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/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"
+
+using ::testing::_;
+using ::testing::DoAll;
+using ::testing::InSequence;
+using ::testing::Mock;
+using ::testing::Return;
+using ::testing::SaveArg;
+
+namespace {
+
+// Input audio format.
+const media::AudioParameters::Format kInputFormat =
+    media::AudioParameters::AUDIO_PCM_LOW_LATENCY;
+const int kNumChannels = 1;

[miu, 2015/10/26 22:41:04]

Consider testing more than one channel (e.g., to test the sample-interleaving
code).

[ajose, 2015/10/27 18:41:28]

Done.

+const int kBitsPerSample = 16;
+const int kSamplingRate = 48000;
+const int kFramesPerBuffer = 480;
+
+}  // namespace
+
+namespace content {
+
+ACTION_P(RunClosure, closure) {
+  closure.Run();
+}
+
+class AudioTrackRecorderTest : public testing::Test {
+ public:
+  AudioTrackRecorderTest()
+      : params1_(kInputFormat,
+                 media::CHANNEL_LAYOUT_MONO,
+                 kSamplingRate,
+                 kBitsPerSample,
+                 kFramesPerBuffer),
+        params2_(kInputFormat,
+                 media::CHANNEL_LAYOUT_STEREO,
+                 kSamplingRate,
+                 kBitsPerSample,
+                 kFramesPerBuffer),
+        source_(kNumChannels, 440, kSamplingRate) {
+    PrepareBlinkTrackOfType(MEDIA_DEVICE_AUDIO_CAPTURE);
+    audio_track_recorder_.reset(new AudioTrackRecorder(
+        blink_track_, base::Bind(&AudioTrackRecorderTest::OnEncodedAudio,
+                                 base::Unretained(this))));
+  }
+
+  ~AudioTrackRecorderTest() {
+    audio_track_recorder_.reset();
+    blink_track_.setExtraData(nullptr);
+
+    // experiment with fixing leaks
+    delete audio_source_;
+    native_track_.reset();
+    adapter_ = nullptr;
+    capturer_ = nullptr;
+  }
+
+  scoped_ptr<media::AudioBus> NextAudioBus(const base::TimeDelta& duration) {
+    const int num_samples = static_cast<int>((kSamplingRate * duration) /
+                                             base::TimeDelta::FromSeconds(1));
+    scoped_ptr<media::AudioBus> bus(
+        media::AudioBus::Create(kNumChannels, num_samples));
+    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());
+    DoOnEncodedAudio(params, *encoded_data, timestamp);
+  }
+
+  const base::MessageLoop message_loop_;
+
+  // ATR and WebMediaStreamTrack for fooling it.
+  scoped_ptr<AudioTrackRecorder> audio_track_recorder_;

[miu, 2015/10/26 22:41:04]

nit: Don't need separate heap allocation here, just:

  AudioTrackRecorder audio_track_recorder_;

[ajose, 2015/10/27 18:41:28]

Moving ATR to the initialization list of the tests ctr (and changing
PrepareBlinkTrackOfType() to return a blink::WebMediaStreamTrack) is giving me
segfaults I haven't been able to resolve. I can keep looking at it.

+  blink::WebMediaStreamTrack blink_track_;
+
+  // test
+  blink::WebMediaConstraints constraints_;
+  scoped_refptr<WebRtcAudioCapturer> capturer_;
+  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
+  scoped_ptr<WebRtcLocalAudioTrack> native_track_;
+  blink::WebMediaStreamSource blink_audio_source_;
+  MediaStreamAudioSource* audio_source_;
+  // test
+
+  // Two different sets of AudioParameters for testing re-init of ATR.
+  media::AudioParameters params1_;
+  media::AudioParameters params2_;
+
+  // AudioSource for creating AudioBuses.
+  media::SineWaveAudioSource source_;
+
+ 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.
+  // Taken from media::SpeechRecognitionAudioSinkTest
+  void PrepareBlinkTrackOfType(const MediaStreamType device_type) {
+    StreamDeviceInfo device_info(device_type, "Mock device", "mock_device_id");
+    MockMediaConstraintFactory constraint_factory;
+    //const blink::WebMediaConstraints constraints =
+    constraints_ =
+        constraint_factory.CreateWebMediaConstraints();
+    //scoped_refptr<WebRtcAudioCapturer> capturer(
+    capturer_ =
+        WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints_, NULL,
+                                            NULL);
+    adapter_ =
+        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL);
+    native_track_ = make_scoped_ptr(
+        new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL));
+    //blink::WebMediaStreamSource blink_audio_source;
+    blink_audio_source_.initialize(base::UTF8ToUTF16("dummy_source_id"),
+                                  blink::WebMediaStreamSource::TypeAudio,
+                                  base::UTF8ToUTF16("dummy_source_name"),
+                                  false /* remote */, true /* readonly */);
+    MediaStreamSource::SourceStoppedCallback cb;
+    audio_source_ = new MediaStreamAudioSource(-1, device_info, cb, NULL);
+    blink_audio_source_.setExtraData(
+        //new MediaStreamAudioSource(-1, device_info, cb, NULL));
+        audio_source_);
+    blink_track_.initialize(blink::WebString::fromUTF8("dummy_track"),
+                            blink_audio_source_);
+    blink_track_.setExtraData(native_track_.release());
+  }
+
+  DISALLOW_COPY_AND_ASSIGN(AudioTrackRecorderTest);
+};
+
+TEST_F(AudioTrackRecorderTest, OnData) {
+  audio_track_recorder_->OnSetFormat(params1_);
+  InSequence s;
+  base::RunLoop run_loop;
+  base::Closure quit_closure = run_loop.QuitClosure();
+
+  // TODO(ajose): consider adding WillOnce(SaveArg...) and inspecting, as done
+  // in VTR unittests.
+  // TODO(ajose): Using 10ms chunks due to hard-coded 100fps framerate.
+  // Need to figure out what to do about framerate.
+  const base::TimeTicks time1 = base::TimeTicks::Now();
+  EXPECT_CALL(*this, DoOnEncodedAudio(_, _, time1)).Times(1);
+  audio_track_recorder_->OnData(
+      *NextAudioBus(base::TimeDelta::FromMilliseconds(10)), time1);
+
+  // Send more audio.
+  const base::TimeTicks time2 = base::TimeTicks::Now();
+  EXPECT_CALL(*this, DoOnEncodedAudio(_, _, _)).Times(1);
+  audio_track_recorder_->OnData(
+      *NextAudioBus(base::TimeDelta::FromMilliseconds(10)), time2);
+
+  // Send audio with different params to force ATR to re-init.
+  const base::TimeTicks time3 = base::TimeTicks::Now();
+  EXPECT_CALL(*this, DoOnEncodedAudio(_, _, _))
+      .Times(1)
+      .WillOnce(RunClosure(quit_closure));
+  audio_track_recorder_->OnData(
+      *NextAudioBus(base::TimeDelta::FromMilliseconds(10)), time3);
+
+  run_loop.Run();
+  Mock::VerifyAndClearExpectations(this);
+}
+
+}  // namespace content