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

content/renderer/media/webrtc/peer_connection_remote_audio_source.cc

Index: content/renderer/media/webrtc/peer_connection_remote_audio_source.cc
diff --git a/content/renderer/media/webrtc/peer_connection_remote_audio_source.cc b/content/renderer/media/webrtc/peer_connection_remote_audio_source.cc
new file mode 100644
index 0000000000000000000000000000000000000000..91800553da1ea46d22acafe2d80230b415c295a0
--- /dev/null
+++ b/content/renderer/media/webrtc/peer_connection_remote_audio_source.cc
@@ -0,0 +1,139 @@
+// Copyright 2015 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/peer_connection_remote_audio_source.h"
+
+#include "base/logging.h"
+#include "media/base/audio_bus.h"
+
+namespace content {
+
+namespace {
+// Used as an identifier for the down-casters.
+void* const kClassIdentifier = const_cast<void**>(&kClassIdentifier);
+}  // namespace
+
+PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack(
+    const scoped_refptr<webrtc::AudioTrackInterface>& track_interface)
+    : MediaStreamAudioTrack(false /* is_local_track*/),
+      track_interface_(track_interface) {
+  DVLOG(1)
+      << "PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack()";
+}
+
+PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack() {
+  DVLOG(1)
+      << "PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack()";
+}
+
+// static
+PeerConnectionRemoteAudioTrack* PeerConnectionRemoteAudioTrack::From(
+    MediaStreamAudioTrack* track) {
+  if (track && track->GetClassIdentifier() == kClassIdentifier)
+    return static_cast<PeerConnectionRemoteAudioTrack*>(track);
+  return nullptr;
+}
+
+void PeerConnectionRemoteAudioTrack::SetEnabled(bool enabled) {
+  DCHECK(main_render_thread_checker_.CalledOnValidThread());
+
+  // This affects the shared state of the source for whether or not it's a part
+  // of the mixed audio that's rendered for remote tracks from WebRTC.
+  // All tracks from the same source will share this state and thus can step
+  // on each other's toes.
+  // This is also why we can't check the enabled state for equality with
+  // |enabled| before setting the mixing enabled state. This track's enabled
+  // state and the shared state might not be the same.
+  track_interface_->set_enabled(enabled);
+
+  MediaStreamAudioTrack::SetEnabled(enabled);
+}
+
+void* PeerConnectionRemoteAudioTrack::GetClassIdentifier() const {
+  return kClassIdentifier;
+}
+
+PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource(
+    const scoped_refptr<webrtc::AudioTrackInterface>& track_interface)
+    : MediaStreamAudioSource(false /* is_local_source */),
+      track_interface_(track_interface),
+      is_started_(false) {
+  DCHECK(track_interface_.get());
+  DVLOG(1)
+      << "PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource()";
+}
+
+PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource() {
+  DVLOG(1)
+      << "PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource()";
+  // Superclass will call StopSource() just in case.
+}
+
+scoped_ptr<MediaStreamAudioTrack>
+PeerConnectionRemoteAudioSource::CreateMediaStreamAudioTrack(
+    const std::string& id) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  return make_scoped_ptr<MediaStreamAudioTrack>(
+      new PeerConnectionRemoteAudioTrack(track_interface_));
+}
+
+void PeerConnectionRemoteAudioSource::DoStopSource() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (is_stopped_)
+    return;
+  if (is_started_) {
+    track_interface_->RemoveSink(this);
+    VLOG(1) << "Stopped PeerConnection remote audio source with id="
+            << track_interface_->id();
+  }
+  is_stopped_ = true;
+}
+
+bool PeerConnectionRemoteAudioSource::EnsureSourceIsStarted() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  if (is_stopped_)
+    return false;
+  if (is_started_)
+    return true;
+
+  VLOG(1) << "Starting PeerConnection remote audio source with id="
+          << track_interface_->id();
+  track_interface_->AddSink(this);
+  is_started_ = true;
+  return true;
+}
+
+void PeerConnectionRemoteAudioSource::OnData(
+    const void* audio_data, int bits_per_sample, int sample_rate,
+    size_t number_of_channels, size_t number_of_frames) {
+  // TODO(tommi): We should get the timestamp from WebRTC.
+  base::TimeTicks playout_time(base::TimeTicks::Now());
+
+  if (!audio_bus_ ||
+      static_cast<size_t>(audio_bus_->channels()) != number_of_channels ||
+      static_cast<size_t>(audio_bus_->frames()) != number_of_frames) {
+    audio_bus_ = media::AudioBus::Create(number_of_channels,
+                                         number_of_frames);
+  }
+
+  audio_bus_->FromInterleaved(audio_data, number_of_frames,
+                              bits_per_sample / 8);
+
+  media::AudioParameters params = MediaStreamAudioSource::GetAudioParameters();
+  if (!params.IsValid() ||
+      params.format() != media::AudioParameters::AUDIO_PCM_LOW_LATENCY ||
+      static_cast<size_t>(params.channels()) != number_of_channels ||
+      params.sample_rate() != sample_rate ||
+      static_cast<size_t>(params.frames_per_buffer()) != number_of_frames) {
+    MediaStreamAudioSource::SetFormat(media::AudioParameters(
+        media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+        media::GuessChannelLayout(number_of_channels),
+        sample_rate, 16, number_of_frames));
+  }
+
+  MediaStreamAudioSource::DeliverDataToTracks(*audio_bus_, playout_time);
+}
+
+}  // namespace content