Moved AudioUtil static functions to AudioManager interfaces

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"
 #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
 #include "media/audio/linux/alsa_input.h"
 #include "media/audio/linux/alsa_output.h"
 #include "media/audio/linux/alsa_wrapper.h"
 #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"
 
 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,
 // hence surround devices are not stored in the list.
 static const char* kInvalidAudioInputDevices[] = {
   "default",
   "null",
   "pulse",
   "dmix",
   "surround",
@@ skipping 45 matching lines @@
 void AudioManagerLinux::ShowAudioInputSettings() {
   ShowLinuxAudioInputSettings();
 }
 
 void AudioManagerLinux::GetAudioInputDeviceNames(
     media::AudioDeviceNames* device_names) {
   DCHECK(device_names->empty());
   GetAlsaAudioInputDevices(device_names);
 }
 
+AudioParameters AudioManagerLinux::GetInputStreamParameters(
+    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.
   static const char kPcmInterfaceName[] = "pcm";
   int card = -1;
 
   // Loop through the sound cards to get ALSA device hints.
   while (!wrapper_->CardNext(&card) && card >= 0) {
     void** hints = NULL;
     int error = wrapper_->DeviceNameHint(card, kPcmInterfaceName, &hints);
@@ skipping 137 matching lines @@
   DCHECK_EQ(AudioParameters::AUDIO_PCM_LINEAR, params.format());
   return MakeInputStream(params, device_id);
 }
 
 AudioInputStream* AudioManagerLinux::MakeLowLatencyInputStream(
     const AudioParameters& params, const std::string& device_id) {
   DCHECK_EQ(AudioParameters::AUDIO_PCM_LOW_LATENCY, params.format());
   return MakeInputStream(params, device_id);
 }
 
+AudioParameters AudioManagerLinux::GetPreferredOutputStreamParameters(
+    const AudioParameters& input_params) {
+  static const int kDefaultOutputBufferSize = 512;
+  ChannelLayout channel_layout = CHANNEL_LAYOUT_STEREO;
+  int sample_rate = kDefaultSampleRate;
+  int buffer_size = kDefaultOutputBufferSize;
+  int bits_per_sample = 16;
+  int input_channels = 0;
+  if (input_params.IsValid()) {
+    // 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.
+    // TODO(dalecurtis): This should include bits per channel and channel layout
+    // eventually.
+    sample_rate = input_params.sample_rate();
+    bits_per_sample = input_params.bits_per_sample();
+    channel_layout = input_params.channel_layout();
+    input_channels = input_params.input_channels();
+    buffer_size = std::min(input_params.frames_per_buffer(), buffer_size);
+  }
+
+  return AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, input_channels,
+      sample_rate, bits_per_sample, buffer_size);
+}
+
 AudioOutputStream* AudioManagerLinux::MakeOutputStream(
     const AudioParameters& params) {
   std::string device_name = AlsaPcmOutputStream::kAutoSelectDevice;
   if (CommandLine::ForCurrentProcess()->HasSwitch(
           switches::kAlsaOutputDevice)) {
     device_name = CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
         switches::kAlsaOutputDevice);
   }
   return new AlsaPcmOutputStream(device_name, params, wrapper_.get(), this);
 }
@@ skipping 21 matching lines @@
   if (CommandLine::ForCurrentProcess()->HasSwitch(switches::kUsePulseAudio)) {
     AudioManager* manager = AudioManagerPulse::Create();
     if (manager)
       return manager;
   }
 #endif
 
   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 = GetAudioHardwareBufferSize();
-  if (input_params.frames_per_buffer() < buffer_size)
-    buffer_size = input_params.frames_per_buffer();
-
-  // TODO(dalecurtis): This should include bits per channel and channel layout
-  // eventually.
-  return AudioParameters(
-      AudioParameters::AUDIO_PCM_LOW_LATENCY, input_params.channel_layout(),
-      input_params.sample_rate(), 16, buffer_size);
-}
-
 }  // namespace media