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| author | dalecurtis@google.com <dalecurtis@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-05-08 22:53:28 +0000 |
|---|---|---|
| committer | dalecurtis@google.com <dalecurtis@google.com@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-05-08 22:53:28 +0000 |
| commit | a70013babca3b661bbc3ef198ad2f58c88ec4078 (patch) | |
| tree | 88f54b2897dd67430bcd97d94aee42788c6c1bf4 /remoting/codec | |
| parent | bfa6c1d83fc19226ad8c2ffd3bb958955b26279a (diff) | |
| download | chromium_src-a70013babca3b661bbc3ef198ad2f58c88ec4078.zip chromium_src-a70013babca3b661bbc3ef198ad2f58c88ec4078.tar.gz chromium_src-a70013babca3b661bbc3ef198ad2f58c88ec4078.tar.bz2 | |
Reduce jitter from uneven SincResampler buffer size requests.
Ensures all buffer requests are for the same size. Reduces jitter
by allowing clients with specific buffer size requirements to avoid
overreading.
BUG=none
TEST=media_unittests.
R=crogers@google.com, henrika@chromium.org, sergeyu@chromium.org
Review URL: https://codereview.chromium.org/14189035
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@199032 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'remoting/codec')
| -rw-r--r-- | remoting/codec/audio_encoder_opus.cc | 9 |
1 files changed, 6 insertions, 3 deletions
diff --git a/remoting/codec/audio_encoder_opus.cc b/remoting/codec/audio_encoder_opus.cc index 15160df..2efcf90 100644 --- a/remoting/codec/audio_encoder_opus.cc +++ b/remoting/codec/audio_encoder_opus.cc @@ -77,9 +77,12 @@ void AudioEncoderOpus::InitEncoder() { if (sampling_rate_ != kOpusSamplingRate) { resample_buffer_.reset( new char[kFrameSamples * kBytesPerSample * channels_]); + // TODO(sergeyu): Figure out the right buffer size to use per packet instead + // of using media::SincResampler::kDefaultRequestSize. resampler_.reset(new media::MultiChannelResampler( channels_, static_cast<double>(sampling_rate_) / kOpusSamplingRate, + media::SincResampler::kDefaultRequestSize, base::Bind(&AudioEncoderOpus::FetchBytesToResample, base::Unretained(this)))); resampler_bus_ = media::AudioBus::Create(channels_, kFrameSamples); @@ -88,7 +91,7 @@ void AudioEncoderOpus::InitEncoder() { // Drop leftover data because it's for different sampling rate. leftover_samples_ = 0; leftover_buffer_size_ = - frame_size_ + media::SincResampler::kMaximumLookAheadSize; + frame_size_ + media::SincResampler::kDefaultRequestSize; leftover_buffer_.reset( new int16[leftover_buffer_size_ * channels_]); } @@ -159,7 +162,7 @@ scoped_ptr<AudioPacket> AudioEncoderOpus::Encode( encoded_packet->set_channels(channels_); int prefetch_samples = - resampler_.get() ? media::SincResampler::kMaximumLookAheadSize : 0; + resampler_.get() ? media::SincResampler::kDefaultRequestSize : 0; int samples_wanted = frame_size_ + prefetch_samples; while (leftover_samples_ + samples_in_packet >= samples_wanted) { @@ -181,7 +184,7 @@ scoped_ptr<AudioPacket> AudioEncoderOpus::Encode( resampling_data_ = reinterpret_cast<const char*>(pcm_buffer); resampling_data_pos_ = 0; resampling_data_size_ = samples_wanted * channels_ * kBytesPerSample; - resampler_->Resample(resampler_bus_.get(), kFrameSamples); + resampler_->Resample(kFrameSamples, resampler_bus_.get()); resampling_data_ = NULL; samples_consumed = resampling_data_pos_ / channels_ / kBytesPerSample; |