MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webaudio_capturer_source.cc

Index: content/renderer/media/webaudio_capturer_source.cc
diff --git a/content/renderer/media/webaudio_capturer_source.cc b/content/renderer/media/webaudio_capturer_source.cc
deleted file mode 100644
index bcebcbb3e24e523288827db01d6b740a019a29fd..0000000000000000000000000000000000000000
--- a/content/renderer/media/webaudio_capturer_source.cc
+++ /dev/null
@@ -1,143 +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_capturer_source.h"
-
-#include "base/logging.h"
-#include "base/time/time.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-
-using media::AudioBus;
-using media::AudioFifo;
-using media::AudioParameters;
-using media::ChannelLayout;
-using media::CHANNEL_LAYOUT_MONO;
-using media::CHANNEL_LAYOUT_STEREO;
-
-static const int kMaxNumberOfBuffersInFifo = 5;
-
-namespace content {
-
-WebAudioCapturerSource::WebAudioCapturerSource(
-    const blink::WebMediaStreamSource& blink_source)
-    : track_(NULL),
-      audio_format_changed_(false),
-      blink_source_(blink_source) {
-}
-
-WebAudioCapturerSource::~WebAudioCapturerSource() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  removeFromBlinkSource();
-}
-
-void WebAudioCapturerSource::setFormat(
-    size_t number_of_channels, float sample_rate) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
-           << sample_rate << ")";
-
-  // If the channel count is greater than 8, use discrete layout. However,
-  // anything beyond 8 is ignored by the subsequent (WebRTC) audio pipeline.
-  ChannelLayout channel_layout =
-      number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
-                             : media::GuessChannelLayout(number_of_channels);
-
-  base::AutoLock auto_lock(lock_);
-
-  // Set the format used by this WebAudioCapturerSource. We are using 10ms data
-  // as buffer size since that is the native buffer size of WebRtc packet
-  // running on.
-  params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
-                sample_rate, 16, sample_rate / 100);
-
-  // Take care of the discrete channel layout case.
-  params_.set_channels_for_discrete(number_of_channels);
-
-  audio_format_changed_ = true;
-
-  wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
-  capture_bus_ = AudioBus::Create(params_);
-
-  fifo_.reset(new AudioFifo(
-      params_.channels(),
-      kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
-}
-
-void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  DCHECK(track);
-  base::AutoLock auto_lock(lock_);
-  track_ = track;
-}
-
-void WebAudioCapturerSource::Stop() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  {
-    base::AutoLock auto_lock(lock_);
-    track_ = NULL;
-  }
-  // removeFromBlinkSource() should not be called while |lock_| is acquired,
-  // as it could result in a deadlock.
-  removeFromBlinkSource();
-}
-
-void WebAudioCapturerSource::consumeAudio(
-    const blink::WebVector<const float*>& audio_data,
-    size_t number_of_frames) {
-  base::AutoLock auto_lock(lock_);
-  if (!track_)
-    return;
-
-  // Update the downstream client if the audio format has been changed.
-  if (audio_format_changed_) {
-    track_->OnSetFormat(params_);
-    audio_format_changed_ = false;
-  }
-
-  wrapper_bus_->set_frames(number_of_frames);
-
-  // Make sure WebKit is honoring what it told us up front
-  // about the channels.
-  DCHECK_EQ(params_.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]));
-
-  // Handle mismatch between WebAudio buffer-size and WebRTC.
-  int available = fifo_->max_frames() - fifo_->frames();
-  if (available < static_cast<int>(number_of_frames)) {
-    NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
-    return;
-  }
-
-  // Compute the estimated capture time of the first sample frame of audio that
-  // will be consumed from the FIFO in the loop below.
-  base::TimeTicks estimated_capture_time = base::TimeTicks::Now() -
-      fifo_->frames() * base::TimeDelta::FromSeconds(1) / params_.sample_rate();
-
-  fifo_->Push(wrapper_bus_.get());
-  while (fifo_->frames() >= capture_bus_->frames()) {
-    fifo_->Consume(capture_bus_.get(), 0, capture_bus_->frames());
-    track_->Capture(*capture_bus_, estimated_capture_time, false);
-
-    // Advance the estimated capture time for the next FIFO consume operation.
-    estimated_capture_time +=
-        capture_bus_->frames() * base::TimeDelta::FromSeconds(1) /
-            params_.sample_rate();
-  }
-}
-
-// If registered as audio consumer in |blink_source_|, deregister from
-// |blink_source_| and stop keeping a reference to |blink_source_|.
-// Failure to call this method when stopping the track might leave an invalid
-// WebAudioCapturerSource reference still registered as an audio consumer on
-// the blink side.
-void WebAudioCapturerSource::removeFromBlinkSource() {
-  if (!blink_source_.isNull()) {
-    blink_source_.removeAudioConsumer(this);
-    blink_source_.reset();
-  }
-}
-
-}  // namespace content