Introduce AudioBufferMemoryPool to avoid thrashing on audio buffers.

media/base/audio_buffer_converter.cc

 // Copyright 2014 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_buffer_converter.h"
 
 #include <algorithm>
 #include <cmath>
 
 #include "base/logging.h"
-#include "media/base/audio_buffer.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_decoder_config.h"
 #include "media/base/audio_timestamp_helper.h"
 #include "media/base/sinc_resampler.h"
 #include "media/base/timestamp_constants.h"
 #include "media/base/vector_math.h"
 
 namespace media {
 
 // Is the config presented by |buffer| a config change from |params|?
 static bool IsConfigChange(const AudioParameters& params,
                            const scoped_refptr<AudioBuffer>& buffer) {
   return buffer->sample_rate() != params.sample_rate() ||
          buffer->channel_count() != params.channels() ||
          buffer->channel_layout() != params.channel_layout();
 }
 
 AudioBufferConverter::AudioBufferConverter(const AudioParameters& output_params)
     : output_params_(output_params),
       input_params_(output_params),
       last_input_buffer_offset_(0),
       input_frames_(0),
       buffered_input_frames_(0.0),
       io_sample_rate_ratio_(1.0),
       timestamp_helper_(output_params_.sample_rate()),
-      is_flushing_(false) {}
+      is_flushing_(false),
+      pool_(new AudioBufferMemoryPool()) {}
 
 AudioBufferConverter::~AudioBufferConverter() {}
 
 void AudioBufferConverter::AddInput(const scoped_refptr<AudioBuffer>& buffer) {
   // On EOS flush any remaining buffered data.
   if (buffer->end_of_stream()) {
     Flush();
     queued_outputs_.push_back(buffer);
     return;
   }
@@ skipping 132 matching lines @@
     // How many calls to ProvideInput() we can satisfy completely.
     int chunks = input_frames_ / input_params_.frames_per_buffer();
 
     // How many output frames that corresponds to:
     request_frames = chunks * audio_converter_->ChunkSize();
   }
 
   if (!request_frames)
     return;
 
-  scoped_refptr<AudioBuffer> output_buffer =
-      AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
-                                output_params_.channel_layout(),
-                                output_params_.channels(),
-                                output_params_.sample_rate(),
-                                request_frames);
+  scoped_refptr<AudioBuffer> output_buffer = AudioBuffer::CreateBuffer(
+      kSampleFormatPlanarF32, output_params_.channel_layout(),
+      output_params_.channels(), output_params_.sample_rate(), request_frames,
+      pool_);
   std::unique_ptr<AudioBus> output_bus =
       AudioBus::CreateWrapper(output_buffer->channel_count());
 
   int frames_remaining = request_frames;
 
   // The AudioConverter wants requests of a fixed size, so we'll slide an
   // AudioBus of that size across the |output_buffer|.
   while (frames_remaining != 0) {
     // It's important that this is a multiple of AudioBus::kChannelAlignment in
     // all requests except for the last, otherwise downstream SIMD optimizations
@@ skipping 34 matching lines @@
   is_flushing_ = false;
   audio_converter_->Reset();
   DCHECK_EQ(input_frames_, 0);
   DCHECK_EQ(last_input_buffer_offset_, 0);
   DCHECK_LT(buffered_input_frames_, 1.0);
   DCHECK(queued_inputs_.empty());
   buffered_input_frames_ = 0.0;
 }
 
 }  // namespace media