Fix VirtualAudioInputStream callback timing issues.

content/renderer/media/audio_renderer_mixer_manager.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 "content/renderer/media/audio_renderer_mixer_manager.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/renderer_audio_output_device.h"
@@ skipping 34 matching lines @@
     const media::AudioParameters& params) {
   const MixerKey key(source_render_view_id, params);
   base::AutoLock auto_lock(mixers_lock_);
 
   AudioRendererMixerMap::iterator it = mixers_.find(key);
   if (it != mixers_.end()) {
     it->second.ref_count++;
     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.
   if (!output_params.IsValid())
     output_params = params;
 
   media::AudioRendererMixer* mixer;
   if (sink_for_testing_) {
     mixer = new media::AudioRendererMixer(
         params, output_params, sink_for_testing_);
@@ skipping 20 matching lines @@
 
   // Only remove the mixer if AudioRendererMixerManager is the last owner.
   it->second.ref_count--;
   if (it->second.ref_count == 0) {
     delete it->second.mixer;
     mixers_.erase(it);
   }
 }
 
 }  // namespace content