Enable audio capture on Android

media/audio/android/audio_manager_android.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/android/audio_manager_android.h"
 
 #include "base/logging.h"
 #include "media/audio/android/opensles_input.h"
 #include "media/audio/android/opensles_output.h"
 #include "media/audio/audio_manager.h"
+#include "media/audio/audio_util.h"
 #include "media/audio/fake_audio_input_stream.h"
 
 namespace media {
 
 // Maximum number of output streams that can be open simultaneously.
 static const int kMaxOutputStreams = 10;
 
 AudioManager* CreateAudioManager() {
   return new AudioManagerAndroid();
 }
 
 AudioManagerAndroid::AudioManagerAndroid() {
   SetMaxOutputStreamsAllowed(kMaxOutputStreams);
 }
 
 AudioManagerAndroid::~AudioManagerAndroid() {
   Shutdown();
 }
 
 bool AudioManagerAndroid::HasAudioOutputDevices() {
   return true;
 }
 
 bool AudioManagerAndroid::HasAudioInputDevices() {
+  return true;
+}
+
+bool AudioManagerAndroid::CanShowAudioInputSettings() {
   return false;
 }
 
+void AudioManagerAndroid::GetAudioInputDeviceNames(
+    media::AudioDeviceNames* device_names) {
+  if (device_names->empty()) {
+    device_names->push_front(media::AudioDeviceName(
+        kDefaultDeviceName, kDefaultDeviceId));
+  }

[tommi (sloooow) - chröme, 2013/02/11 09:52:58]

This implementation is making a few assumptions that I'm not sure are safe. 
Would it make sense to add something like this?

} else {
  DCHECK_EQ(device_names->size(), 1u);
  DCHECK_EQ(device_names->at(0).unique_id,
            kDefaultDeviceId);
}

Alternatively, the implementation could be changed to either:

DCHECK(device_names->empty());
device_names->push_front(
    media::AudioDeviceName(kDefaultDeviceName, kDefaultDeviceId));

or

device_names->clear();
device_names->push_front(
    media::AudioDeviceName(kDefaultDeviceName, kDefaultDeviceId));


Personally, I'd prefer the one with DCHECK(device_names->empty());

[leozwang1, 2013/02/11 18:03:43]

Done.

+}
+
+AudioParameters
+AudioManagerAndroid::GetPreferredLowLatencyOutputStreamParameters(
+    const AudioParameters& input_params) {
+  // TODO(leozwang): Android defines the minimal buffer size requirment
+  // we should follow it. From Android 4.1, a new audio low latency api
+  // set was introduced and is under development, we want to take advantage
+  // of it.
+  int buffer_size = GetAudioHardwareBufferSize();
+  if (input_params.frames_per_buffer() < buffer_size)
+    buffer_size = input_params.frames_per_buffer();
+
+  return AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY, input_params.channel_layout(),
+      input_params.sample_rate(), 16, buffer_size);

[tommi (sloooow) - chröme, 2013/02/11 09:52:58]

Is audio always 16 bit?

[leozwang1, 2013/02/11 18:03:43]

opensl only supports 8bit or 16bit, webrtc uses 16bit as the default. Maybe fix
it to 16bit in this patch? (Linux uses 16bit too.)

+}
+
 AudioOutputStream* AudioManagerAndroid::MakeLinearOutputStream(
       const AudioParameters& params) {
   DCHECK_EQ(AudioParameters::AUDIO_PCM_LINEAR, params.format());
   return new OpenSLESOutputStream(this, params);
 }
 
 AudioOutputStream* AudioManagerAndroid::MakeLowLatencyOutputStream(
       const AudioParameters& params) {
   DCHECK_EQ(AudioParameters::AUDIO_PCM_LOW_LATENCY, params.format());
   return new OpenSLESOutputStream(this, params);
 }
 
 AudioInputStream* AudioManagerAndroid::MakeLinearInputStream(
     const AudioParameters& params, const std::string& device_id) {
   DCHECK_EQ(AudioParameters::AUDIO_PCM_LINEAR, params.format());
   return new OpenSLESInputStream(this, params);
 }
 
 AudioInputStream* AudioManagerAndroid::MakeLowLatencyInputStream(
     const AudioParameters& params, const std::string& device_id) {
   DCHECK_EQ(AudioParameters::AUDIO_PCM_LOW_LATENCY, params.format());
   return new OpenSLESInputStream(this, params);
 }
 
 }  // namespace media