Moved AudioUtil static functions to AudioManager interfaces

media/audio/linux/audio_manager_linux.cc

Index: media/audio/linux/audio_manager_linux.cc
diff --git a/media/audio/linux/audio_manager_linux.cc b/media/audio/linux/audio_manager_linux.cc
index fa8612573fe95397f905062cfd531aa20af5684b..aab5247b37a19aa97e95b337b2a96ad74734624a 100644
--- a/media/audio/linux/audio_manager_linux.cc
+++ b/media/audio/linux/audio_manager_linux.cc
@@ -11,7 +11,7 @@
 #include "base/process_util.h"
 #include "base/stl_util.h"
 #include "media/audio/audio_output_dispatcher.h"
-#include "media/audio/audio_util.h"
+#include "media/audio/audio_parameters.h"
 #if defined(USE_CRAS)
 #include "media/audio/cras/audio_manager_cras.h"
 #endif
@@ -21,6 +21,7 @@
 #if defined(USE_PULSEAUDIO)
 #include "media/audio/pulse/audio_manager_pulse.h"
 #endif
+#include "media/base/channel_layout.h"
 #include "media/base/limits.h"
 #include "media/base/media_switches.h"
 
@@ -29,6 +30,9 @@ namespace media {
 // Maximum number of output streams that can be open simultaneously.
 static const int kMaxOutputStreams = 50;
 
+// Default sample rate for input and output streams.
+static const int kDefaultSampleRate = 48000;
+
 // Since "default", "pulse" and "dmix" devices are virtual devices mapped to
 // real devices, we remove them from the list to avoiding duplicate counting.
 // In addition, note that we support no more than 2 channels for recording,
@@ -94,6 +98,23 @@ void AudioManagerLinux::GetAudioInputDeviceNames(
   GetAlsaAudioInputDevices(device_names);
 }
 
+AudioParameters AudioManagerLinux::GetDefaultOutputStreamParameters() {
+  static const int kDefaultOutputBufferSize = 512;
+
+  return AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, CHANNEL_LAYOUT_STEREO,
+      kDefaultSampleRate, 16, kDefaultOutputBufferSize);
+}
+
+AudioParameters AudioManagerLinux::GetDefaultInputStreamParameters(
+    const std::string& device_id) {
+  static const int kDefaultInputBufferSize = 1024;
+
+  return AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, CHANNEL_LAYOUT_STEREO,
+      kDefaultSampleRate, 16, kDefaultInputBufferSize);
+}
+
 void AudioManagerLinux::GetAlsaAudioInputDevices(
     media::AudioDeviceNames* device_names) {
   // Constants specified by the ALSA API for device hints.
@@ -299,9 +320,9 @@ AudioParameters AudioManagerLinux::GetPreferredLowLatencyOutputStreamParameters(
   // 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 = GetAudioHardwareBufferSize();
-  if (input_params.frames_per_buffer() < buffer_size)
-    buffer_size = input_params.frames_per_buffer();
+  AudioParameters default_params = GetDefaultOutputStreamParameters();
+  int buffer_size = std::min(input_params.frames_per_buffer(),
+                             default_params.frames_per_buffer());
 
   // TODO(dalecurtis): This should include bits per channel and channel layout
   // eventually.