Index: media/audio/audio_output_resampler.cc |
diff --git a/media/audio/audio_output_resampler.cc b/media/audio/audio_output_resampler.cc |
index 34343d75337c6598126a3696f252690a9bdfdefb..c945aae4ad4adf1530d8978d8452eda6a59b352f 100644 |
--- a/media/audio/audio_output_resampler.cc |
+++ b/media/audio/audio_output_resampler.cc |
@@ -132,6 +132,16 @@ void AudioOutputResampler::Initialize() { |
io_ratio_ *= static_cast<double>(params_.channels()) / |
output_params_.channels(); |
+ // We allow the clients to set the buffer size for low latency path. |
+ if (output_params_.format() == AudioParameters::AUDIO_PCM_LOW_LATENCY && |
+ params_.frames_per_buffer() != output_params_.frames_per_buffer()) { |
scherkus (not reviewing)
2012/09/17 16:27:23
doesn't this essentially disable the FIFO when sam
no longer working on chromium
2012/09/17 21:56:28
True, we should not do this.
|
+ output_params_.Reset(output_params_.format(), |
+ output_params_.channel_layout(), |
+ output_params_.sample_rate(), |
+ output_params_.bits_per_sample(), |
+ params_.frames_per_buffer()); |
+ } |
+ |
// Since the resampler / output device may want a different buffer size than |
// the caller asked for, we need to use a FIFO to ensure that both sides |
// read in chunk sizes they're configured for. |