Reduce jitter from uneven SincResampler buffer size requests.

remoting/codec/audio_encoder_opus.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "remoting/codec/audio_encoder_opus.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/time.h"
 #include "media/base/audio_bus.h"
@@ skipping 62 matching lines @@
 
   frame_size_ = sampling_rate_ * kFrameSizeMs /
       base::Time::kMillisecondsPerSecond;
 
   if (sampling_rate_ != kOpusSamplingRate) {
     resample_buffer_.reset(
         new char[kFrameSamples * kBytesPerSample * channels_]);
     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);
   }
 
   // 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_]);
 }
 
 void AudioEncoderOpus::DestroyEncoder() {
   if (encoder_) {
     opus_encoder_destroy(encoder_);
     encoder_ = NULL;
   }
 
@@ skipping 50 matching lines @@
   const int16* next_sample =
       reinterpret_cast<const int16*>(packet->data(0).data());
 
   // Create a new packet of encoded data.
   scoped_ptr<AudioPacket> encoded_packet(new AudioPacket());
   encoded_packet->set_encoding(AudioPacket::ENCODING_OPUS);
   encoded_packet->set_sampling_rate(kOpusSamplingRate);
   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) {
     const int16* pcm_buffer = NULL;
 
     // Combine the packet with the leftover samples, if any.
     if (leftover_samples_ > 0) {
       pcm_buffer = leftover_buffer_.get();
       int samples_to_copy = samples_wanted - leftover_samples_;
       memcpy(leftover_buffer_.get() + leftover_samples_ * channels_,
              next_sample, samples_to_copy * kBytesPerSample * channels_);
     } else {
       pcm_buffer = next_sample;
     }
 
     // Resample data if necessary.
     int samples_consumed = 0;
     if (resampler_.get()) {
       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;
 
       resampler_bus_->ToInterleaved(kFrameSamples, kBytesPerSample,
                                     resample_buffer_.get());
       pcm_buffer = reinterpret_cast<int16*>(resample_buffer_.get());
     } else {
       samples_consumed = frame_size_;
     }
 
@@ skipping 37 matching lines @@
   }
 
   // Return NULL if there's nothing in the packet.
   if (encoded_packet->data_size() == 0)
     return scoped_ptr<AudioPacket>();
 
   return encoded_packet.Pass();
 }
 
 }  // namespace remoting