Fix VirtualAudioInputStream callback timing issues.

content/renderer/media/audio_renderer_mixer_manager.cc

Index: content/renderer/media/audio_renderer_mixer_manager.cc
diff --git a/content/renderer/media/audio_renderer_mixer_manager.cc b/content/renderer/media/audio_renderer_mixer_manager.cc
index 05782d231fb618d66e8e824388f045e8866185db..339f6aea60bc3fc18d6c2c1b6689466cc3553aa3 100644
--- a/content/renderer/media/audio_renderer_mixer_manager.cc
+++ b/content/renderer/media/audio_renderer_mixer_manager.cc
@@ -52,12 +52,21 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
     return it->second.mixer;
   }
 
+  int buffer_size = hardware_buffer_size_;
+#if defined(OS_WIN) || defined(OS_MACOSX)

[miu, 2013/01/15 23:28:35]

Remove #if, since this would seem to apply to all platforms.     Or, am I
missing something?

[justinlin, 2013/01/15 23:52:45]

Renderer side mixing is only turned on for OSX/Win right now, so you won't see
any of these problems on Linux yet.

[miu, 2013/01/16 03:43:03]

It's a flag for multiple platforms, enabled by default for mac/win, and disabled
(but can be enabled) in linux.

Reference:
http://code.google.com/searchframe#OAMlx_jo-ck/src/content/renderer/render_vi...

Perhaps Dale could chime in on the right approach here?

[justinlin, 2013/01/16 20:15:23]

I added a TODO to do this if we don't get to renderer side device changes + some
mirroring code changes first (we would be able to update params on renderer side
too in that case, and I think we can avoid resampling altogether).

+  // Force at least a 1024 buffer size until we have renderer-side device
+  // changes. WebRTC audio mirroring will use the input device's buffer size to

[DaleCurtis, 2013/01/15 23:30:03]

WebRTC doesn't use this code?

[justinlin, 2013/01/15 23:52:45]

Done. We'd typically use audio mirroring with webrtc, but right it's not
directly tied to that.

+  // drive mirroring, so we need the output device's buffer size to be big
+  // enough to avoid garbled audio.
+  buffer_size = std::max(hardware_buffer_size_, 1024);
+#endif
+
   // Create output parameters based on the audio hardware configuration for
   // passing on to the output sink.  Force to 16-bit output for now since we
   // know that works well for WebAudio and WebRTC.
   media::AudioParameters output_params(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY, params.channel_layout(),
-      hardware_sample_rate_, 16, hardware_buffer_size_);
+      hardware_sample_rate_, 16, buffer_size);
 
   // If we've created invalid output parameters, simply pass on the input params
   // and let the browser side handle automatic fallback.