audio_manager_linux: Allow CRAS to decide if sample rate conversion is needed.

media/audio/linux/audio_manager_linux.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/linux/audio_manager_linux.h"
 
 #include "base/command_line.h"
 #include "base/environment.h"
 #include "base/logging.h"
 #include "base/nix/xdg_util.h"
@@ skipping 332 matching lines @@
   return new AudioManagerLinux();
 }
 
 AudioParameters AudioManagerLinux::GetPreferredLowLatencyOutputStreamParameters(
     const AudioParameters& input_params) {
   // Since Linux doesn't actually have a low latency path the hardware buffer
   // size is quite large in order to prevent glitches with general usage.  Some
   // clients, such as WebRTC, have a more limited use case and work acceptably
   // with a smaller buffer size.  The check below allows clients which want to
   // try a smaller buffer size on Linux to do so.
-  int buffer_size = 0;
-  if (static_cast<size_t>(input_params.frames_per_buffer()) <
-      GetAudioHardwareBufferSize()) {
+  int buffer_size = GetAudioHardwareBufferSize();
+  if (static_cast<size_t>(input_params.frames_per_buffer()) < buffer_size)

[DaleCurtis, 2012/09/25 20:49:46]

Drop static_cast now.

[scherkus (not reviewing), 2012/09/25 20:52:39]

nit: you can remove static_cast<>

     buffer_size = input_params.frames_per_buffer();
-  } else {
-    buffer_size = GetAudioHardwareBufferSize();
-  }
+
+  int sample_rate = GetAudioHardwareSampleRate();
+  // CRAS will sample rate convert if needed, pass the desired sample rate.

[DaleCurtis, 2012/09/25 20:49:46]

s/pass/so pass through input sample rate./

+  if (UseCras())
+    sample_rate = input_params.sample_rate();
 
   // TODO(dalecurtis): This should include bits per channel and channel layout
   // eventually.
   return AudioParameters(
       AudioParameters::AUDIO_PCM_LOW_LATENCY, input_params.channel_layout(),
-      GetAudioHardwareSampleRate(), 16, buffer_size);
+      sample_rate, 16, buffer_size);
 }
 
 }  // namespace media