Enable low latency mode for audio playback on Android

content/renderer/media/webrtc_audio_renderer.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 "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "content/renderer/media/audio_device_factory.h"
@@ skipping 20 matching lines @@
 // rates below adds restrictions and Initialize() will fail if the user selects
 // any rate outside these ranges.
 const int kValidOutputRates[] = {96000, 48000, 44100, 32000, 16000};
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
 const int kValidOutputRates[] = {48000, 44100};
 #elif defined(OS_ANDROID)
 // TODO(leozwang): We want to use native sampling rate on Android to achieve
 // low latency, currently 16000 is used to work around audio problem on some
 // Android devices.
 const int kValidOutputRates[] = {48000, 44100, 16000};
+const int kDefaultOutputBufferSize = 2048;
 #else
 const int kValidOutputRates[] = {44100};
 #endif
 
 // TODO(xians): Merge the following code to WebRtcAudioCapturer, or remove.
 enum AudioFramesPerBuffer {
   k160,
   k320,
   k440,
   k480,
@@ skipping 111 matching lines @@
                       sample_rate, 16, buffer_size);
 
   // Set up audio parameters for the sink, i.e., the native audio output stream.
   // We strive to open up using native parameters to achieve best possible
   // performance and to ensure that no FIFO is needed on the browser side to
   // match the client request. Any mismatch between the source and the sink is
   // taken care of in this class instead using a pull FIFO.
 
   media::AudioParameters sink_params;
 
+#if defined(OS_ANDROID)
+  buffer_size = kDefaultOutputBufferSize;
+#else
   buffer_size = hardware_config->GetOutputBufferSize();
+#endif
+
   sink_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                     channel_layout, channels, 0, sample_rate, 16, buffer_size);
 
   // Create a FIFO if re-buffering is required to match the source input with
   // the sink request. The source acts as provider here and the sink as
   // consumer.
   fifo_delay_milliseconds_ = 0;
   if (source_params.frames_per_buffer() != sink_params.frames_per_buffer()) {
     DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer()
              << " to " << sink_params.frames_per_buffer();
@@ skipping 156 matching lines @@
   }
 
   // De-interleave each channel and convert to 32-bit floating-point
   // with nominal range -1.0 -> +1.0 to match the callback format.
   audio_bus->FromInterleaved(buffer_.get(),
                              audio_bus->frames(),
                              sizeof(buffer_[0]));
 }
 
 }  // namespace content