Add audio support in WebrtcConnectionToHost, audio unittest

remoting/protocol/webrtc_audio_sink_adapter.cc

Index: remoting/protocol/webrtc_audio_sink_adapter.cc
diff --git a/remoting/protocol/webrtc_audio_sink_adapter.cc b/remoting/protocol/webrtc_audio_sink_adapter.cc
new file mode 100644
index 0000000000000000000000000000000000000000..5e2da715ad11d8dd500ef5806365400d430555a9
--- /dev/null
+++ b/remoting/protocol/webrtc_audio_sink_adapter.cc
@@ -0,0 +1,80 @@
+// 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_sink_adapter.h"
+
+#include "base/callback.h"
+#include "remoting/proto/audio.pb.h"
+#include "remoting/protocol/audio_stub.h"
+
+namespace remoting {
+namespace protocol {
+
+WebrtcAudioSinkAdapter::WebrtcAudioSinkAdapter(
+    scoped_refptr<webrtc::MediaStreamInterface> stream,
+    base::WeakPtr<AudioStub> audio_stub) {
+  audio_stub_ = audio_stub;
+
+  media_stream_ = std::move(stream);
+
+  webrtc::AudioTrackVector audio_tracks = media_stream_->GetAudioTracks();
+
+  // Caller must verify that the media stream contains audio tracks.
+  DCHECK(!audio_tracks.empty());
+
+  if (audio_tracks.size() > 1U) {
+    LOG(WARNING) << "Received media stream with multiple audio tracks.";
+  }
+
+  audio_track_ = audio_tracks[0];
+  audio_track_->GetSource()->AddSink(this);
+}
+
+WebrtcAudioSinkAdapter::~WebrtcAudioSinkAdapter() {
+  audio_track_->GetSource()->RemoveSink(this);
+}
+
+void WebrtcAudioSinkAdapter::OnData(const void* audio_data,
+                                    int bits_per_sample,
+                                    int sample_rate,
+                                    size_t number_of_channels,
+                                    size_t number_of_frames) {
+  if (!audio_stub_)
+    return;
+
+  std::unique_ptr<AudioPacket> audio_packet(new AudioPacket());
+  audio_packet->set_encoding(AudioPacket::ENCODING_RAW);
+
+  switch (sample_rate) {
+    case 44100:
+      audio_packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_44100);
+      break;
+    case 48000:
+      audio_packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_48000);
+      break;
+    default:
+      LOG(WARNING) << "Unsupported sampling rate: " << sample_rate;
+      return;
+  }
+
+  if (bits_per_sample != 16) {
+    LOG(WARNING) << "Unsupported bits/sample: " << bits_per_sample;
+    return;
+  }
+  audio_packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
+
+  if (number_of_channels != 2) {
+    LOG(WARNING) << "Unsupported number of channels: " << number_of_channels;
+    return;
+  }
+  audio_packet->set_channels(AudioPacket::CHANNELS_STEREO);
+
+  size_t data_size =
+      number_of_frames * number_of_channels * (bits_per_sample / 8);
+  audio_packet->add_data(reinterpret_cast<const char*>(audio_data), data_size);
+  audio_stub_->ProcessAudioPacket(std::move(audio_packet), base::Closure());
+}
+
+}  // namespace protocol
+}  // namespace remoting