Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/webrtc_audio_source_adapter_unittest.cc

+// 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 <numeric>
+#include <vector>
+
+#include "base/memory/ptr_util.h"
+#include "base/message_loop/message_loop.h"
+#include "base/run_loop.h"
+#include "remoting/proto/audio.pb.h"
+#include "remoting/protocol/fake_audio_source.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
+#include "third_party/webrtc/base/refcount.h"
+
+namespace remoting {
+namespace protocol {
+
+namespace {
+
+const int kSampleRate = 48000;
+const int kBytesPerSample = 2;
+const int kChannels = 2;
+constexpr base::TimeDelta kFrameDuration =
+    base::TimeDelta::FromMilliseconds(10);
+
+class FakeAudioSink : public webrtc::AudioTrackSinkInterface{
+ public:
+  FakeAudioSink() {}
+  ~FakeAudioSink() override {}
+
+  void OnData(const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              size_t number_of_channels,
+              size_t number_of_samples) override {
+    EXPECT_EQ(kSampleRate, sample_rate);
+    EXPECT_EQ(kBytesPerSample * 8, bits_per_sample);
+    EXPECT_EQ(kChannels, static_cast<int>(number_of_channels));
+    EXPECT_EQ(kSampleRate * kFrameDuration / base::TimeDelta::FromSeconds(1),
+              static_cast<int>(number_of_samples));
+    const int16_t* samples = reinterpret_cast<const int16_t*>(audio_data);
+    samples_.insert(samples_.end(), samples,
+                    samples + number_of_samples * kChannels);
+  }
+
+  const std::vector<int16_t>& samples() { return samples_; }
+
+ private:
+  std::vector<int16_t> samples_;
+};
+
+}  // namespace
+
+class WebrtcAudioSourceAdapterTest : public testing::Test {
+ public:
+  void SetUp() override {
+    audio_source_adapter_ = new rtc::RefCountedObject<WebrtcAudioSourceAdapter>(
+        message_loop_.task_runner());
+    audio_source_ = new FakeAudioSource();
+    audio_source_adapter_->Start(base::WrapUnique(audio_source_));
+    audio_source_adapter_->AddSink(&sink_);
+    base::RunLoop().RunUntilIdle();
+  }
+
+  void TearDown() override {
+    audio_source_adapter_ = nullptr;
+    base::RunLoop().RunUntilIdle();
+  }
+
+ protected:
+  base::MessageLoop message_loop_;
+  FakeAudioSource* audio_source_;
+  scoped_refptr<WebrtcAudioSourceAdapter> audio_source_adapter_;
+  FakeAudioSink sink_;
+};
+
+TEST_F(WebrtcAudioSourceAdapterTest, PartialFrames) {
+  int16_t sample_value = 1;
+  std::vector<int> frame_sizes_ms = {10, 12, 18, 2, 5, 7, 55, 13, 8};
+  for (int frame_size_ms : frame_sizes_ms) {
+    int num_samples = frame_size_ms * kSampleRate / 1000;
+    std::vector<int16_t> data(num_samples * kChannels);
+    for (int i = 0; i < num_samples; ++i) {
+      data[i * kChannels] = sample_value;
+      data[i * kChannels + 1] = -sample_value;
+      ++sample_value;
+    }
+
+    std::unique_ptr<AudioPacket> packet(new AudioPacket());
+    packet->add_data(reinterpret_cast<char*>(&(data[0])),
+                     num_samples * kChannels * sizeof(int16_t));
+    packet->set_encoding(AudioPacket::ENCODING_RAW);
+    packet->set_sampling_rate(AudioPacket::SAMPLING_RATE_48000);
+    packet->set_bytes_per_sample(AudioPacket::BYTES_PER_SAMPLE_2);
+    packet->set_channels(AudioPacket::CHANNELS_STEREO);
+    audio_source_->callback().Run(std::move(packet));
+  }
+
+  int total_length_ms =
+      std::accumulate(frame_sizes_ms.begin(), frame_sizes_ms.end(), 0,
+                      [](int sum, int x) { return sum + x; });
+  const std::vector<int16_t>& received = sink_.samples();
+  int total_samples = total_length_ms * kSampleRate / 1000;
+  ASSERT_EQ(total_samples * kChannels, static_cast<int>(received.size()));
+  sample_value = 1;
+  for (int i = 0; i < total_samples; ++i) {
+    ASSERT_EQ(sample_value, received[i * kChannels]) << i;
+    ASSERT_EQ(-sample_value, received[i * kChannels + 1]);
+    ++sample_value;
+  }
+}
+
+
+}  // namespace protocol
+}  // namespace remoting