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

content/renderer/media/webrtc_local_audio_track.cc

Index: content/renderer/media/webrtc_local_audio_track.cc
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
deleted file mode 100644
index cb48668eaeccc09a670c0a150b5d439e8d515778..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_track.cc
+++ /dev/null
@@ -1,241 +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/webrtc_local_audio_track.h"
-
-#include <stdint.h>
-
-#include <limits>
-
-#include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/media_stream_audio_sink_owner.h"
-#include "content/renderer/media/media_stream_audio_track_sink.h"
-#include "content/renderer/media/webaudio_capturer_source.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
-
-namespace content {
-
-WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
-    WebRtcLocalAudioTrackAdapter* adapter,
-    const scoped_refptr<WebRtcAudioCapturer>& capturer,
-    WebAudioCapturerSource* webaudio_source)
-    : MediaStreamAudioTrack(true),
-      adapter_(adapter),
-      capturer_(capturer),
-      webaudio_source_(webaudio_source) {
-  DCHECK(capturer.get() || webaudio_source);
-  signal_thread_checker_.DetachFromThread();
-
-  adapter_->Initialize(this);
-
-  DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
-}
-
-WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
-  DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
-  // Users might not call Stop() on the track.
-  Stop();
-}
-
-media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
-  DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  if (webaudio_source_.get()) {
-    return media::AudioParameters();
-  } else {
-    return capturer_->GetOutputFormat();
-  }
-}
-
-void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
-                                    base::TimeTicks estimated_capture_time,
-                                    bool force_report_nonzero_energy) {
-  DCHECK(capture_thread_checker_.CalledOnValidThread());
-  DCHECK(!estimated_capture_time.is_null());
-
-  // Calculate the signal level regardless of whether the track is disabled or
-  // enabled.  If |force_report_nonzero_energy| is true, |audio_bus| contains
-  // post-processed data that may be all zeros even though the signal contained
-  // energy before the processing.  In this case, report nonzero energy even if
-  // the energy of the data in |audio_bus| is zero.
-  const float minimum_signal_level =
-      force_report_nonzero_energy ? 1.0f / std::numeric_limits<int16_t>::max()
-                                  : 0.0f;
-  const float signal_level = std::max(
-      minimum_signal_level,
-      std::min(1.0f, level_calculator_->Calculate(audio_bus)));
-  const int signal_level_as_pcm16 =
-      static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
-                       0.5f /* rounding to nearest int */);
-  adapter_->SetSignalLevel(signal_level_as_pcm16);
-
-  scoped_refptr<WebRtcAudioCapturer> capturer;
-  SinkList::ItemList sinks;
-  SinkList::ItemList sinks_to_notify_format;
-  {
-    base::AutoLock auto_lock(lock_);
-    capturer = capturer_;
-    sinks = sinks_.Items();
-    sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
-  }
-
-  // Notify the tracks on when the format changes. This will do nothing if
-  // |sinks_to_notify_format| is empty.
-  for (const auto& sink : sinks_to_notify_format)
-    sink->OnSetFormat(audio_parameters_);
-
-  // Feed the data to the sinks.
-  // TODO(jiayl): we should not pass the real audio data down if the track is
-  // disabled. This is currently done so to feed input to WebRTC typing
-  // detection and should be changed when audio processing is moved from
-  // WebRTC to the track.
-  for (const auto& sink : sinks)
-    sink->OnData(audio_bus, estimated_capture_time);
-}
-
-void WebRtcLocalAudioTrack::OnSetFormat(
-    const media::AudioParameters& params) {
-  DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
-  // If the source is restarted, we might have changed to another capture
-  // thread.
-  capture_thread_checker_.DetachFromThread();
-  DCHECK(capture_thread_checker_.CalledOnValidThread());
-
-  audio_parameters_ = params;
-  level_calculator_.reset(new MediaStreamAudioLevelCalculator());
-
-  base::AutoLock auto_lock(lock_);
-  // Remember to notify all sinks of the new format.
-  sinks_.TagAll();
-}
-
-void WebRtcLocalAudioTrack::SetAudioProcessor(
-    const scoped_refptr<MediaStreamAudioProcessor>& processor) {
-  // if the |processor| does not have audio processing, which can happen if
-  // kDisableAudioTrackProcessing is set set or all the constraints in
-  // the |processor| are turned off. In such case, we pass NULL to the
-  // adapter to indicate that no stats can be gotten from the processor.
-  adapter_->SetAudioProcessor(processor->has_audio_processing() ?
-      processor : NULL);
-}
-
-void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
-  // This method is called from webrtc, on the signaling thread, when the local
-  // description is set and from the main thread from WebMediaPlayerMS::load
-  // (via WebRtcLocalAudioRenderer::Start).
-  DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
-         signal_thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
-  base::AutoLock auto_lock(lock_);
-
-  // Verify that |sink| is not already added to the list.
-  DCHECK(!sinks_.Contains(
-      MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)));
-
-  // Create (and add to the list) a new MediaStreamAudioTrackSink
-  // which owns the |sink| and delagates all calls to the
-  // MediaStreamAudioSink interface. It will be tagged in the list, so
-  // we remember to call OnSetFormat() on the new sink.
-  scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
-      new MediaStreamAudioSinkOwner(sink));
-  sinks_.AddAndTag(sink_owner.get());
-}
-
-void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
-  // See AddSink for additional context.  When local audio is stopped from
-  // webrtc, we'll be called here on the signaling thread.
-  DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
-         signal_thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
-
-  scoped_refptr<MediaStreamAudioTrackSink> removed_item;
-  {
-    base::AutoLock auto_lock(lock_);
-    removed_item = sinks_.Remove(
-        MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
-  }
-
-  // Clear the delegate to ensure that no more capture callbacks will
-  // be sent to this sink. Also avoids a possible crash which can happen
-  // if this method is called while capturing is active.
-  if (removed_item.get())
-    removed_item->Reset();
-}
-
-void WebRtcLocalAudioTrack::Start() {
-  DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcLocalAudioTrack::Start()";
-  if (webaudio_source_.get()) {
-    // If the track is hooking up with WebAudio, do NOT add the track to the
-    // capturer as its sink otherwise two streams in different clock will be
-    // pushed through the same track.
-    webaudio_source_->Start(this);
-  } else if (capturer_.get()) {
-    capturer_->AddTrack(this);
-  }
-
-  SinkList::ItemList sinks;
-  {
-    base::AutoLock auto_lock(lock_);
-    sinks = sinks_.Items();
-  }
-  for (SinkList::ItemList::const_iterator it = sinks.begin();
-       it != sinks.end();
-       ++it) {
-    (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateLive);
-  }
-}
-
-void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
-  DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  if (adapter_.get())
-    adapter_->set_enabled(enabled);
-}
-
-void WebRtcLocalAudioTrack::Stop() {
-  DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
-  if (!capturer_.get() && !webaudio_source_.get())
-    return;
-
-  if (webaudio_source_.get()) {
-    // Called Stop() on the |webaudio_source_| explicitly so that
-    // |webaudio_source_| won't push more data to the track anymore.
-    // Also note that the track is not registered as a sink to the |capturer_|
-    // in such case and no need to call RemoveTrack().
-    webaudio_source_->Stop();
-  } else {
-    // It is necessary to call RemoveTrack on the |capturer_| to avoid getting
-    // audio callback after Stop().
-    capturer_->RemoveTrack(this);
-  }
-
-  // Protect the pointers using the lock when accessing |sinks_| and
-  // setting the |capturer_| to NULL.
-  SinkList::ItemList sinks;
-  {
-    base::AutoLock auto_lock(lock_);
-    sinks = sinks_.Items();
-    sinks_.Clear();
-    webaudio_source_ = NULL;
-    capturer_ = NULL;
-  }
-
-  for (SinkList::ItemList::const_iterator it = sinks.begin();
-       it != sinks.end();
-       ++it){
-    (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
-    (*it)->Reset();
-  }
-}
-
-webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
-  DCHECK(main_render_thread_checker_.CalledOnValidThread());
-  return adapter_.get();
-}
-
-}  // namespace content