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

content/renderer/media/webrtc/webrtc_audio_sink.cc

Index: content/renderer/media/webrtc/webrtc_audio_sink.cc
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_audio_sink.cc
similarity index 29%
rename from content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
rename to content/renderer/media/webrtc/webrtc_audio_sink.cc
index c86881b07a90eed64bca82048846e137536a2b21..d7500b45af3bcf5d7ce8bfad90802a63e2cb6864 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_audio_sink.cc
@@ -1,134 +1,154 @@
-// Copyright 2014 The Chromium Authors. All rights reserved.
+// Copyright 2016 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/webrtc_local_audio_track_adapter.h"
+#include "content/renderer/media/webrtc/webrtc_audio_sink.h"
 
+#include <algorithm>
+#include <limits>
+
+#include "base/bind.h"
+#include "base/bind_helpers.h"
 #include "base/location.h"
 #include "base/logging.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
-#include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-#include "content/renderer/render_thread_impl.h"
-#include "third_party/webrtc/api/mediastreaminterface.h"
+#include "base/message_loop/message_loop.h"
 
 namespace content {
 
-static const char kAudioTrackKind[] = "audio";
-
-scoped_refptr<WebRtcLocalAudioTrackAdapter>
-WebRtcLocalAudioTrackAdapter::Create(
-    const std::string& label,
-    webrtc::AudioSourceInterface* track_source) {
-  // TODO(tommi): Change this so that the signaling thread is one of the
-  // parameters to this method.
-  scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner;
-  RenderThreadImpl* current = RenderThreadImpl::current();
-  if (current) {
-    PeerConnectionDependencyFactory* pc_factory =
-        current->GetPeerConnectionDependencyFactory();
-    signaling_task_runner = pc_factory->GetWebRtcSignalingThread();
-    LOG_IF(ERROR, !signaling_task_runner) << "No signaling thread!";
-  } else {
-    LOG(WARNING) << "Assuming single-threaded operation for unit test.";
-  }
-
-  rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter =
-      new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>(
-          label, track_source, std::move(signaling_task_runner));
-  return adapter;
-}
-
-WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
+WebRtcAudioSink::WebRtcAudioSink(
     const std::string& label,
-    webrtc::AudioSourceInterface* track_source,
+    scoped_refptr<webrtc::AudioSourceInterface> track_source,
     scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
-    : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
-      owner_(NULL),
-      track_source_(track_source),
-      signaling_task_runner_(std::move(signaling_task_runner)) {}
-
-WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
+    : adapter_(new rtc::RefCountedObject<Adapter>(
+          label, std::move(track_source), std::move(signaling_task_runner))),
+      fifo_(base::Bind(&WebRtcAudioSink::DeliverRebufferedAudio,
+                       base::Unretained(this))) {
+  DVLOG(1) << "WebRtcAudioSink::WebRtcAudioSink()";
 }
 
-void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
-  DCHECK(!owner_);
-  DCHECK(owner);
-  owner_ = owner;
+WebRtcAudioSink::~WebRtcAudioSink() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DVLOG(1) << "WebRtcAudioSink::~WebRtcAudioSink()";
 }
 
-void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
+void WebRtcAudioSink::SetAudioProcessor(
     scoped_refptr<MediaStreamAudioProcessor> processor) {
+  DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(processor.get());
-  DCHECK(!audio_processor_);
-  audio_processor_ = std::move(processor);
+  adapter_->set_processor(std::move(processor));
 }
 
-void WebRtcLocalAudioTrackAdapter::SetLevel(
+void WebRtcAudioSink::SetLevel(
     scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
+  DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(level.get());
-  DCHECK(!level_);
-  level_ = std::move(level);
+  adapter_->set_level(std::move(level));
+}
+
+void WebRtcAudioSink::OnEnabledChanged(bool enabled) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  adapter_->signaling_task_runner()->PostTask(
+      FROM_HERE,
+      base::Bind(
+          base::IgnoreResult(&WebRtcAudioSink::Adapter::set_enabled),
+          adapter_, enabled));
+}
+
+void WebRtcAudioSink::OnData(const media::AudioBus& audio_bus,
+                             base::TimeTicks estimated_capture_time) {
+  DCHECK(audio_thread_checker_.CalledOnValidThread());
+  // The following will result in zero, one, or multiple synchronous calls to
+  // DeliverRebufferedAudio().
+  fifo_.Push(audio_bus);
+}
+
+void WebRtcAudioSink::OnSetFormat(const media::AudioParameters& params) {
+  // On a format change, the thread delivering audio might have also changed.
+  audio_thread_checker_.DetachFromThread();
+  DCHECK(audio_thread_checker_.CalledOnValidThread());
+
+  DCHECK(params.IsValid());
+  params_ = params;
+  fifo_.Reset(params_.frames_per_buffer());
+  const int num_pcm16_data_elements =
+      params_.frames_per_buffer() * params_.channels();
+  interleaved_data_.reset(new int16_t[num_pcm16_data_elements]);
 }
 
-std::string WebRtcLocalAudioTrackAdapter::kind() const {
-  return kAudioTrackKind;
+void WebRtcAudioSink::DeliverRebufferedAudio(const media::AudioBus& audio_bus,
+                                             int frame_delay) {
+  DCHECK(audio_thread_checker_.CalledOnValidThread());
+  DCHECK(params_.IsValid());
+
+  // TODO(miu): Why doesn't a WebRTC sink care about reference time passed to
+  // OnData(), and the |frame_delay| here?  How is AV sync achieved otherwise?
+
+  // 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());
+  adapter_->DeliverPCMToWebRtcSinks(interleaved_data_.get(),
+                                    params_.sample_rate(),
+                                    audio_bus.channels(),
+                                    audio_bus.frames());
 }
 
-bool WebRtcLocalAudioTrackAdapter::set_enabled(bool enable) {
-  // If we're not called on the signaling thread, we need to post a task to
-  // change the state on the correct thread.
-  if (signaling_task_runner_ &&
-      !signaling_task_runner_->BelongsToCurrentThread()) {
-    signaling_task_runner_->PostTask(FROM_HERE,
-        base::Bind(
-            base::IgnoreResult(&WebRtcLocalAudioTrackAdapter::set_enabled),
-            this, enable));
-    return true;
+WebRtcAudioSink::Adapter::Adapter(
+    const std::string& label,
+    scoped_refptr<webrtc::AudioSourceInterface> source,
+    scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
+    : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
+      source_(std::move(source)),
+      signaling_task_runner_(std::move(signaling_task_runner)) {
+  DCHECK(signaling_task_runner_);
+}
+
+WebRtcAudioSink::Adapter::~Adapter() {}
+
+void WebRtcAudioSink::Adapter::DeliverPCMToWebRtcSinks(
+    const int16_t* audio_data,
+    int sample_rate,
+    size_t number_of_channels,
+    size_t number_of_frames) {
+  base::AutoLock auto_lock(lock_);
+  for (webrtc::AudioTrackSinkInterface* sink : sinks_) {
+    sink->OnData(audio_data, sizeof(int16_t) * 8, sample_rate,
+                 number_of_channels, number_of_frames);
   }
+}
+
+std::string WebRtcAudioSink::Adapter::kind() const {
+  return webrtc::MediaStreamTrackInterface::kAudioKind;
+}
 
+bool WebRtcAudioSink::Adapter::set_enabled(bool enable) {
+  DCHECK(!signaling_task_runner_ ||
+         signaling_task_runner_->RunsTasksOnCurrentThread());
   return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
       set_enabled(enable);
 }
 
-void WebRtcLocalAudioTrackAdapter::AddSink(
-    webrtc::AudioTrackSinkInterface* sink) {
+void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
   DCHECK(!signaling_task_runner_ ||
          signaling_task_runner_->RunsTasksOnCurrentThread());
   DCHECK(sink);
-#ifndef NDEBUG
-  // Verify that |sink| has not been added.
-  for (ScopedVector<WebRtcAudioSinkAdapter>::const_iterator it =
-           sink_adapters_.begin();
-       it != sink_adapters_.end(); ++it) {
-    DCHECK(!(*it)->IsEqual(sink));
-  }
-#endif
-
-  std::unique_ptr<WebRtcAudioSinkAdapter> adapter(
-      new WebRtcAudioSinkAdapter(sink));
-  owner_->AddSink(adapter.get());
-  sink_adapters_.push_back(adapter.release());
+  base::AutoLock auto_lock(lock_);
+  DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
+  sinks_.push_back(sink);
 }
 
-void WebRtcLocalAudioTrackAdapter::RemoveSink(
+void WebRtcAudioSink::Adapter::RemoveSink(
     webrtc::AudioTrackSinkInterface* sink) {
   DCHECK(!signaling_task_runner_ ||
          signaling_task_runner_->RunsTasksOnCurrentThread());
-  DCHECK(sink);
-  for (ScopedVector<WebRtcAudioSinkAdapter>::iterator it =
-           sink_adapters_.begin();
-       it != sink_adapters_.end(); ++it) {
-    if ((*it)->IsEqual(sink)) {
-      owner_->RemoveSink(*it);
-      sink_adapters_.erase(it);
-      return;
-    }
-  }
+  base::AutoLock auto_lock(lock_);
+  const auto it = std::find(sinks_.begin(), sinks_.end(), sink);
+  if (it != sinks_.end())
+    sinks_.erase(it);
 }
 
-bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
+bool WebRtcAudioSink::Adapter::GetSignalLevel(int* level) {
   DCHECK(!signaling_task_runner_ ||
          signaling_task_runner_->RunsTasksOnCurrentThread());
 
@@ -146,16 +166,16 @@ bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
 }
 
 rtc::scoped_refptr<webrtc::AudioProcessorInterface>
-WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
+WebRtcAudioSink::Adapter::GetAudioProcessor() {
   DCHECK(!signaling_task_runner_ ||
          signaling_task_runner_->RunsTasksOnCurrentThread());
   return audio_processor_.get();
 }
 
-webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
+webrtc::AudioSourceInterface* WebRtcAudioSink::Adapter::GetSource() const {
   DCHECK(!signaling_task_runner_ ||
          signaling_task_runner_->RunsTasksOnCurrentThread());
-  return track_source_;
+  return source_.get();
 }
 
 }  // namespace content