#include "content/renderer/media/audio_device_factory.h"
#include "content/renderer/media/media_stream_dispatcher.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc_audio_renderer.h"
#include "content/renderer/render_frame_impl.h"
#include "media/audio/audio_output_device.h"
#include "media/base/audio_block_fifo.h"
sink_params_ = media::AudioParameters(source_params_.format(),
source_params_.channel_layout(), source_params_.sample_rate(),
source_params_.bits_per_sample(),
-#if defined(OS_ANDROID)
- // On Android, input and output use the same sample rate. In order to
- // use the low latency mode, we need to use the buffer size suggested by
- // the AudioManager for the sink. It will later be used to decide
- // the buffer size of the shared memory buffer.
- frames_per_buffer_,
-#else
- 2 * source_params_.frames_per_buffer(),
-#endif
+ WebRtcAudioRenderer::GetOptimalBufferSize(source_params_.sample_rate(),
+ frames_per_buffer_),
// If DUCKING is enabled on the source, it needs to be enabled on the
// sink as well.
source_params_.effects() | implicit_ducking_effect);