Add AudioTrackRecorder for audio component of MediaStream recording.

content/renderer/media/audio_track_recorder_unittest.cc

+// 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/audio_track_recorder.h"
+
+#include "base/run_loop.h"
+#include "base/stl_util.h"
+#include "base/strings/utf_string_conversions.h"
+#include "content/renderer/media/media_stream_audio_source.h"
+#include "content/renderer/media/mock_media_constraint_factory.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
+#include "content/renderer/media/webrtc_local_audio_track.h"
+#include "media/audio/simple_sources.h"
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "third_party/WebKit/public/web/WebHeap.h"
+#include "third_party/opus/src/include/opus.h"
+
+using ::testing::_;
+using ::testing::DoAll;
+using ::testing::InSequence;
+using ::testing::Mock;
+using ::testing::Return;
+using ::testing::SaveArg;
+using ::testing::TestWithParam;
+using ::testing::ValuesIn;
+
+namespace {
+
+// Input audio format.
+const media::AudioParameters::Format kDefaultInputFormat =
+    media::AudioParameters::AUDIO_PCM_LOW_LATENCY;
+const int kDefaultBitsPerSample = 16;
+const int kDefaultSampleRate = 48000;
+// The |frames_per_buffer| field of AudioParameters is not used by ATR.
+const int kIgnoreFramesPerBuffer = 1;
+const int kOpusMaxBufferDurationMS = 120;
+
+}  // namespace
+
+namespace content {
+
+ACTION_P(RunClosure, closure) {
+  closure.Run();
+}
+
+struct ATRTestParams {
+  const media::AudioParameters::Format input_format;
+  const media::ChannelLayout channel_layout;
+  const int sample_rate;
+  const int bits_per_sample;
+};
+
+const ATRTestParams kATRTestParams[] = {
+    // Equivalent to default settings:
+    {media::AudioParameters::AUDIO_PCM_LOW_LATENCY, /* input format */
+     media::CHANNEL_LAYOUT_STEREO,                  /* channel layout */
+     kDefaultSampleRate,                            /* sample rate */
+     kDefaultBitsPerSample},                        /* bits per sample */
+    // Change to mono:
+    {media::AudioParameters::AUDIO_PCM_LOW_LATENCY, media::CHANNEL_LAYOUT_MONO,
+     kDefaultSampleRate, kDefaultBitsPerSample},
+    // Different sampling rate as well:
+    {media::AudioParameters::AUDIO_PCM_LOW_LATENCY, media::CHANNEL_LAYOUT_MONO,
+     24000, kDefaultBitsPerSample},
+    {media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+     media::CHANNEL_LAYOUT_STEREO, 8000, kDefaultBitsPerSample},
+};
+
+class AudioTrackRecorderTest : public TestWithParam<ATRTestParams> {
+ public:
+  // Initialize |first_params_| based on test parameters, and |second_params_|
+  // to always be the same thing.
+  AudioTrackRecorderTest()
+      : first_params_(GetParam().input_format,
+                      GetParam().channel_layout,
+                      GetParam().sample_rate,
+                      GetParam().bits_per_sample,
+                      kIgnoreFramesPerBuffer),
+        second_params_(kDefaultInputFormat,
+                       media::CHANNEL_LAYOUT_STEREO,
+                       kDefaultSampleRate,
+                       kDefaultBitsPerSample,
+                       kIgnoreFramesPerBuffer),
+        first_source_(first_params_.channels(),     /* # channels */
+                      440,                          /* frequency */
+                      first_params_.sample_rate()), /* sample rate */
+        second_source_(second_params_.channels(),
+                       440,
+                       second_params_.sample_rate()),
+        opus_decoder_(nullptr) {
+    ResetDecoder(first_params_);
+    PrepareBlinkTrack();
+    audio_track_recorder_.reset(new AudioTrackRecorder(
+        blink_track_, base::Bind(&AudioTrackRecorderTest::OnEncodedAudio,
+                                 base::Unretained(this))));
+  }
+
+  ~AudioTrackRecorderTest() {
+    opus_decoder_destroy(opus_decoder_);
+    opus_decoder_ = nullptr;
+    audio_track_recorder_.reset();
+    blink_track_.reset();
+    blink::WebHeap::collectAllGarbageForTesting();
+  }
+
+  void ResetDecoder(const media::AudioParameters& params) {
+    if (opus_decoder_) {
+      opus_decoder_destroy(opus_decoder_);
+      opus_decoder_ = nullptr;
+    }
+
+    int error;
+    opus_decoder_ =
+        opus_decoder_create(params.sample_rate(), params.channels(), &error);
+    EXPECT_TRUE(error == OPUS_OK && opus_decoder_);
+
+    max_frames_per_buffer_ =
+        kOpusMaxBufferDurationMS * params.sample_rate() / 1000;
+    buffer_.reset(new float[max_frames_per_buffer_ * params.channels()]);
+  }
+
+  scoped_ptr<media::AudioBus> GetFirstSourceAudioBus() {
+    scoped_ptr<media::AudioBus> bus(media::AudioBus::Create(
+        first_params_.channels(),
+        first_params_.sample_rate() *
+            audio_track_recorder_->GetOpusBufferDuration(
+                first_params_.sample_rate()) /
+            1000));
+    first_source_.OnMoreData(bus.get(), 0);
+    return bus.Pass();
+  }
+  scoped_ptr<media::AudioBus> GetSecondSourceAudioBus() {
+    scoped_ptr<media::AudioBus> bus(media::AudioBus::Create(
+        second_params_.channels(),
+        second_params_.sample_rate() *
+            audio_track_recorder_->GetOpusBufferDuration(
+                second_params_.sample_rate()) /
+            1000));
+    second_source_.OnMoreData(bus.get(), 0);
+    return bus.Pass();
+  }
+
+  MOCK_METHOD3(DoOnEncodedAudio,
+               void(const media::AudioParameters& params,
+                    std::string encoded_data,
+                    base::TimeTicks timestamp));
+
+  void OnEncodedAudio(const media::AudioParameters& params,
+                      scoped_ptr<std::string> encoded_data,
+                      base::TimeTicks timestamp) {
+    EXPECT_TRUE(!encoded_data->empty());
+
+    // Decode |encoded_data| and check we get the expected number of frames
+    // per buffer.
+    EXPECT_EQ(
+        params.sample_rate() *
+            audio_track_recorder_->GetOpusBufferDuration(params.sample_rate()) /
+            1000,
+        opus_decode_float(
+            opus_decoder_,
+            reinterpret_cast<uint8*>(string_as_array(encoded_data.get())),
+            encoded_data->size(), buffer_.get(), max_frames_per_buffer_, 0));
+
+    DoOnEncodedAudio(params, *encoded_data, timestamp);
+  }
+
+  const base::MessageLoop message_loop_;
+
+  // ATR and WebMediaStreamTrack for fooling it.
+  scoped_ptr<AudioTrackRecorder> audio_track_recorder_;
+  blink::WebMediaStreamTrack blink_track_;
+
+  // Two different sets of AudioParameters for testing re-init of ATR.
+  const media::AudioParameters first_params_;
+  const media::AudioParameters second_params_;
+
+  // AudioSources for creating AudioBuses.
+  media::SineWaveAudioSource first_source_;
+  media::SineWaveAudioSource second_source_;
+
+  // Decoder for verifying data was properly encoded.
+  OpusDecoder* opus_decoder_;
+  int max_frames_per_buffer_;
+  scoped_ptr<float[]> buffer_;
+
+ private:
+  // Prepares a blink track of a given MediaStreamType and attaches the native
+  // track, which can be used to capture audio data and pass it to the producer.
+  // Adapted from media::WebRTCLocalAudioSourceProviderTest.
+  void PrepareBlinkTrack() {
+    MockMediaConstraintFactory constraint_factory;
+    scoped_refptr<WebRtcAudioCapturer> capturer(
+        WebRtcAudioCapturer::CreateCapturer(
+            -1, StreamDeviceInfo(),
+            constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
+    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
+        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+    scoped_ptr<WebRtcLocalAudioTrack> native_track(
+        new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
+    blink::WebMediaStreamSource audio_source;
+    audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
+                            blink::WebMediaStreamSource::TypeAudio,
+                            base::UTF8ToUTF16("dummy_source_name"),
+                            false /* remote */, true /* readonly */);
+    blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
+                            audio_source);
+    blink_track_.setExtraData(native_track.release());
+  }
+
+  DISALLOW_COPY_AND_ASSIGN(AudioTrackRecorderTest);
+};
+
+TEST_P(AudioTrackRecorderTest, OnData) {
+  InSequence s;
+  base::RunLoop run_loop;
+  base::Closure quit_closure = run_loop.QuitClosure();
+
+  // Give ATR initial audio parameters.
+  audio_track_recorder_->OnSetFormat(first_params_);
+  // TODO(ajose): consider adding WillOnce(SaveArg...) and inspecting, as done
+  // in VTR unittests. http://crbug.com/548856
+  const base::TimeTicks time1 = base::TimeTicks::Now();
+  EXPECT_CALL(*this, DoOnEncodedAudio(_, _, time1)).Times(1);
+  audio_track_recorder_->OnData(*GetFirstSourceAudioBus(), time1);
+
+  // Send more audio.
+  const base::TimeTicks time2 = base::TimeTicks::Now();
+  EXPECT_CALL(*this, DoOnEncodedAudio(_, _, _))
+      .Times(1)
+      // Only reset the decoder once we've heard back:
+      .WillOnce(RunClosure(base::Bind(&AudioTrackRecorderTest::ResetDecoder,
+                                      base::Unretained(this), second_params_)));
+  audio_track_recorder_->OnData(*GetFirstSourceAudioBus(), time2);
+
+  // Give ATR new audio parameters.
+  audio_track_recorder_->OnSetFormat(second_params_);
+
+  // Send audio with different params.
+  const base::TimeTicks time3 = base::TimeTicks::Now();
+  EXPECT_CALL(*this, DoOnEncodedAudio(_, _, _))
+      .Times(1)
+      .WillOnce(RunClosure(quit_closure));
+  audio_track_recorder_->OnData(*GetSecondSourceAudioBus(), time3);
+
+  run_loop.Run();
+  Mock::VerifyAndClearExpectations(this);
+}
+
+INSTANTIATE_TEST_CASE_P(, AudioTrackRecorderTest, ValuesIn(kATRTestParams));
+
+}  // namespace content