Fix/Add content_unittests to work for the code behind --enable-audio-track-processing

content/renderer/media/webrtc_local_audio_track_unittest.cc

 // Copyright 2013 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/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.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_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_capturer_source.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ skipping 110 matching lines @@
   ~MockMediaStreamAudioSink() {}
   int OnData(const int16* audio_data,
              int sample_rate,
              int number_of_channels,
              int number_of_frames,
              const std::vector<int>& channels,
              int audio_delay_milliseconds,
              int current_volume,
              bool need_audio_processing,
              bool key_pressed) OVERRIDE {
+    EXPECT_EQ(params_.sample_rate(), sample_rate);
+    EXPECT_EQ(params_.channels(), number_of_channels);
+    EXPECT_EQ(params_.frames_per_buffer(), number_of_frames);
     CaptureData(channels.size(),
-                sample_rate,
-                number_of_channels,
-                number_of_frames,
                 audio_delay_milliseconds,
                 current_volume,
                 need_audio_processing,
                 key_pressed);
     return 0;
   }
-  MOCK_METHOD8(CaptureData,
+  MOCK_METHOD5(CaptureData,
                void(int number_of_network_channels,
-                    int sample_rate,
-                    int number_of_channels,
-                    int number_of_frames,
                     int audio_delay_milliseconds,
                     int current_volume,
                     bool need_audio_processing,
                     bool key_pressed));
-  MOCK_METHOD1(OnSetFormat, void(const media::AudioParameters& params));
+  void OnSetFormat(const media::AudioParameters& params) {
+    params_ = params;
+    FormatIsSet();
+  }
+  MOCK_METHOD0(FormatIsSet, void());
+
+  const media::AudioParameters& audio_params() const { return params_; }
+
+ private:
+  media::AudioParameters params_;
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
     blink::WebMediaConstraints constraints;
@@ skipping 28 matching lines @@
   track->Start();
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
 
   // Connect a number of network channels to the audio track.
   static const int kNumberOfNetworkChannels = 4;
   for (int i = 0; i < kNumberOfNetworkChannels; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(
         adapter.get())->GetRenderer()->AddChannel(i);
   }
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
-  const media::AudioParameters params = capturer_->source_audio_parameters();
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(Return());
+  EXPECT_CALL(*sink, FormatIsSet());
   EXPECT_CALL(*sink,
       CaptureData(kNumberOfNetworkChannels,
-                  params.sample_rate(),
-                  params.channels(),
-                  params.sample_rate() / 100,
                   0,
                   0,
-                  true,
+                  _,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
-
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
 
   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
   capturer_->Stop();
 }
 
 // The same setup as ConnectAndDisconnectOneSink, but enable and disable the
 // audio track on the fly. When the audio track is disabled, there is no data
 // callback to the sink; when the audio track is enabled, there comes data
@@ skipping 10 matching lines @@
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
   static_cast<webrtc::AudioTrackInterface*>(
       adapter.get())->GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track->GetAudioAdapter()->enabled());
   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(false));
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->source_audio_parameters();
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, OnSetFormat(_)).Times(1);
+  EXPECT_CALL(*sink, FormatIsSet()).Times(1);
   EXPECT_CALL(*sink,
-              CaptureData(1,
-                          params.sample_rate(),
-                          params.channels(),
-                          params.sample_rate() / 100,
-                          0,
-                          0,
-                          true,
-                          false)).Times(0);
+              CaptureData(1, 0, 0, _, false)).Times(0);
+  EXPECT_EQ(sink->audio_params().frames_per_buffer(),
+            params.sample_rate() / 100);
   track->AddSink(sink.get());
   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   event.Reset();
   EXPECT_CALL(*sink,
-              CaptureData(1,
-                          params.sample_rate(),
-                          params.channels(),
-                          params.sample_rate() / 100,
-                          0,
-                          0,
-                          true,
-                          false)).Times(AtLeast(1))
+              CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event));
   EXPECT_TRUE(track->GetAudioAdapter()->set_enabled(true));
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
 
   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
   capturer_->Stop();
   track.reset();
 }
 
 // Create multiple audio tracks and enable/disable them, verify that the audio
 // callbacks appear/disappear.
 // Flaky due to a data race, see http://crbug.com/295418
 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*capturer_source_.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
     new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
   static_cast<webrtc::AudioTrackInterface*>(
       adapter_1.get())->GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track_1->GetAudioAdapter()->enabled());
   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
   const media::AudioParameters params = capturer_->source_audio_parameters();
   base::WaitableEvent event_1(false, false);
-  EXPECT_CALL(*sink_1, OnSetFormat(_)).WillOnce(Return());
+  EXPECT_CALL(*sink_1, FormatIsSet()).WillOnce(Return());
   EXPECT_CALL(*sink_1,
-      CaptureData(1,
-                  params.sample_rate(),
-                  params.channels(),
-                  params.sample_rate() / 100,
-                  0,
-                  0,
-                  true,
-                  false)).Times(AtLeast(1))
+      CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
+  EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
+            params.sample_rate() / 100);
   track_1->AddSink(sink_1.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
     new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(
       adapter_2.get())->GetRenderer()->AddChannel(1);
   EXPECT_TRUE(track_2->GetAudioAdapter()->enabled());
 
   // Verify both |sink_1| and |sink_2| get data.
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
 
   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
-  EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(Return());
-  EXPECT_CALL(*sink_1,
-      CaptureData(1,
-                  params.sample_rate(),
-                  params.channels(),
-                  params.sample_rate() / 100,
-                  0,
-                  0,
-                  true,
-                  false)).Times(AtLeast(1))
+  EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(Return());
+  EXPECT_CALL(*sink_1, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
-  EXPECT_CALL(*sink_2,
-      CaptureData(1,
-                  params.sample_rate(),
-                  params.channels(),
-                  params.sample_rate() / 100,
-                  0,
-                  0,
-                  true,
-                  false)).Times(AtLeast(1))
+  EXPECT_EQ(sink_1->audio_params().frames_per_buffer(),
+            params.sample_rate() / 100);
+  EXPECT_CALL(*sink_2, CaptureData(1, 0, 0, _, false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_2));
+  EXPECT_EQ(sink_2->audio_params().frames_per_buffer(),
+            params.sample_rate() / 100);
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
   EXPECT_TRUE(event_2.TimedWait(TestTimeouts::tiny_timeout()));
 
   track_1->RemoveSink(sink_1.get());
   track_1->Stop();
   track_1.reset();
 
   EXPECT_CALL(*capturer_source_.get(), OnStop()).WillOnce(Return());
   track_2->RemoveSink(sink_2.get());
@@ skipping 38 matching lines @@
   static_cast<webrtc::AudioTrackInterface*>(
       adapter_1.get())->GetRenderer()->AddChannel(0);
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Verify the data flow by connecting the sink to |track_1|.
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   event.Reset();
-  EXPECT_CALL(*sink, OnSetFormat(_)).WillOnce(SignalEvent(&event));
-  EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, true, false))
+  EXPECT_CALL(*sink, FormatIsSet()).WillOnce(SignalEvent(&event));
+  EXPECT_CALL(*sink, CaptureData(_, 0, 0, _, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   track_1->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Start the second audio track will not start the |capturer_source_|
   // since it has been started.
   EXPECT_CALL(*capturer_source_.get(), OnStart()).Times(0);
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_2(
       new WebRtcLocalAudioTrack(adapter_2, capturer_, NULL));
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(
       adapter_2.get())->GetRenderer()->AddChannel(1);
 
   // Stop the capturer will clear up the track lists in the capturer.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
   capturer_->Stop();
 
   // Adding a new track to the capturer.
   track_2->AddSink(sink.get());
-  EXPECT_CALL(*sink, OnSetFormat(_)).Times(0);
+  EXPECT_CALL(*sink, FormatIsSet()).Times(0);
 
   // Stop the capturer again will not trigger stopping the source of the
   // capturer again..
   event.Reset();
   EXPECT_CALL(*capturer_source_.get(), OnStop()).Times(0);
   capturer_->Stop();
 }
 
 // Create a new capturer with new source, connect it to a new audio track.
 TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   // Setup the first audio track and start it.
   EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*capturer_source_.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_1(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track_1(
       new WebRtcLocalAudioTrack(adapter_1, capturer_, NULL));
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
 
   // Connect a number of network channels to the |track_1|.
   static const int kNumberOfNetworkChannelsForTrack1 = 2;
   for (int i = 0; i < kNumberOfNetworkChannelsForTrack1; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(
         adapter_1.get())->GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_1| to |track_1|.
   scoped_ptr<MockMediaStreamAudioSink> sink_1(new MockMediaStreamAudioSink());
-  EXPECT_CALL(
-      *sink_1.get(),
-      CaptureData(
-          kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, true, false))
+  EXPECT_CALL(*sink_1.get(),
+              CaptureData(kNumberOfNetworkChannelsForTrack1,
+                          0, 0, _, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
-  EXPECT_CALL(*sink_1.get(), OnSetFormat(_)).Times(AnyNumber());
+  EXPECT_CALL(*sink_1.get(), FormatIsSet()).Times(AnyNumber());
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   blink::WebMediaConstraints constraints;
   StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
                           std::string(), std::string());
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
       WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL));
   scoped_refptr<MockCapturerSource> new_source(
       new MockCapturerSource(new_capturer.get()));
@@ skipping 16 matching lines @@
 
   // Connect a number of network channels to the |track_2|.
   static const int kNumberOfNetworkChannelsForTrack2 = 3;
   for (int i = 0; i < kNumberOfNetworkChannelsForTrack2; ++i) {
     static_cast<webrtc::AudioTrackInterface*>(
         adapter_2.get())->GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_2| to |track_2|.
   scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(
-      *sink_2,
-      CaptureData(
-          kNumberOfNetworkChannelsForTrack2, new_param.sample_rate(),
-          new_param.channels(), _, 0, 0, true, false))
+  EXPECT_CALL(*sink_2,
+              CaptureData(kNumberOfNetworkChannelsForTrack2, 0, 0, _, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
-  EXPECT_CALL(*sink_2, OnSetFormat(_)).WillOnce(SignalEvent(&event));
+  EXPECT_CALL(*sink_2, FormatIsSet()).WillOnce(SignalEvent(&event));
   track_2->AddSink(sink_2.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stopping the new source will stop the second track.
   event.Reset();
   EXPECT_CALL(*new_source.get(), OnStop())
       .Times(1).WillOnce(SignalEvent(&event));
   new_capturer->Stop();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stop the capturer of the first audio track.
   EXPECT_CALL(*capturer_source_.get(), OnStop());
   capturer_->Stop();
 }
 
 
 // Make sure a audio track can deliver packets with a buffer size smaller than
 // 10ms when it is not connected with a peer connection.
 TEST_F(WebRtcLocalAudioTrackTest, TrackWorkWithSmallBufferSize) {
   // Setup a capturer which works with a buffer size smaller than 10ms.
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128);
 
   // Create a capturer with new source which works with the format above.
-  blink::WebMediaConstraints constraints;
+  MockMediaConstraintFactory factory;
+  factory.DisableDefaultAudioConstraints();
   scoped_refptr<WebRtcAudioCapturer> capturer(
       WebRtcAudioCapturer::CreateCapturer(
           -1,
           StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
                            "", "", params.sample_rate(),
                            params.channel_layout(),
                            params.frames_per_buffer()),
-          constraints,
+          factory.CreateWebMediaConstraints(),
           NULL));
   scoped_refptr<MockCapturerSource> source(
       new MockCapturerSource(capturer.get()));
   EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
   capturer->SetCapturerSourceForTesting(source, params);
 
   // Setup a audio track, connect it to the capturer and start it.
   EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
   EXPECT_CALL(*source.get(), OnStart());
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
       WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
   scoped_ptr<WebRtcLocalAudioTrack> track(
       new WebRtcLocalAudioTrack(adapter, capturer, NULL));
   static_cast<WebRtcLocalAudioSourceProvider*>(
       track->audio_source_provider())->SetSinkParamsForTesting(params);
   track->Start();
 
   // Verify the data flow by connecting the |sink| to |track|.
   scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
   base::WaitableEvent event(false, false);
-  EXPECT_CALL(*sink, OnSetFormat(_)).Times(1);
+  EXPECT_CALL(*sink, FormatIsSet()).Times(1);
   // Verify the sinks are getting the packets with an expecting buffer size.
 #if defined(OS_ANDROID)
   const int expected_buffer_size = params.sample_rate() / 100;
 #else
   const int expected_buffer_size = params.frames_per_buffer();
 #endif
   EXPECT_CALL(*sink, CaptureData(
-      0, params.sample_rate(), params.channels(), expected_buffer_size,
-      0, 0, true, false))
+      0, 0, 0, _, false))
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
+  EXPECT_EQ(expected_buffer_size, sink->audio_params().frames_per_buffer());
 
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source, OnStop()).Times(1);
   capturer->Stop();
 }
 
 }  // namespace content