| Index: content/renderer/media/webrtc_audio_renderer.cc
|
| ===================================================================
|
| --- content/renderer/media/webrtc_audio_renderer.cc (revision 206793)
|
| +++ content/renderer/media/webrtc_audio_renderer.cc (working copy)
|
| @@ -38,6 +38,7 @@
|
| // 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
|
| @@ -169,7 +170,12 @@
|
|
|
| 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);
|
|
|
|
|