Add AudioTrackRecorder for audio component of MediaStream recording.

content/renderer/media/audio_track_recorder_unittest.cc

Index: content/renderer/media/audio_track_recorder_unittest.cc
diff --git a/content/renderer/media/audio_track_recorder_unittest.cc b/content/renderer/media/audio_track_recorder_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..25de69104b2bbd97f05d9fe88e512b50c9ba5c84
--- /dev/null
+++ b/content/renderer/media/audio_track_recorder_unittest.cc
@@ -0,0 +1,176 @@
+// 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"
+#include "third_party/WebKit/public/web/WebHeap.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 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),
+        mono_source_(1 /* # channels */, 440, kSamplingRate),
+        stereo_source_(2 /* # channels */, 440, kSamplingRate) {
+    PrepareBlinkTrack();
+    audio_track_recorder_.reset(new AudioTrackRecorder(
+        blink_track_, base::Bind(&AudioTrackRecorderTest::OnEncodedAudio,
+                                 base::Unretained(this))));
+  }
+
+  ~AudioTrackRecorderTest() {
+    audio_track_recorder_.reset();
+    blink_track_.reset();
+    blink::WebHeap::collectAllGarbageForTesting();
+  }
+
+  scoped_ptr<media::AudioBus> NextAudioBus(
+      int num_channels, const base::TimeDelta& duration) {
+    // Only supports up to two channels for now.
+    EXPECT_GE(num_channels, 1);
+    EXPECT_LE(num_channels, 2);

[mcasas, 2015/10/28 20:03:00]

why not
EXPECT(num_channels == 1 || num_channels == 2)
?

[ajose, 2015/10/28 21:54:06]

Done.

+    const int num_samples = static_cast<int>((kSamplingRate * duration) /
+                                             base::TimeDelta::FromSeconds(1));
+    scoped_ptr<media::AudioBus> bus(
+        media::AudioBus::Create(num_channels, num_samples));
+    (num_channels == 1) ? mono_source_.OnMoreData(bus.get(), 0)
+                        : stereo_source_.OnMoreData(bus.get(), 0);

[mcasas, 2015/10/28 20:03:00]

I understand the code but I think it's the first time 
I see this in Chromium. Can you find other examples of
it in the code base? Otherwise revert to if-else plz.

[ajose, 2015/10/28 21:54:06]

Done.

+    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_;
+  blink::WebMediaStreamTrack blink_track_;
+
+  // Two different sets of AudioParameters for testing re-init of ATR.
+  media::AudioParameters params1_;
+  media::AudioParameters params2_;
+
+  // AudioSources for creating AudioBuses.
+  media::SineWaveAudioSource mono_source_;
+  media::SineWaveAudioSource stereo_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.
+  // 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_F(AudioTrackRecorderTest, OnData) {
+
+  InSequence s;
+  base::RunLoop run_loop;
+  base::Closure quit_closure = run_loop.QuitClosure();
+
+  // Give ATR initial audio parameters.
+  audio_track_recorder_->OnSetFormat(params1_);
+  // 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.

[mcasas, 2015/10/28 20:03:00]

TODO(s) need to have an associated crbug.com. 
Or they are just comments. Consider then a 
crbug.com to "improve ATRTest" for different
framerates and/or inspecting encoded results.,
and label it GoodFirstBug :)

[ajose, 2015/10/28 21:54:06]

SGTM. Will add a bug for the framerate stuff if it's not resolved soon.

+  const base::TimeTicks time1 = base::TimeTicks::Now();
+  EXPECT_CALL(*this, DoOnEncodedAudio(_, _, time1)).Times(1);
+  audio_track_recorder_->OnData(
+      *NextAudioBus(params1_.channels(),
+                    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(params1_.channels(),
+                    base::TimeDelta::FromMilliseconds(10)), time2);
+
+  // Give ATR new audio parameters.
+  audio_track_recorder_->OnSetFormat(params2_);
+  // 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(
+      *NextAudioBus(params2_.channels(),
+                    base::TimeDelta::FromMilliseconds(10)), time3);
+
+  run_loop.Run();
+  Mock::VerifyAndClearExpectations(this);
+}
+
+}  // namespace content