Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/webrtc_audio_source_adapter.cc

Index: remoting/protocol/webrtc_audio_source_adapter.cc
diff --git a/remoting/protocol/webrtc_audio_source_adapter.cc b/remoting/protocol/webrtc_audio_source_adapter.cc
new file mode 100644
index 0000000000000000000000000000000000000000..d9d9ca3817c602da9b5cd03b65d90bc92b1ce434
--- /dev/null
+++ b/remoting/protocol/webrtc_audio_source_adapter.cc
@@ -0,0 +1,191 @@
+// 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 "remoting/protocol/webrtc_audio_source_adapter.h"
+
+#include "base/bind.h"
+#include "base/logging.h"
+#include "base/synchronization/lock.h"
+#include "base/threading/thread_checker.h"
+#include "remoting/proto/audio.pb.h"
+#include "remoting/protocol/audio_source.h"
+
+namespace remoting {
+namespace protocol {
+
+static const int kChannels = 2;
+static const int kBytesPerSample = 2;
+
+// Frame size expected by webrtc::AudioTrackSinkInterface.
+static constexpr base::TimeDelta kAudioFrameDuration =
+    base::TimeDelta::FromMilliseconds(10);
+
+class WebrtcAudioSourceAdapter::Core {
+ public:
+  Core();
+  ~Core();
+
+  void Start(std::unique_ptr<AudioSource> audio_source);
+  void Pause(bool pause);
+  void AddSink(webrtc::AudioTrackSinkInterface* sink);
+  void RemoveSink(webrtc::AudioTrackSinkInterface* sink);
+
+ private:
+  void OnAudioPacket(std::unique_ptr<AudioPacket> packet);
+
+  std::unique_ptr<AudioSource> audio_source_;
+
+  bool paused_ = false;
+
+  int sampling_rate_ = 0;
+
+  // webrtc::AudioTrackSinkInterface expects to get audio in 10ms frames (see
+  // kAudioFrameDuration). AudioSource may generate AudioPackets for time
+  // intervals that are not multiple of 10ms. In that case the left-over samples
+  // are kept in |partial_frame_| until the next AudioPacket is captured by the
+  // AudioSource.
+  std::vector<uint8_t> partial_frame_;
+
+  base::ObserverList<webrtc::AudioTrackSinkInterface> audio_sinks_;
+  base::Lock audio_sinks_lock_;
+
+  base::ThreadChecker thread_checker_;
+};
+
+WebrtcAudioSourceAdapter::Core::Core() {
+  thread_checker_.DetachFromThread();
+}
+
+WebrtcAudioSourceAdapter::Core::~Core() {}
+
+void WebrtcAudioSourceAdapter::Core::Start(
+    std::unique_ptr<AudioSource> audio_source) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  audio_source_ = std::move(audio_source);
+  audio_source_->Start(
+      base::Bind(&Core::OnAudioPacket, base::Unretained(this)));
+}
+
+void WebrtcAudioSourceAdapter::Core::Pause(bool pause) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  paused_ = pause;
+}
+
+void WebrtcAudioSourceAdapter::Core::AddSink(
+    webrtc::AudioTrackSinkInterface* sink) {
+  // Can be called on any thread.
+  base::AutoLock lock(audio_sinks_lock_);
+  audio_sinks_.AddObserver(sink);
+}
+
+void WebrtcAudioSourceAdapter::Core::RemoveSink(
+    webrtc::AudioTrackSinkInterface* sink) {
+  // Can be called on any thread.
+  base::AutoLock lock(audio_sinks_lock_);
+  audio_sinks_.RemoveObserver(sink);
+}
+
+void WebrtcAudioSourceAdapter::Core::OnAudioPacket(
+    std::unique_ptr<AudioPacket> packet) {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  if (paused_)
+    return;
+
+  DCHECK_EQ(packet->channels(), kChannels);
+  DCHECK_EQ(packet->bytes_per_sample(), kBytesPerSample);
+
+  if (sampling_rate_ != packet->sampling_rate()) {
+    sampling_rate_ = packet->sampling_rate();
+    partial_frame_.clear();
+  }
+
+  size_t samples_per_frame =
+      kAudioFrameDuration * sampling_rate_ / base::TimeDelta::FromSeconds(1);
+  size_t bytes_per_frame = kBytesPerSample * kChannels * samples_per_frame;
+
+  const std::string& data = packet->data(0);
+
+  size_t position = 0;
+
+  base::AutoLock lock(audio_sinks_lock_);
+
+  if (!partial_frame_.empty()) {
+    size_t bytes_to_append =
+        std::min(bytes_per_frame - partial_frame_.size(), data.size());
+    position += bytes_to_append;
+    partial_frame_.insert(partial_frame_.end(), data.data(),
+                             data.data() + bytes_to_append);
+    if (partial_frame_.size() < bytes_per_frame) {
+      // Still don't have full frame.
+      return;
+    }
+
+    // Here |partial_frame_| always contains a full frame.
+    DCHECK_EQ(partial_frame_.size(), bytes_per_frame);
+
+    FOR_EACH_OBSERVER(webrtc::AudioTrackSinkInterface, audio_sinks_,
+                      OnData(&partial_frame_.front(), kBytesPerSample * 8,
+                             sampling_rate_, kChannels, samples_per_frame));
+  }
+
+  while (position + bytes_per_frame <= data.size()) {
+    FOR_EACH_OBSERVER(webrtc::AudioTrackSinkInterface, audio_sinks_,
+                      OnData(data.data() + position, kBytesPerSample * 8,
+                             sampling_rate_, kChannels, samples_per_frame));
+    position += bytes_per_frame;
+  }
+
+  partial_frame_.assign(data.data() + position, data.data() + data.size());
+}
+
+WebrtcAudioSourceAdapter::WebrtcAudioSourceAdapter(
+    scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner)
+    : audio_task_runner_(audio_task_runner), core_(new Core()) {}
+
+WebrtcAudioSourceAdapter::~WebrtcAudioSourceAdapter() {
+  audio_task_runner_->DeleteSoon(FROM_HERE, core_.release());
+}
+
+void WebrtcAudioSourceAdapter::Start(
+    std::unique_ptr<AudioSource> audio_source) {
+  audio_task_runner_->PostTask(
+      FROM_HERE, base::Bind(&Core::Start, base::Unretained(core_.get()),
+                            base::Passed(&audio_source)));
+}
+
+void WebrtcAudioSourceAdapter::Pause(bool pause) {
+  audio_task_runner_->PostTask(
+      FROM_HERE,
+      base::Bind(&Core::Pause, base::Unretained(core_.get()), pause));
+}
+
+WebrtcAudioSourceAdapter::SourceState WebrtcAudioSourceAdapter::state() const {
+  return kLive;
+}
+
+bool WebrtcAudioSourceAdapter::remote() const {
+  return false;
+}
+
+void WebrtcAudioSourceAdapter::RegisterAudioObserver(AudioObserver* observer) {}
+
+void WebrtcAudioSourceAdapter::UnregisterAudioObserver(
+    AudioObserver* observer) {}
+
+void WebrtcAudioSourceAdapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
+  core_->AddSink(sink);
+}
+void WebrtcAudioSourceAdapter::RemoveSink(
+    webrtc::AudioTrackSinkInterface* sink) {
+  core_->RemoveSink(sink);
+}
+
+void WebrtcAudioSourceAdapter::RegisterObserver(
+    webrtc::ObserverInterface* observer) {}
+void WebrtcAudioSourceAdapter::UnregisterObserver(
+    webrtc::ObserverInterface* observer) {}
+
+}  // namespace protocol
+}  // namespace remoting