Flip AudioOutputResampler to on by default.

media/audio/audio_manager_base.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 "media/audio/audio_manager_base.h"
 
 #include "base/bind.h"
 #include "base/command_line.h"
 #include "base/message_loop_proxy.h"
 #include "base/threading/thread.h"
@@ skipping 139 matching lines @@
   DCHECK(GetMessageLoop()->BelongsToCurrentThread());
 
   AudioOutputDispatchersMap::iterator it = output_dispatchers_.find(params);
   if (it != output_dispatchers_.end())
     return new AudioOutputProxy(it->second);
 
   base::TimeDelta close_delay =
       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,
         GetPreferredLowLatencyOutputStreamParameters(params.channel_layout()),
         close_delay);
     output_dispatchers_[params] = dispatcher;
     return new AudioOutputProxy(dispatcher);
   }
 
 #if defined(ENABLE_AUDIO_MIXER)
   // TODO(dalecurtis): Browser side mixing has a couple issues that must be
@@ skipping 109 matching lines @@
 AudioParameters AudioManagerBase::GetPreferredLowLatencyOutputStreamParameters(
     ChannelLayout channel_layout) {
   // TODO(dalecurtis): This should include bits per channel and channel layout
   // eventually.
   return AudioParameters(
       AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
       GetAudioHardwareSampleRate(), 16, GetAudioHardwareBufferSize());
 }
 
 }  // namespace media