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..cb24f43113b8f6a386716751ae28e32a60b78c39 100644
--- a/media/audio/win/audio_low_latency_output_win.cc
+++ b/media/audio/win/audio_low_latency_output_win.cc
@@ -234,6 +234,13 @@ 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 +258,17 @@ 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 +281,6 @@ 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 +383,9 @@ 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);

[DaleCurtis, 2013/11/27 02:45:08]

I left this here, but if it's truly constant we could query it during Open()
instead.

[henrika (OOO until Aug 14), 2013/11/27 08:24:39]

Should be OK to use in Open. Can't see any cons by having it here actually.

   error = FAILED(hr);
   PLOG_IF(ERROR, error) << "Failed to acquire IAudioClock interface: "
                         << std::hex << hr;
@@ -408,7 +406,7 @@ 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 +428,7 @@ 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 +500,7 @@ 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.