MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/media_stream_audio_unittest.cc

Index: content/renderer/media/media_stream_audio_unittest.cc
diff --git a/content/renderer/media/media_stream_audio_unittest.cc b/content/renderer/media/media_stream_audio_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..060dca37b0706d0276decba38199d7792efaf851
--- /dev/null
+++ b/content/renderer/media/media_stream_audio_unittest.cc
@@ -0,0 +1,450 @@
+// 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 <stdint.h>
+
+#include "base/atomicops.h"
+#include "base/synchronization/lock.h"
+#include "base/synchronization/waitable_event.h"
+#include "base/test/test_timeouts.h"
+#include "base/threading/platform_thread.h"
+#include "base/threading/thread_checker.h"
+#include "content/public/renderer/media_stream_audio_sink.h"
+#include "content/renderer/media/media_stream_audio_source.h"
+#include "content/renderer/media/media_stream_audio_track.h"
+#include "media/base/audio_bus.h"
+#include "media/base/audio_parameters.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "third_party/WebKit/public/platform/WebString.h"
+#include "third_party/WebKit/public/web/WebHeap.h"
+
+namespace content {
+
+namespace {
+
+constexpr int kSampleRate = 8000;
+constexpr int kBufferSize = kSampleRate / 100;
+
+// The maximum integer that can be exactly represented by the float data type.
+constexpr int kMaxValueSafelyConvertableToFloat = 1 << 24;
+
+// A simple MediaStreamAudioSource that spawns a real-time audio thread and
+// emits audio samples with monotonically-increasing sample values. Includes
+// hooks for the unit tests to confirm lifecycle status and to change audio
+// format.
+class FakeMediaStreamAudioSource
+    : public MediaStreamAudioSource,
+      public base::PlatformThread::Delegate {
+ public:
+  FakeMediaStreamAudioSource()
+      : MediaStreamAudioSource(true), stop_event_(true, false),
+        next_buffer_size_(kBufferSize), sample_count_(0) {}
+
+  ~FakeMediaStreamAudioSource() final {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    EnsureSourceIsStopped();
+  }
+
+  bool was_started() const {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    return !thread_.is_null();
+  }
+
+  bool was_stopped() const {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    return stop_event_.IsSignaled();
+  }
+
+  void SetBufferSize(int new_buffer_size) {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    base::subtle::NoBarrier_Store(&next_buffer_size_, new_buffer_size);
+  }
+
+ protected:
+  bool EnsureSourceIsStarted() final {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    if (was_started())
+      return true;
+    if (was_stopped())
+      return false;
+    base::PlatformThread::CreateWithPriority(
+        0, this, &thread_, base::ThreadPriority::REALTIME_AUDIO);
+    return true;
+  }
+
+  void EnsureSourceIsStopped() final {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    if (was_stopped())
+      return;
+    stop_event_.Signal();
+    if (was_started())
+      base::PlatformThread::Join(thread_);
+  }
+
+  void ThreadMain() override {
+    while (!stop_event_.IsSignaled()) {
+      // If needed, notify of the new format and re-create |audio_bus_|.
+      const int buffer_size = base::subtle::NoBarrier_Load(&next_buffer_size_);
+      if (!audio_bus_ || audio_bus_->frames() != buffer_size) {
+        MediaStreamAudioSource::SetFormat(media::AudioParameters(
+            media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+            media::CHANNEL_LAYOUT_MONO, kSampleRate, 16, buffer_size));
+        audio_bus_ = media::AudioBus::Create(1, buffer_size);
+      }
+
+      // Deliver the next chunk of audio data. Each sample value is its offset
+      // from the very first sample.
+      float* const data = audio_bus_->channel(0);
+      for (int i = 0; i < buffer_size; ++i)
+        data[i] = ++sample_count_;
+      CHECK_LT(sample_count_, kMaxValueSafelyConvertableToFloat);
+      MediaStreamAudioSource::DeliverDataToTracks(*audio_bus_,
+                                                  base::TimeTicks::Now());
+
+      // Sleep before producing the next chunk of audio.
+      base::PlatformThread::Sleep(base::TimeDelta::FromMicroseconds(
+          base::Time::kMicrosecondsPerSecond * buffer_size / kSampleRate));
+    }
+  }
+
+ private:
+  base::ThreadChecker main_thread_checker_;
+
+  base::PlatformThreadHandle thread_;
+  mutable base::WaitableEvent stop_event_;
+
+  base::subtle::Atomic32 next_buffer_size_;
+  std::unique_ptr<media::AudioBus> audio_bus_;
+  int sample_count_;
+
+  DISALLOW_COPY_AND_ASSIGN(FakeMediaStreamAudioSource);
+};
+
+// A simple MediaStreamAudioSink that consumes audio and confirms the sample
+// values. Includes hooks for the unit tests to monitor the format and flow of
+// audio, whether the audio is silent, and the propagation of the "enabled"
+// state.
+class FakeMediaStreamAudioSink : public MediaStreamAudioSink {
+ public:
+  enum EnableState {
+    NO_ENABLE_NOTIFICATION,
+    WAS_ENABLED,
+    WAS_DISABLED
+  };
+
+  FakeMediaStreamAudioSink()
+      : MediaStreamAudioSink(), expected_sample_count_(-1),
+        num_on_data_calls_(0), audio_is_silent_(true), was_ended_(false),
+        enable_state_(NO_ENABLE_NOTIFICATION) {}
+
+  ~FakeMediaStreamAudioSink() final {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+  }
+
+  media::AudioParameters params() const {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    base::AutoLock auto_lock(params_lock_);
+    return params_;
+  }
+
+  int num_on_data_calls() const {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    return base::subtle::NoBarrier_Load(&num_on_data_calls_);
+  }
+
+  bool is_audio_silent() const {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    return !!base::subtle::NoBarrier_Load(&audio_is_silent_);
+  }
+
+  bool was_ended() const {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    return was_ended_;
+  }
+
+  EnableState enable_state() const {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    return enable_state_;
+  }
+
+  void OnSetFormat(const media::AudioParameters& params) final {
+    ASSERT_TRUE(params.IsValid());
+    base::AutoLock auto_lock(params_lock_);
+    params_ = params;
+  }
+
+  void OnData(const media::AudioBus& audio_bus,
+              base::TimeTicks estimated_capture_time) final {
+    ASSERT_TRUE(params_.IsValid());
+    ASSERT_FALSE(was_ended_);
+
+    ASSERT_EQ(params_.channels(), audio_bus.channels());
+    ASSERT_EQ(params_.frames_per_buffer(), audio_bus.frames());
+    if (audio_bus.AreFramesZero()) {
+      base::subtle::NoBarrier_Store(&audio_is_silent_, 1);
+      expected_sample_count_ = -1;  // Reset for when audio comes back.
+    } else {
+      base::subtle::NoBarrier_Store(&audio_is_silent_, 0);
+      const float* const data = audio_bus.channel(0);
+      if (expected_sample_count_ == -1)
+        expected_sample_count_ = static_cast<int64_t>(data[0]);
+      CHECK_LE(expected_sample_count_ + audio_bus.frames(),
+               kMaxValueSafelyConvertableToFloat);
+      for (int i = 0; i < audio_bus.frames(); ++i) {
+        const float expected_sample_value = expected_sample_count_;
+        ASSERT_EQ(expected_sample_value, data[i]);
+        ++expected_sample_count_;
+      }
+    }
+
+    ASSERT_TRUE(!estimated_capture_time.is_null());
+    ASSERT_LT(last_estimated_capture_time_, estimated_capture_time);
+    last_estimated_capture_time_ = estimated_capture_time;
+
+    base::subtle::NoBarrier_AtomicIncrement(&num_on_data_calls_, 1);
+  }
+
+  void OnReadyStateChanged(
+      blink::WebMediaStreamSource::ReadyState state) final {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    if (state == blink::WebMediaStreamSource::ReadyStateEnded)
+      was_ended_ = true;
+  }
+
+  void OnEnabledChanged(bool enabled) final {
+    CHECK(main_thread_checker_.CalledOnValidThread());
+    enable_state_ = enabled ? WAS_ENABLED : WAS_DISABLED;
+  }
+
+ private:
+  base::ThreadChecker main_thread_checker_;
+
+  mutable base::Lock params_lock_;
+  media::AudioParameters params_;
+  int expected_sample_count_;
+  base::TimeTicks last_estimated_capture_time_;
+  base::subtle::Atomic32 num_on_data_calls_;
+  base::subtle::Atomic32 audio_is_silent_;
+  bool was_ended_;
+  EnableState enable_state_;
+
+  DISALLOW_COPY_AND_ASSIGN(FakeMediaStreamAudioSink);
+};
+
+}  // namespace
+
+class MediaStreamAudioTest : public ::testing::Test {
+ protected:
+  void SetUp() override {
+    blink_audio_source_.initialize(blink::WebString::fromUTF8("audio_id"),
+                                   blink::WebMediaStreamSource::TypeAudio,
+                                   blink::WebString::fromUTF8("audio_track"),
+                                   false /* remote */);
+    blink_audio_track_.initialize(blink_audio_source_.id(),
+                                  blink_audio_source_);
+  }
+
+  void TearDown() override {
+    blink_audio_track_.reset();
+    blink_audio_source_.reset();
+    blink::WebHeap::collectAllGarbageForTesting();
+  }
+
+  FakeMediaStreamAudioSource* source() const {
+    return static_cast<FakeMediaStreamAudioSource*>(
+        MediaStreamAudioSource::From(blink_audio_source_));
+  }
+
+  MediaStreamAudioTrack* track() const {
+    return MediaStreamAudioTrack::From(blink_audio_track_);
+  }
+
+  blink::WebMediaStreamSource blink_audio_source_;
+  blink::WebMediaStreamTrack blink_audio_track_;
+};
+
+// Tests that a simple source-->track-->sink connection and audio data flow
+// works.
+TEST_F(MediaStreamAudioTest, BasicUsage) {
+  // Create the source, but it should not be started yet.
+  ASSERT_FALSE(source());
+  blink_audio_source_.setExtraData(new FakeMediaStreamAudioSource());
+  ASSERT_TRUE(source());
+  EXPECT_FALSE(source()->was_started());
+  EXPECT_FALSE(source()->was_stopped());
+
+  // Connect a track to the source. This should auto-start the source.
+  ASSERT_FALSE(track());
+  EXPECT_TRUE(source()->ConnectToTrack(blink_audio_track_));
+  ASSERT_TRUE(track());
+  EXPECT_TRUE(source()->was_started());
+  EXPECT_FALSE(source()->was_stopped());
+
+  // Connect a sink to the track. This should begin audio flow to the
+  // sink. Wait and confirm that three OnData() calls were made from the audio
+  // thread.
+  FakeMediaStreamAudioSink sink;
+  EXPECT_FALSE(sink.was_ended());
+  track()->AddSink(&sink);
+  const int start_count = sink.num_on_data_calls();
+  while (sink.num_on_data_calls() - start_count < 3)
+    base::PlatformThread::Sleep(TestTimeouts::tiny_timeout());
+
+  // Check that the audio parameters propagated to the track and sink.
+  const media::AudioParameters expected_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY, media::CHANNEL_LAYOUT_MONO,
+      kSampleRate, 16, kBufferSize);
+  EXPECT_TRUE(expected_params.Equals(track()->GetOutputFormat()));
+  EXPECT_TRUE(expected_params.Equals(sink.params()));
+
+  // Stop the track. Since this was the last track connected to the source, the
+  // source should automatically stop. In addition, the sink should receive a
+  // ReadyStateEnded notification.
+  track()->Stop();
+  EXPECT_TRUE(source()->was_started());
+  EXPECT_TRUE(source()->was_stopped());
+  EXPECT_TRUE(sink.was_ended());
+
+  track()->RemoveSink(&sink);
+}
+
+// Tests that "ended" tracks can be connected after the source has stopped.
+TEST_F(MediaStreamAudioTest, ConnectTrackAfterSourceStopped) {
+  // Create the source, connect one track, and stop it. This should
+  // automatically stop the source.
+  blink_audio_source_.setExtraData(new FakeMediaStreamAudioSource());
+  ASSERT_TRUE(source());
+  EXPECT_TRUE(source()->ConnectToTrack(blink_audio_track_));
+  track()->Stop();
+  EXPECT_TRUE(source()->was_started());
+  EXPECT_TRUE(source()->was_stopped());
+
+  // Now, connect another track. ConnectToTrack() will return false, but there
+  // should be a MediaStreamAudioTrack instance created and owned by the
+  // blink::WebMediaStreamTrack.
+  blink::WebMediaStreamTrack another_blink_track;
+  another_blink_track.initialize(blink_audio_source_.id(), blink_audio_source_);
+  EXPECT_FALSE(MediaStreamAudioTrack::From(another_blink_track));
+  EXPECT_FALSE(source()->ConnectToTrack(another_blink_track));
+  EXPECT_TRUE(MediaStreamAudioTrack::From(another_blink_track));
+}
+
+// Tests that a sink is immediately "ended" when connected to a stopped track.
+TEST_F(MediaStreamAudioTest, AddSinkToStoppedTrack) {
+  // Create a track and stop it. Then, when adding a sink, the sink should get
+  // the ReadyStateEnded notification immediately.
+  MediaStreamAudioTrack track(true);
+  track.Stop();
+  FakeMediaStreamAudioSink sink;
+  EXPECT_FALSE(sink.was_ended());
+  track.AddSink(&sink);
+  EXPECT_TRUE(sink.was_ended());
+  EXPECT_EQ(0, sink.num_on_data_calls());
+  track.RemoveSink(&sink);
+}
+
+// Tests that audio format changes at the source propagate to the track and
+// sink.
+TEST_F(MediaStreamAudioTest, FormatChangesPropagate) {
+  // Create a source, connect it to track, and connect the track to a
+  // sink.
+  blink_audio_source_.setExtraData(new FakeMediaStreamAudioSource());
+  ASSERT_TRUE(source());
+  EXPECT_TRUE(source()->ConnectToTrack(blink_audio_track_));
+  ASSERT_TRUE(track());
+  FakeMediaStreamAudioSink sink;
+  ASSERT_TRUE(!sink.params().IsValid());
+  track()->AddSink(&sink);
+
+  // Wait until valid parameters are propagated to the sink, and then confirm
+  // the parameters are correct at the track and the sink.
+  while (!sink.params().IsValid())
+    base::PlatformThread::Sleep(TestTimeouts::tiny_timeout());
+  const media::AudioParameters expected_params(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY, media::CHANNEL_LAYOUT_MONO,
+      kSampleRate, 16, kBufferSize);
+  EXPECT_TRUE(expected_params.Equals(track()->GetOutputFormat()));
+  EXPECT_TRUE(expected_params.Equals(sink.params()));
+
+  // Now, trigger a format change by doubling the buffer size.
+  source()->SetBufferSize(kBufferSize * 2);
+
+  // Wait until the new buffer size propagates to the sink.
+  while (sink.params().frames_per_buffer() == kBufferSize)
+    base::PlatformThread::Sleep(TestTimeouts::tiny_timeout());
+  EXPECT_EQ(kBufferSize * 2, track()->GetOutputFormat().frames_per_buffer());
+  EXPECT_EQ(kBufferSize * 2, sink.params().frames_per_buffer());
+
+  track()->RemoveSink(&sink);
+}
+
+// Tests that tracks deliver audio when enabled and silent audio when
+// disabled. Whenever a track is enabled or disabled, the sink's
+// OnEnabledChanged() method should be called.
+TEST_F(MediaStreamAudioTest, EnableAndDisableTracks) {
+  // Create a source and connect it to track.
+  blink_audio_source_.setExtraData(new FakeMediaStreamAudioSource());
+  ASSERT_TRUE(source());
+  EXPECT_TRUE(source()->ConnectToTrack(blink_audio_track_));
+  ASSERT_TRUE(track());
+
+  // Connect the track to a sink and expect the sink to be notified that the
+  // track is enabled.
+  FakeMediaStreamAudioSink sink;
+  EXPECT_TRUE(sink.is_audio_silent());
+  EXPECT_EQ(FakeMediaStreamAudioSink::NO_ENABLE_NOTIFICATION,
+            sink.enable_state());
+  track()->AddSink(&sink);
+  EXPECT_EQ(FakeMediaStreamAudioSink::WAS_ENABLED, sink.enable_state());
+
+  // Wait until non-silent audio reaches the sink.
+  while (sink.is_audio_silent())
+    base::PlatformThread::Sleep(TestTimeouts::tiny_timeout());
+
+  // Now, disable the track and expect the sink to be notified.
+  track()->SetEnabled(false);
+  EXPECT_EQ(FakeMediaStreamAudioSink::WAS_DISABLED, sink.enable_state());
+
+  // Wait until silent audio reaches the sink.
+  while (!sink.is_audio_silent())
+    base::PlatformThread::Sleep(TestTimeouts::tiny_timeout());
+
+  // Create a second track and a second sink, but this time the track starts out
+  // disabled. Expect the sink to be notified at the start that the track is
+  // disabled.
+  blink::WebMediaStreamTrack another_blink_track;
+  another_blink_track.initialize(blink_audio_source_.id(), blink_audio_source_);
+  EXPECT_TRUE(source()->ConnectToTrack(another_blink_track));
+  MediaStreamAudioTrack::From(another_blink_track)->SetEnabled(false);
+  FakeMediaStreamAudioSink another_sink;
+  MediaStreamAudioTrack::From(another_blink_track)->AddSink(&another_sink);
+  EXPECT_EQ(FakeMediaStreamAudioSink::WAS_DISABLED,
+            another_sink.enable_state());
+
+  // Wait until OnData() is called on the second sink. Expect the audio to be
+  // silent.
+  const int start_count = another_sink.num_on_data_calls();
+  while (another_sink.num_on_data_calls() == start_count)
+    base::PlatformThread::Sleep(TestTimeouts::tiny_timeout());
+  EXPECT_TRUE(another_sink.is_audio_silent());
+
+  // Now, enable the second track and expect the second sink to be notified.
+  MediaStreamAudioTrack::From(another_blink_track)->SetEnabled(true);
+  EXPECT_EQ(FakeMediaStreamAudioSink::WAS_ENABLED, another_sink.enable_state());
+
+  // Wait until non-silent audio reaches the second sink.
+  while (another_sink.is_audio_silent())
+    base::PlatformThread::Sleep(TestTimeouts::tiny_timeout());
+
+  // The first track and sink should not have been affected by changing the
+  // enabled state of the second track and sink. They should still be disabled,
+  // with silent audio being consumed at the sink.
+  EXPECT_EQ(FakeMediaStreamAudioSink::WAS_DISABLED, sink.enable_state());
+  EXPECT_TRUE(sink.is_audio_silent());
+
+  MediaStreamAudioTrack::From(another_blink_track)->RemoveSink(&another_sink);
+  track()->RemoveSink(&sink);
+}
+
+}  // namespace content