Index: content/renderer/media/audio_renderer_impl.cc |
diff --git a/content/renderer/media/audio_renderer_impl.cc b/content/renderer/media/audio_renderer_impl.cc |
index e2568da79c05e7dd85f7f114c96f7960ea5d41dd..22fe62ede773c3a8b64f7d665aed406bff3c1934 100644 |
--- a/content/renderer/media/audio_renderer_impl.cc |
+++ b/content/renderer/media/audio_renderer_impl.cc |
@@ -16,26 +16,6 @@ |
#include "media/audio/audio_util.h" |
#include "media/base/filter_host.h" |
-// We define GetBufferSizeForSampleRate() instead of using |
-// GetAudioHardwareBufferSize() in audio_util because we're using |
-// the AUDIO_PCM_LINEAR flag, instead of AUDIO_PCM_LOW_LATENCY, |
-// which the audio_util functions assume. |
-// |
-// See: http://code.google.com/p/chromium/issues/detail?id=103627 |
-// for a more detailed description of the subtleties. |
-static size_t GetBufferSizeForSampleRate(int sample_rate) { |
- // kNominalBufferSize has been tested on Windows, Mac OS X, and Linux |
- // using the low-latency audio codepath (SyncSocket implementation) |
- // with the AUDIO_PCM_LINEAR flag. |
- const size_t kNominalBufferSize = 2048; |
- |
- if (sample_rate <= 48000) |
- return kNominalBufferSize; |
- else if (sample_rate <= 96000) |
- return kNominalBufferSize * 2; |
- return kNominalBufferSize * 4; |
-} |
- |
AudioRendererImpl::AudioRendererImpl(media::AudioRendererSink* sink) |
: AudioRendererBase(), |
bytes_per_second_(0), |
@@ -91,7 +71,7 @@ bool AudioRendererImpl::OnInitialize(int bits_per_channel, |
if (!is_initialized_) { |
sink_->Initialize( |
- GetBufferSizeForSampleRate(sample_rate), |
+ media::SelectSamplesPerPacket(sample_rate), |
audio_parameters_.channels, |
audio_parameters_.sample_rate, |
audio_parameters_.format, |