// as buffer size since that is the native buffer size of WebRtc packet
// running on.
params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
- channel_layout, number_of_channels, 0, sample_rate, 16,
+ channel_layout, number_of_channels, sample_rate, 16,
sample_rate / 100);
audio_format_changed_ = true;