Flip AudioOutputResampler to on by default.

media/audio/audio_manager_base.cc

Index: media/audio/audio_manager_base.cc
diff --git a/media/audio/audio_manager_base.cc b/media/audio/audio_manager_base.cc
index 1d8a4540f988c41a38f8b5f09b7e27cb05f42cdb..910ca8615347333bb7a600cebf82d07581188623 100644
--- a/media/audio/audio_manager_base.cc
+++ b/media/audio/audio_manager_base.cc
@@ -157,7 +157,7 @@ AudioOutputStream* AudioManagerBase::MakeAudioOutputStreamProxy(
       base::TimeDelta::FromSeconds(kStreamCloseDelaySeconds);
 
   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
-  if (cmd_line->HasSwitch(switches::kEnableAudioOutputResampler) &&
+  if (!cmd_line->HasSwitch(switches::kDisableAudioOutputResampler) &&
       params.format() == AudioParameters::AUDIO_PCM_LOW_LATENCY) {

[scherkus (not reviewing), 2012/09/13 12:43:09]

to confirm -- this will affect web audio + web rtc as well, right?

[DaleCurtis, 2012/09/14 09:04:44]

Henrik says WebRTC will be okay since it uses the same buffer size in most
cases. We may hit the FIFO every now and then though, but so long as delay is
accurate and buffer sizes relatively small it should be okay. WebAudio uses the
same method for determining buffer size, so it won't be affected.

The automatic fallback though will certainly affect both of them, so they'll
have more users, but some of them may be grumpy about performance.

     scoped_refptr<AudioOutputDispatcher> dispatcher = new AudioOutputResampler(
         this, params,