Reduce jitter from uneven SincResampler buffer size requests.

media/base/audio_converter.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 "media/base/audio_converter.h"
 
 #include <algorithm>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
@@ skipping 32 matching lines @@
       // since we'll mix into the AudioBus from the output stream.
       unmixed_audio_ = AudioBus::Create(
           input_params.channels(), output_params.frames_per_buffer());
     }
   }
 
   // Only resample if necessary since it's expensive.
   if (input_params.sample_rate() != output_params.sample_rate()) {
     DVLOG(1) << "Resampling from " << input_params.sample_rate() << " to "
              << output_params.sample_rate();
-    double io_sample_rate_ratio = input_params.sample_rate() /
+    const double io_sample_rate_ratio = input_params.sample_rate() /
         static_cast<double>(output_params.sample_rate());
+    const int request_size = disable_fifo ? SincResampler::kDefaultBlockSize :
+        input_params.frames_per_buffer();
     resampler_.reset(new MultiChannelResampler(
         downmix_early_ ? output_params.channels() :
             input_params.channels(),
-        io_sample_rate_ratio, base::Bind(
+        io_sample_rate_ratio, request_size, base::Bind(
             &AudioConverter::ProvideInput, base::Unretained(this))));
   }
 
   input_frame_duration_ = base::TimeDelta::FromMicroseconds(
       base::Time::kMicrosecondsPerSecond /
       static_cast<double>(input_params.sample_rate()));
   output_frame_duration_ = base::TimeDelta::FromMicroseconds(
       base::Time::kMicrosecondsPerSecond /
       static_cast<double>(output_params.sample_rate()));
 
-  if (disable_fifo)
+  // The resampler can be configured to work with a specific buffer size, so a
+  // FIFO is not necessary when resampling.
+  if (disable_fifo || resampler_)
     return;
 
-  // 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.
-  if (resampler_.get() ||
-      input_params.frames_per_buffer() != output_params.frames_per_buffer()) {
+  // Since the 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.
+  if (input_params.frames_per_buffer() != output_params.frames_per_buffer()) {
     DVLOG(1) << "Rebuffering from " << input_params.frames_per_buffer()
              << " to " << output_params.frames_per_buffer();
     audio_fifo_.reset(new AudioPullFifo(
         downmix_early_ ? output_params.channels() :
             input_params.channels(),
         input_params.frames_per_buffer(), base::Bind(
             &AudioConverter::SourceCallback,
             base::Unretained(this))));
   }
 }
@@ skipping 127 matching lines @@
 
 void AudioConverter::ProvideInput(int resampler_frame_delay, AudioBus* dest) {
   resampler_frame_delay_ = resampler_frame_delay;
   if (audio_fifo_)
     audio_fifo_->Consume(dest, dest->frames());
   else
     SourceCallback(0, dest);
 }
 
 }  // namespace media