Increase the buffer size in AudioRendererImpl to fix muted playback rate

content/browser/renderer_host/media/audio_renderer_host.cc

Index: content/browser/renderer_host/media/audio_renderer_host.cc
diff --git a/content/browser/renderer_host/media/audio_renderer_host.cc b/content/browser/renderer_host/media/audio_renderer_host.cc
index d3dbb75526c1f715174f19541edea2729c8c16fe..ac5051ef1a9dae0216a0762ab116d033f0daca2b 100644
--- a/content/browser/renderer_host/media/audio_renderer_host.cc
+++ b/content/browser/renderer_host/media/audio_renderer_host.cc
@@ -8,7 +8,6 @@
 #include "base/metrics/histogram.h"
 #include "base/process.h"
 #include "base/shared_memory.h"
-#include "content/browser/renderer_host/media/audio_common.h"
 #include "content/browser/renderer_host/media/audio_sync_reader.h"
 #include "content/browser/renderer_host/media/media_observer.h"
 #include "content/common/media/audio_messages.h"
@@ -200,7 +199,8 @@ void AudioRendererHost::OnCreateStream(
 
   // Select the hardware packet size if not specified.
   if (!audio_params.samples_per_packet) {
-    audio_params.samples_per_packet = SelectSamplesPerPacket(audio_params);
+    audio_params.samples_per_packet =
+        media::SelectSamplesPerPacket(audio_params.sample_rate);
   }

[Chris Rogers, 2012/02/22 23:08:53]

Do we need to handle the case where packet size is not specified?

I just talked with Nicholas Fullagar about NaCl/Pepper and he said they don't
ever pass in 0.  And I'm pretty sure we never do for <audio>/WebRTC/WebAudio

It would be great to DCHECK this and return failure instead of needing to call
this function.

   uint32 packet_size = audio_params.GetPacketSize();