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

content/renderer/media/webaudio_media_stream_source.cc

Index: content/renderer/media/webaudio_media_stream_source.cc
diff --git a/content/renderer/media/webaudio_media_stream_source.cc b/content/renderer/media/webaudio_media_stream_source.cc
deleted file mode 100644
index 7f4c75d37ff6f15c6fc451b96e8345aa83aad7b6..0000000000000000000000000000000000000000
--- a/content/renderer/media/webaudio_media_stream_source.cc
+++ /dev/null
@@ -1,111 +0,0 @@
-// 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 "content/renderer/media/webaudio_media_stream_source.h"
-
-#include "base/bind.h"
-#include "base/bind_helpers.h"
-#include "base/logging.h"
-
-namespace content {
-
-WebAudioMediaStreamSource::WebAudioMediaStreamSource(
-    blink::WebMediaStreamSource* blink_source)
-    : MediaStreamAudioSource(false /* is_remote */),
-      is_registered_consumer_(false),
-      fifo_(base::Bind(&WebAudioMediaStreamSource::DeliverRebufferedAudio,
-                       base::Unretained(this))),
-      blink_source_(*blink_source) {
-  DVLOG(1) << "WebAudioMediaStreamSource::WebAudioMediaStreamSource()";
-}
-
-WebAudioMediaStreamSource::~WebAudioMediaStreamSource() {
-  DVLOG(1) << "WebAudioMediaStreamSource::~WebAudioMediaStreamSource()";
-  EnsureSourceIsStopped();
-}
-
-void WebAudioMediaStreamSource::setFormat(size_t number_of_channels,
-                                          float sample_rate) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  VLOG(1) << "WebAudio media stream source changed format to: channels="
-          << number_of_channels << ", sample_rate=" << sample_rate;
-
-  // If the channel count is greater than 8, use discrete layout. However,
-  // anything beyond 8 is ignored by some audio tracks/sinks.
-  media::ChannelLayout channel_layout =
-      number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
-                             : media::GuessChannelLayout(number_of_channels);
-
-  // Set the format used by this WebAudioMediaStreamSource. We are using 10ms
-  // data as a buffer size since that is the native buffer size of WebRtc packet
-  // running on.
-  //
-  // TODO(miu): Re-evaluate whether this is needed. For now (this refactoring),
-  // I did not want to change behavior. http://crbug.com/577874
-  fifo_.Reset(sample_rate / 100);
-  media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                                channel_layout, sample_rate, 16,
-                                fifo_.frames_per_buffer());
-  // Take care of the discrete channel layout case.
-  params.set_channels_for_discrete(number_of_channels);
-  MediaStreamAudioSource::SetFormat(params);
-
-  if (!wrapper_bus_ || wrapper_bus_->channels() != params.channels())
-    wrapper_bus_ = media::AudioBus::CreateWrapper(params.channels());
-}
-
-bool WebAudioMediaStreamSource::EnsureSourceIsStarted() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (is_registered_consumer_)
-    return true;
-  if (blink_source_.isNull() || !blink_source_.requiresAudioConsumer())
-    return false;
-  VLOG(1) << "Starting WebAudio media stream source.";
-  blink_source_.addAudioConsumer(this);
-  is_registered_consumer_ = true;
-  return true;
-}
-
-void WebAudioMediaStreamSource::EnsureSourceIsStopped() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (!is_registered_consumer_)
-    return;
-  is_registered_consumer_ = false;
-  DCHECK(!blink_source_.isNull());
-  blink_source_.removeAudioConsumer(this);
-  blink_source_.reset();
-  VLOG(1) << "Stopped WebAudio media stream source. Final audio parameters={"
-          << GetAudioParameters().AsHumanReadableString() << "}.";
-}
-
-void WebAudioMediaStreamSource::consumeAudio(
-    const blink::WebVector<const float*>& audio_data,
-    size_t number_of_frames) {
-  // TODO(miu): Plumbing is needed to determine the actual capture timestamp
-  // of the audio, instead of just snapshotting TimeTicks::Now(), for proper
-  // audio/video sync.  http://crbug.com/335335
-  current_reference_time_ = base::TimeTicks::Now();
-
-  wrapper_bus_->set_frames(number_of_frames);
-  DCHECK_EQ(wrapper_bus_->channels(), static_cast<int>(audio_data.size()));
-  for (size_t i = 0; i < audio_data.size(); ++i)
-    wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
-
-  // The following will result in zero, one, or multiple synchronous calls to
-  // DeliverRebufferedAudio().
-  fifo_.Push(*wrapper_bus_);
-}
-
-void WebAudioMediaStreamSource::DeliverRebufferedAudio(
-    const media::AudioBus& audio_bus,
-    int frame_delay) {
-  const base::TimeTicks reference_time =
-      current_reference_time_ +
-      base::TimeDelta::FromMicroseconds(
-          frame_delay * base::Time::kMicrosecondsPerSecond /
-          MediaStreamAudioSource::GetAudioParameters().sample_rate());
-  MediaStreamAudioSource::DeliverDataToTracks(audio_bus, reference_time);
-}
-
-}  // namespace content