Moved AudioUtil static functions to AudioManager interfaces

media/audio/win/audio_manager_win.cc

Index: media/audio/win/audio_manager_win.cc
diff --git a/media/audio/win/audio_manager_win.cc b/media/audio/win/audio_manager_win.cc
index bd809b8c6cd2195cbc12e7fecaed8355444a41cb..fe485f69ebe6719ef1723d715c6cf8ff08ad3af3 100644
--- a/media/audio/win/audio_manager_win.cc
+++ b/media/audio/win/audio_manager_win.cc
@@ -20,7 +20,7 @@
 #include "base/process_util.h"
 #include "base/string_number_conversions.h"
 #include "base/string_util.h"
-#include "media/audio/audio_util.h"
+#include "media/audio/audio_parameters.h"
 #include "media/audio/win/audio_device_listener_win.h"
 #include "media/audio/win/audio_low_latency_input_win.h"
 #include "media/audio/win/audio_low_latency_output_win.h"
@@ -31,6 +31,7 @@
 #include "media/audio/win/wavein_input_win.h"
 #include "media/audio/win/waveout_output_win.h"
 #include "media/base/bind_to_loop.h"
+#include "media/base/channel_layout.h"
 #include "media/base/limits.h"
 #include "media/base/media_switches.h"
 
@@ -47,6 +48,9 @@ DEFINE_GUID(AM_KSCATEGORY_AUDIO, 0x6994ad04, 0x93ef, 0x11d0,
 
 namespace media {
 
+using WASAPIAudioInputStream::HardwareChannelCount;

[DaleCurtis, 2013/02/27 02:46:11]

I think this makes things a little confusing since we've got both input and
output methods in the file.  I'd prefer if you used the fully qualified names
below.

[no longer working on chromium, 2013/03/01 16:44:33]

true, done now. Thanks for pointing out.

+using WASAPIAudioInputStream::HardwareSampleRate;
+
 // Maximum number of output streams that can be open simultaneously.
 static const int kMaxOutputStreams = 50;
 
@@ -61,6 +65,10 @@ static const int kWinMaxChannels = 8;
 // play.
 static const int kNumInputBuffers = 3;
 
+// Buffer size to use for input and output stream when a proper size can't be
+// determined from the system
+static const int kFallbackBufferSize = 2048;
+
 static int GetVersionPartAsInt(DWORDLONG num) {
   return static_cast<int>(num & 0xffff);
 }
@@ -255,6 +263,62 @@ void AudioManagerWin::GetAudioInputDeviceNames(
   }
 }
 
+AudioParameters AudioManagerWin::GetDefaultOutputStreamParameters() {
+  const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
+  ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
+  int sample_rate = 0;
+  int buffer_size = 0;
+  if (!CoreAudioUtil::IsSupported()) {
+    // Fall back to Windows Wave implementation on Windows XP or lower
+    // and use 48kHz as default input sample rate, kFallbackBufferSize as
+    // default buffer size.
+    sample_rate = 48000;
+    buffer_size = kFallbackBufferSize;
+  } else if (cmd_line->HasSwitch(switches::kEnableExclusiveAudio)) {
+    // TODO(crogers): tune these values for best possible WebAudio performance.
+    // WebRTC works well at 48kHz and a buffer size of 480 samples will be used
+    // for this case. Note that exclusive mode is experimental.
+    // This sample rate will be combined with a buffer size of 256 samples,
+    // which corresponds to an output delay of ~5.33ms.
+    sample_rate = 48000;
+    buffer_size = 256;
+  } else {
+    // Hardware sample-rate on Windows can be configured, so we must query.
+    // TODO(henrika): improve possibility to specify an audio endpoint.
+    // Use the default device (same as for Wave) for now to be compatible.
+    sample_rate = HardwareSampleRate();
+
+    AudioParameters params;
+    HRESULT hr = CoreAudioUtil::GetPreferredAudioParameters(eRender, eConsole,
+                                                            &params);
+    buffer_size = FAILED(hr) ? kFallbackBufferSize : params.frames_per_buffer();
+    channel_layout = WASAPIAudioOutputStream::HardwareChannelLayout();
+  }
+
+  return AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
+      sample_rate, 16, buffer_size);
+}
+
+AudioParameters AudioManagerWin::GetDefaultInputStreamParameters(
+    const std::string& device_id) {
+  int sample_rate = 0;
+  ChannelLayout channel_layout = CHANNEL_LAYOUT_NONE;
+  if (!CoreAudioUtil::IsSupported()) {
+    sample_rate = 48000;
+    channel_layout = CHANNEL_LAYOUT_STEREO;
+  } else {
+    sample_rate = HardwareSampleRate(device_id);
+    channel_layout = HardwareChannelCount(device_id) == 1 ?
+            CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
+  }
+
+  // TODO(Henrika): improve the default buffer size value for input stream.
+  return AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
+      sample_rate, 16, kFallbackBufferSize);
+}
+
 // Factory for the implementations of AudioOutputStream for AUDIO_PCM_LINEAR
 // mode.
 // - PCMWaveOutAudioOutputStream: Based on the waveOut API.
@@ -362,16 +426,19 @@ AudioParameters AudioManagerWin::GetPreferredLowLatencyOutputStreamParameters(
   int bits_per_sample = input_params.bits_per_sample();
   ChannelLayout channel_layout = input_params.channel_layout();
   int input_channels = input_params.input_channels();
+  int buffer_size = input_params.frames_per_buffer();
   if (CoreAudioUtil::IsSupported()) {
-    sample_rate = GetAudioHardwareSampleRate();
-    bits_per_sample = 16;
-    channel_layout = WASAPIAudioOutputStream::HardwareChannelLayout();
+    AudioParameters default_params = GetDefaultOutputStreamParameters();
+    sample_rate = default_params.sample_rate();
+    bits_per_sample = default_params.bits_per_sample();
+    channel_layout = default_params.channel_layout();
+    buffer_size = default_params.frames_per_buffer();
   }
 
   // TODO(dalecurtis): This should include hardware bits per channel eventually.
   return AudioParameters(
       AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
-      sample_rate, bits_per_sample, GetAudioHardwareBufferSize());
+      sample_rate, bits_per_sample, buffer_size);
 }
 
 }  // namespace media