MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/webrtc_audio_sink_adapter.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 "base/logging.h"
-#include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
-#include "media/base/audio_bus.h"
-#include "third_party/webrtc/api/mediastreaminterface.h"
-
-namespace content {
-
-WebRtcAudioSinkAdapter::WebRtcAudioSinkAdapter(
-    webrtc::AudioTrackSinkInterface* sink)
-    : sink_(sink) {
-  DCHECK(sink);
-}
-
-WebRtcAudioSinkAdapter::~WebRtcAudioSinkAdapter() {
-}
-
-bool WebRtcAudioSinkAdapter::IsEqual(
-    const webrtc::AudioTrackSinkInterface* other) const {
-  return (other == sink_);
-}
-
-void WebRtcAudioSinkAdapter::OnData(const media::AudioBus& audio_bus,
-                                    base::TimeTicks estimated_capture_time) {
-  DCHECK_EQ(audio_bus.frames(), params_.frames_per_buffer());
-  DCHECK_EQ(audio_bus.channels(), params_.channels());
-  // TODO(henrika): Remove this conversion once the interface in libjingle
-  // supports float vectors.
-  audio_bus.ToInterleaved(audio_bus.frames(),
-                          sizeof(interleaved_data_[0]),
-                          interleaved_data_.get());
-  sink_->OnData(interleaved_data_.get(),
-                16,
-                params_.sample_rate(),
-                audio_bus.channels(),
-                audio_bus.frames());
-}
-
-void WebRtcAudioSinkAdapter::OnSetFormat(
-    const media::AudioParameters& params) {
-  DCHECK(params.IsValid());
-  params_ = params;
-  const int num_pcm16_data_elements =
-      params_.frames_per_buffer() * params_.channels();
-  interleaved_data_.reset(new int16_t[num_pcm16_data_elements]);
-}
-
-}  // namespace content