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

content/renderer/media/webrtc/processed_local_audio_source_unittest.cc

Index: content/renderer/media/webrtc/processed_local_audio_source_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_capturer_unittest.cc b/content/renderer/media/webrtc/processed_local_audio_source_unittest.cc
similarity index 18%
rename from content/renderer/media/webrtc_audio_capturer_unittest.cc
rename to content/renderer/media/webrtc/processed_local_audio_source_unittest.cc
index 373b95ba50e17a28e962d3c769253082cb899ab2..0abec8ea56dd6deea7cbc541260bd3d793c1cf8a 100644
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc
+++ b/content/renderer/media/webrtc/processed_local_audio_source_unittest.cc
@@ -1,46 +1,54 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// 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 "base/logging.h"
 #include "build/build_config.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
+#include "content/renderer/media/media_stream_audio_track.h"
+#include "content/renderer/media/mock_audio_device_factory.h"
 #include "content/renderer/media/mock_constraint_factory.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
+#include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_parameters.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
+#include "third_party/WebKit/public/web/WebHeap.h"
 
 using ::testing::_;
 using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::WithArg;
 
 namespace content {
 
 namespace {
 
-class MockCapturerSource : public media::AudioCapturerSource {
- public:
-  MockCapturerSource() {}
-  MOCK_METHOD3(Initialize, void(const media::AudioParameters& params,
-                                CaptureCallback* callback,
-                                int session_id));
-  MOCK_METHOD0(Start, void());
-  MOCK_METHOD0(Stop, void());
-  MOCK_METHOD1(SetVolume, void(double volume));
-  MOCK_METHOD1(SetAutomaticGainControl, void(bool enable));
+// Audio parameters for the VerifyAudioFlowWithoutAudioProcessing test.
+constexpr int kSampleRate = 48000;
+constexpr media::ChannelLayout kChannelLayout = media::CHANNEL_LAYOUT_STEREO;
+constexpr int kRequestedBufferSize = 512;
 
- protected:
-  ~MockCapturerSource() override {}
-};
+// On Android, ProcessedLocalAudioSource forces a 20ms buffer size from the
+// input device.
+#if defined(OS_ANDROID)
+constexpr int kExpectedSourceBufferSize = kSampleRate / 50;
+#else
+constexpr int kExpectedSourceBufferSize = kRequestedBufferSize;
+#endif
+
+// On both platforms, even though audio processing is turned off, the
+// MediaStreamAudioProcessor will force the use of 10ms buffer sizes on the
+// output end of its FIFO.
+constexpr int kExpectedOutputBufferSize = kSampleRate / 100;
 
 class MockMediaStreamAudioSink : public MediaStreamAudioSink {
  public:
   MockMediaStreamAudioSink() {}
   ~MockMediaStreamAudioSink() override {}
+
   void OnData(const media::AudioBus& audio_bus,
               base::TimeTicks estimated_capture_time) override {
     EXPECT_EQ(audio_bus.channels(), params_.channels());
@@ -49,11 +57,12 @@ class MockMediaStreamAudioSink : public MediaStreamAudioSink {
     OnDataCallback();
   }
   MOCK_METHOD0(OnDataCallback, void());
+
   void OnSetFormat(const media::AudioParameters& params) override {
     params_ = params;
-    FormatIsSet();
+    FormatIsSet(params_);
   }
-  MOCK_METHOD0(FormatIsSet, void());
+  MOCK_METHOD1(FormatIsSet, void(const media::AudioParameters& params));
 
  private:
   media::AudioParameters params_;
@@ -61,92 +70,158 @@ class MockMediaStreamAudioSink : public MediaStreamAudioSink {
 
 }  // namespace
 
-class WebRtcAudioCapturerTest : public testing::Test {
+class ProcessedLocalAudioSourceTest : public testing::Test {
  protected:
-  WebRtcAudioCapturerTest()
-#if defined(OS_ANDROID)
-      : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) {
-    // Android works with a buffer size bigger than 20ms.
-#else
-      : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
-#endif
+  ProcessedLocalAudioSourceTest() {}
+
+  ~ProcessedLocalAudioSourceTest() override {}
+
+  void SetUp() override {
+    blink_audio_source_.initialize(blink::WebString::fromUTF8("audio_label"),
+                                   blink::WebMediaStreamSource::TypeAudio,
+                                   blink::WebString::fromUTF8("audio_track"),
+                                   false /* remote */);
+    blink_audio_track_.initialize(blink_audio_source_.id(),
+                                  blink_audio_source_);
   }
 
-  void VerifyAudioParams(const blink::WebMediaConstraints& constraints,
-                         bool need_audio_processing) {
-    const std::unique_ptr<WebRtcAudioCapturer> capturer =
-        WebRtcAudioCapturer::CreateCapturer(
-            -1, StreamDeviceInfo(
-                    MEDIA_DEVICE_AUDIO_CAPTURE, "", "", params_.sample_rate(),
-                    params_.channel_layout(), params_.frames_per_buffer()),
-            constraints, nullptr, nullptr);
-    const scoped_refptr<MockCapturerSource> capturer_source(
-        new MockCapturerSource());
-    EXPECT_CALL(*capturer_source.get(), Initialize(_, capturer.get(), -1));
-    EXPECT_CALL(*capturer_source.get(), SetAutomaticGainControl(true));
-    EXPECT_CALL(*capturer_source.get(), Start());
-    capturer->SetCapturerSource(capturer_source, params_);
-
-    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-    const std::unique_ptr<WebRtcLocalAudioTrack> track(
-        new WebRtcLocalAudioTrack(adapter.get()));
-    capturer->AddTrack(track.get());
-
-    // Connect a mock sink to the track.
-    std::unique_ptr<MockMediaStreamAudioSink> sink(
-        new MockMediaStreamAudioSink());
-    track->AddSink(sink.get());
-
-    int delay_ms = 65;
-    bool key_pressed = true;
-    double volume = 0.9;
-
-    std::unique_ptr<media::AudioBus> audio_bus =
-        media::AudioBus::Create(params_);
-    audio_bus->Zero();
-
-    media::AudioCapturerSource::CaptureCallback* callback =
-        static_cast<media::AudioCapturerSource::CaptureCallback*>(
-            capturer.get());
-
-    // Verify the sink is getting the correct values.
-    EXPECT_CALL(*sink, FormatIsSet());
-    EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1));
-    callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed);
-
-    track->RemoveSink(sink.get());
-    EXPECT_CALL(*capturer_source.get(), Stop());
-    capturer->Stop();
+  void TearDown() override {
+    blink_audio_track_.reset();
+    blink_audio_source_.reset();
+    blink::WebHeap::collectAllGarbageForTesting();
   }
 
-  media::AudioParameters params_;
+  void CreateProcessedLocalAudioSource(
+      const blink::WebMediaConstraints& constraints) {
+    ProcessedLocalAudioSource* const source =
+      new ProcessedLocalAudioSource(
+            -1 /* consumer_render_frame_id is N/A for non-browser tests */,
+            StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "Mock audio device",
+                             "mock_audio_device_id", kSampleRate,
+                             kChannelLayout, kRequestedBufferSize),
+            &mock_dependency_factory_);
+    source->SetAllowInvalidRenderFrameIdForTesting(true);
+    source->SetSourceConstraints(constraints);
+    blink_audio_source_.setExtraData(source);  // Takes ownership.
+  }
+
+  void CheckSourceFormatMatches(const media::AudioParameters& params) {
+    EXPECT_EQ(kSampleRate, params.sample_rate());
+    EXPECT_EQ(kChannelLayout, params.channel_layout());
+    EXPECT_EQ(kExpectedSourceBufferSize, params.frames_per_buffer());
+  }
+
+  void CheckOutputFormatMatches(const media::AudioParameters& params) {
+    EXPECT_EQ(kSampleRate, params.sample_rate());
+    EXPECT_EQ(kChannelLayout, params.channel_layout());
+    EXPECT_EQ(kExpectedOutputBufferSize, params.frames_per_buffer());
+  }
+
+  MockAudioDeviceFactory* mock_audio_device_factory() {
+    return &mock_audio_device_factory_;
+  }
+
+  media::AudioCapturerSource::CaptureCallback* capture_source_callback() const {
+    return static_cast<media::AudioCapturerSource::CaptureCallback*>(
+        ProcessedLocalAudioSource::From(audio_source()));
+  }
+
+  MediaStreamAudioSource* audio_source() const {
+    return MediaStreamAudioSource::From(blink_audio_source_);
+  }
+
+  const blink::WebMediaStreamTrack& blink_audio_track() {
+    return blink_audio_track_;
+  }
+
+ private:
+  MockAudioDeviceFactory mock_audio_device_factory_;
+  MockPeerConnectionDependencyFactory mock_dependency_factory_;
+  blink::WebMediaStreamSource blink_audio_source_;
+  blink::WebMediaStreamTrack blink_audio_track_;
 };
 
-TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) {
-  // Turn off the default constraints to verify that the sink will get packets
-  // with a buffer size smaller than 10ms.
+// Tests a basic end-to-end start-up, track+sink connections, audio flow, and
+// shut-down. The unit tests in media_stream_audio_unittest.cc provide more
+// comprehensive testing of the object graph connections and multi-threading
+// concerns.
+TEST_F(ProcessedLocalAudioSourceTest, VerifyAudioFlowWithoutAudioProcessing) {
+  using ThisTest =
+      ProcessedLocalAudioSourceTest_VerifyAudioFlowWithoutAudioProcessing_Test;
+
+  // Turn off the default constraints so the sink will get audio in chunks of
+  // the native buffer size.
   MockConstraintFactory constraint_factory;
   constraint_factory.DisableDefaultAudioConstraints();
-  VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false);
+
+  CreateProcessedLocalAudioSource(
+      constraint_factory.CreateWebMediaConstraints());
+
+  // Connect the track, and expect the MockCapturerSource to be initialized and
+  // started by ProcessedLocalAudioSource.
+  EXPECT_CALL(*mock_audio_device_factory()->mock_capturer_source(),
+              Initialize(_, capture_source_callback(), -1))
+      .WillOnce(WithArg<0>(Invoke(this, &ThisTest::CheckSourceFormatMatches)));
+  EXPECT_CALL(*mock_audio_device_factory()->mock_capturer_source(),
+              SetAutomaticGainControl(true));
+  EXPECT_CALL(*mock_audio_device_factory()->mock_capturer_source(), Start());
+  ASSERT_TRUE(audio_source()->ConnectToTrack(blink_audio_track()));
+  CheckOutputFormatMatches(audio_source()->GetAudioParameters());
+
+  // Connect a sink to the track.
+  std::unique_ptr<MockMediaStreamAudioSink> sink(
+      new MockMediaStreamAudioSink());
+  EXPECT_CALL(*sink, FormatIsSet(_))
+      .WillOnce(Invoke(this, &ThisTest::CheckOutputFormatMatches));
+  MediaStreamAudioTrack::From(blink_audio_track())->AddSink(sink.get());
+
+  // Feed audio data into the ProcessedLocalAudioSource and expect it to reach
+  // the sink.
+  int delay_ms = 65;
+  bool key_pressed = true;
+  double volume = 0.9;
+  std::unique_ptr<media::AudioBus> audio_bus =
+      media::AudioBus::Create(2, kExpectedSourceBufferSize);
+  audio_bus->Zero();
+  EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1));
+  capture_source_callback()->Capture(audio_bus.get(), delay_ms, volume,
+                                     key_pressed);
+
+  // Expect the ProcessedLocalAudioSource to auto-stop the MockCapturerSource
+  // when the track is stopped.
+  EXPECT_CALL(*mock_audio_device_factory()->mock_capturer_source(), Stop());
+  MediaStreamAudioTrack::From(blink_audio_track())->Stop();
 }
 
-TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) {
+// Tests that the source is not started when invalid audio constraints are
+// present.
+TEST_F(ProcessedLocalAudioSourceTest, FailToStartWithWrongConstraints) {
   MockConstraintFactory constraint_factory;
   const std::string dummy_constraint = "dummy";
   // Set a non-audio constraint.
   constraint_factory.basic().width.setExact(240);
 
-  std::unique_ptr<WebRtcAudioCapturer> capturer(
-      WebRtcAudioCapturer::CreateCapturer(
-          0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "",
-                              params_.sample_rate(), params_.channel_layout(),
-                              params_.frames_per_buffer()),
-          constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
-  EXPECT_TRUE(capturer.get() == NULL);
+  CreateProcessedLocalAudioSource(
+      constraint_factory.CreateWebMediaConstraints());
+
+  // Expect the MockCapturerSource is never initialized/started and the
+  // ConnectToTrack() operation fails due to the invalid constraint.
+  EXPECT_CALL(*mock_audio_device_factory()->mock_capturer_source(),
+              Initialize(_, capture_source_callback(), -1))
+      .Times(0);
+  EXPECT_CALL(*mock_audio_device_factory()->mock_capturer_source(),
+              SetAutomaticGainControl(true)).Times(0);
+  EXPECT_CALL(*mock_audio_device_factory()->mock_capturer_source(), Start())
+      .Times(0);
+  EXPECT_FALSE(audio_source()->ConnectToTrack(blink_audio_track()));
+
+  // Even though ConnectToTrack() failed, there should still have been a new
+  // MediaStreamAudioTrack instance created, owned by the
+  // blink::WebMediaStreamTrack.
+  EXPECT_TRUE(MediaStreamAudioTrack::From(blink_audio_track()));
 }
 
+// TODO(miu): There's a lot of logic in ProcessedLocalAudioSource around
+// constraints processing and validation that should have unit testing.
 
 }  // namespace content