // All events are auto-reset events and non-signaled initially.
 
@@ -234,6 +237,13 @@ bool WASAPIAudioOutputStream::Open() {
   audio_client_ = audio_client;
   audio_render_client_ = audio_render_client;
 
+  hr = audio_client_->GetService(__uuidof(IAudioClock),
+                                 audio_clock_.ReceiveVoid());
+  if (FAILED(hr)) {
+    LOG(ERROR) << "Failed to get IAudioClock service.";
+    return false;
+  }
+
   opened_ = true;
   return true;
 }
@@ -251,6 +261,17 @@ void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) {
 
   source_ = callback;
 
+  // Ensure that the endpoint buffer is prepared with silence.
+  if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) {
+    if (!CoreAudioUtil::FillRenderEndpointBufferWithSilence(
+             audio_client_, audio_render_client_)) {
+      LOG(ERROR) << "Failed to prepare endpoint buffers with silence.";
+      callback->OnError(this);
+      return;
+    }
+  }
+  num_written_frames_ = endpoint_buffer_size_frames_;
+
   // Create and start the thread that will drive the rendering by waiting for
   // render events.
   render_thread_.reset(
@@ -263,18 +284,6 @@ void WASAPIAudioOutputStream::Start(AudioSourceCallback* callback) {
     return;
   }
 
-  // Ensure that the endpoint buffer is prepared with silence.
-  if (share_mode_ == AUDCLNT_SHAREMODE_SHARED) {
-    if (!CoreAudioUtil::FillRenderEndpointBufferWithSilence(
-             audio_client_, audio_render_client_)) {
-      LOG(ERROR) << "Failed to prepare endpoint buffers with silence.";
-      StopThread();
-      callback->OnError(this);
-      return;
-    }
-  }
-  num_written_frames_ = endpoint_buffer_size_frames_;
-
   // Start streaming data between the endpoint buffer and the audio engine.
   HRESULT hr = audio_client_->Start();
   if (FAILED(hr)) {
@@ -377,17 +386,9 @@ void WASAPIAudioOutputStream::Run() {
                           audio_samples_render_event_ };
   UINT64 device_frequency = 0;
 
-  // The IAudioClock interface enables us to monitor a stream's data
-  // rate and the current position in the stream. Allocate it before we
-  // start spinning.
-  ScopedComPtr<IAudioClock> audio_clock;
-  hr = audio_client_->GetService(__uuidof(IAudioClock),
-                                 audio_clock.ReceiveVoid());
-  if (SUCCEEDED(hr)) {
-    // 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);
-  }
+  // 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);
   error = FAILED(hr);
   PLOG_IF(ERROR, error) << "Failed to acquire IAudioClock interface: "
                         << std::hex << hr;
@@ -408,7 +409,7 @@ void WASAPIAudioOutputStream::Run() {
         break;
       case WAIT_OBJECT_0 + 1:
         // |audio_samples_render_event_| has been set.
-        error = !RenderAudioFromSource(audio_clock, device_frequency);
+        error = !RenderAudioFromSource(device_frequency);
         break;
       default:
         error = true;
@@ -430,8 +431,7 @@ void WASAPIAudioOutputStream::Run() {
   }
 }
 
-bool WASAPIAudioOutputStream::RenderAudioFromSource(
-    IAudioClock* audio_clock, UINT64 device_frequency) {
+bool WASAPIAudioOutputStream::RenderAudioFromSource(UINT64 device_frequency) {
   TRACE_EVENT0("audio", "RenderAudioFromSource");
 
   HRESULT hr = S_FALSE;
@@ -503,7 +503,7 @@ bool WASAPIAudioOutputStream::RenderAudioFromSource(
     // unit at the render side.
     UINT64 position = 0;
     int 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.