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 20%
rename from content/renderer/media/webrtc_audio_capturer_unittest.cc
rename to content/renderer/media/webrtc/processed_local_audio_source_unittest.cc
index 373b95ba50e17a28e962d3c769253082cb899ab2..3a2516e65d32ecbe8aaa9eb3408c40b0aa340c8b 100644
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc
+++ b/content/renderer/media/webrtc/processed_local_audio_source_unittest.cc
@@ -1,46 +1,41 @@
-// 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;
 
 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));
-
- protected:
-  ~MockCapturerSource() override {}
-};
+#if defined(OS_ANDROID)
+constexpr int kBufferSize = 960;  // Android works with a 20ms buffer size.

[o1ka, 2016/05/04 08:49:24]

The original comment is "bigger than 20ms" (and the truth seems to be "not less
than"). 
Nit: the comment is confusing since there is not sample rate to be seen around,
so maybe make sample rate a constant here as well?

[miu, 2016/05/04 22:10:09]

Done. Made the audio parameters more explicit with comments to help explain
requested vs. expected source format vs. expected output format. Also made this
more clear in other parts of this test code below (as I was resolving the
Anrdoid trybot failure).

[o1ka, 2016/05/06 16:53:57]

Thanks!

+#else
+constexpr int kBufferSize = 128;
+#endif
 
 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 +44,12 @@ 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 +57,152 @@ 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
+  ProcessedLocalAudioSourceTest()
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
-#endif
+                media::CHANNEL_LAYOUT_STEREO, 48000, 16, kBufferSize) {}
+
+  ~ProcessedLocalAudioSourceTest() override {}
+
+  void SetUp() override {
+    blink_audio_source_.initialize(blink::WebString::fromUTF8("audio_label"),
+                                   blink::WebMediaStreamSource::TypeAudio,
+                                   blink::WebString::fromUTF8("audio_track"),
+                                   false /* remote */, true /* readonly */);
+    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", params_.sample_rate(),
+                             params_.channel_layout(),
+                             params_.frames_per_buffer()),
+            &mock_dependency_factory_);
+    source->SetAllowInvalidRenderFrameIdForTesting(true);
+    source->SetSourceConstraints(constraints);
+    blink_audio_source_.setExtraData(source);  // Takes ownership.
+  }
+
+  void CheckAudioParametersMatch(const media::AudioParameters& params) {
+    EXPECT_TRUE(params_.Equals(params));
+  }
+
+  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_;
+  const media::AudioParameters params_;
+  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) {
+  // 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));
+  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()));
+  CheckAudioParametersMatch(audio_source()->GetAudioParameters());
+
+  // Connect a sink to the track.
+  std::unique_ptr<MockMediaStreamAudioSink> sink(
+      new MockMediaStreamAudioSink());
+  using ThisTest =
+      ProcessedLocalAudioSourceTest_VerifyAudioFlowWithoutAudioProcessing_Test;
+  EXPECT_CALL(*sink, FormatIsSet(_))
+      .WillOnce(Invoke(this, &ThisTest::CheckAudioParametersMatch));
+  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(audio_source()->GetAudioParameters());
+  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