Switch LiveAudio to source provider solution.

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/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_capturer.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"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
@@ skipping 107 matching lines @@
                     bool need_audio_processing,
                     bool key_pressed));
   MOCK_METHOD1(SetCaptureFormat, void(const 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);
     capturer_ = WebRtcAudioCapturer::CreateCapturer();
+    WebRtcLocalAudioSourceProvider* source_provider =
+        static_cast<WebRtcLocalAudioSourceProvider*>(
+            capturer_->audio_source_provider());
+    source_provider->SetSinkParamsForTesting(params_);
     capturer_source_ = new MockCapturerSource();
     EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
         .WillOnce(Return());
     capturer_->SetCapturerSource(capturer_source_,
-                                 media::CHANNEL_LAYOUT_STEREO,
-                                 48000);
+                                 params_.channel_layout(),
+                                 params_.sample_rate());
 
     EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(false))
         .WillOnce(Return());
 
     // Start the audio thread used by the |capturer_source_|.
     audio_thread_.reset(new FakeAudioThread(capturer_));
     audio_thread_->Start();
   }
 
   virtual void TearDown() {
     audio_thread_->Stop();
     audio_thread_.reset();
   }
 
+  media::AudioParameters params_;
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
   scoped_ptr<FakeAudioThread> audio_thread_;
 };
 
 // Creates a capturer and audio track, fakes its audio thread, and
 // connect/disconnect the sink to the audio track on the fly, the sink should
 // get data callback when the track is connected to the capturer but not when
 // the track is disconnected from the capturer.
 TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track->Start();
   EXPECT_TRUE(track->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*>(track.get())->
         GetRenderer()->AddChannel(i);
   }
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink,
       CaptureData(kNumberOfNetworkChannels,
                   params.sample_rate(),
                   params.channels(),
-                  params.frames_per_buffer(),
+                  params.sample_rate() / 100,
                   0,
                   0,
                   false,
                   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(), Stop()).WillOnce(Return());
   track->Stop();
   track = NULL;
 }
 
 // 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
 // callback.
 // TODO(xians): Enable this test after resolving the racing issue that TSAN
 // reports on MediaStreamTrack::enabled();
 TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   track->Start();
   static_cast<webrtc::AudioTrackInterface*>(track.get())->
       GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track->enabled());
   EXPECT_TRUE(track->set_enabled(false));
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*sink, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink,
               CaptureData(1,
                           params.sample_rate(),
                           params.channels(),
-                          params.frames_per_buffer(),
+                          params.sample_rate() / 100,
                           0,
                           0,
                           false,
                           false)).Times(0);
   track->AddSink(sink.get());
   EXPECT_FALSE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   event.Reset();
   EXPECT_CALL(*sink,
               CaptureData(1,
                           params.sample_rate(),
                           params.channels(),
-                          params.frames_per_buffer(),
+                          params.sample_rate() / 100,
                           0,
                           0,
                           false,
                           false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event));
   EXPECT_TRUE(track->set_enabled(true));
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
   track->RemoveSink(sink.get());
 
   EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
   track->Stop();
   track = NULL;
 }
 
 // Create multiple audio tracks and enable/disable them, verify that the audio
 // callbacks appear/disappear.
 TEST_F(WebRtcLocalAudioTrackTest, MultipleAudioTracks) {
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(Return());
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
-    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   track_1->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   EXPECT_TRUE(track_1->enabled());
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
       new MockWebRtcAudioCapturerSink());
   const media::AudioParameters params = capturer_->audio_parameters();
   base::WaitableEvent event_1(false, false);
   EXPECT_CALL(*sink_1, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
-                  params.frames_per_buffer(),
+                  params.sample_rate() / 100,
                   0,
                   0,
                   false,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
   track_1->AddSink(sink_1.get());
   EXPECT_TRUE(event_1.TimedWait(TestTimeouts::tiny_timeout()));
 
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
-    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+    WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
   EXPECT_TRUE(track_2->enabled());
 
   // Verify both |sink_1| and |sink_2| get data.
   event_1.Reset();
   base::WaitableEvent event_2(false, false);
 
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_2(
         new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(*sink_2, SetCaptureFormat(_)).WillOnce(Return());
   EXPECT_CALL(*sink_1,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
-                  params.frames_per_buffer(),
+                  params.sample_rate() / 100,
                   0,
                   0,
                   false,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_1));
   EXPECT_CALL(*sink_2,
       CaptureData(1,
                   params.sample_rate(),
                   params.channels(),
-                  params.frames_per_buffer(),
+                  params.sample_rate() / 100,
                   0,
                   0,
                   false,
                   false)).Times(AtLeast(1))
       .WillRepeatedly(SignalEvent(&event_2));
   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 = NULL;
 
   EXPECT_CALL(*capturer_source_.get(), Stop()).WillOnce(Return());
   track_2->RemoveSink(sink_2.get());
   track_2->Stop();
   track_2 = NULL;
 }
 
 
 // Start one track and verify the capturer is correctly starting its source.
 // And it should be fine to not to call Stop() explicitly.
 TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track->Start();
 
   // When the track goes away, it will automatically stop the
   // |capturer_source_|.
   EXPECT_CALL(*capturer_source_.get(), Stop());
   track->Stop();
   track = NULL;
 }
 
 // Start/Stop tracks and verify the capturer is correctly starting/stopping
 // its source.
 TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   // Starting the first audio track will start the |capturer_source_|.
   base::WaitableEvent event(false, false);
   EXPECT_CALL(*capturer_source_.get(), Start()).WillOnce(SignalEvent(&event));
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Verify the data flow by connecting the sink to |track_1|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink(
       new MockWebRtcAudioCapturerSink());
   event.Reset();
   EXPECT_CALL(*sink, CaptureData(_, _, _, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink, SetCaptureFormat(_)).Times(1);
   track_1->AddSink(sink.get());
 
   // Start the second audio track will not start the |capturer_source_|
   // since it has been started.
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(0);
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
 
   // Stop the first audio track will not stop the |capturer_source_|.
   EXPECT_CALL(*capturer_source_.get(), Stop()).Times(0);
   track_1->RemoveSink(sink.get());
   track_1->Stop();
   track_1 = NULL;
@@ skipping 12 matching lines @@
   track_2 = NULL;
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 }
 
 // Set new source to the existing capturer.
 TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
   // Setup the audio track and start the track.
   EXPECT_CALL(*capturer_source_.get(), Start()).Times(1);
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   track->Start();
 
   // Setting new source to the capturer and the track should still get packets.
   scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
   EXPECT_CALL(*capturer_source_.get(), Stop());
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false));
   EXPECT_CALL(*new_source.get(), Initialize(_, capturer_.get(), 0))
       .WillOnce(Return());
   EXPECT_CALL(*new_source.get(), Start()).WillOnce(Return());
   capturer_->SetCapturerSource(new_source,
-                               media::CHANNEL_LAYOUT_STEREO,
-                               48000);
+                               params_.channel_layout(),
+                               params_.sample_rate());
 
   // Stop the track.
   EXPECT_CALL(*new_source.get(), Stop());
   track->Stop();
   track = NULL;
 }
 
 // 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(), Start()).Times(1);
   RTCMediaConstraints constraints;
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
-      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
   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*>(track_1.get())->
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_1| to |track_1|.
   scoped_ptr<MockWebRtcAudioCapturerSink> sink_1(
       new MockWebRtcAudioCapturerSink());
   EXPECT_CALL(
       *sink_1.get(),
       CaptureData(
           kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, false, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_1.get(), SetCaptureFormat(_)).Times(1);
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
       WebRtcAudioCapturer::CreateCapturer());
+  WebRtcLocalAudioSourceProvider* source_provider =
+      static_cast<WebRtcLocalAudioSourceProvider*>(
+          new_capturer->audio_source_provider());
+  source_provider->SetSinkParamsForTesting(params_);
   scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
   EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
       .WillOnce(Return());
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(false))
       .WillOnce(Return());
   new_capturer->SetCapturerSource(new_source,
                                   media::CHANNEL_LAYOUT_MONO,
                                   44100);
 
   // Start the audio thread used by the new source.
   scoped_ptr<FakeAudioThread> audio_thread(new FakeAudioThread(new_capturer));
   audio_thread->Start();
 
   // Setup the second audio track, connect it to the new capturer and start it.
   EXPECT_CALL(*new_source.get(), Start()).Times(1);
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
-      WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL,
+      WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
           &constraints);
   track_2->Start();
 
   // 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*>(track_2.get())->
         GetRenderer()->AddChannel(i);
   }
   // Verify the data flow by connecting the |sink_2| to |track_2|.
@@ skipping 17 matching lines @@
   audio_thread->Stop();
   audio_thread.reset();
 
   // Stop the first audio track.
   EXPECT_CALL(*capturer_source_.get(), Stop());
   track_1->Stop();
   track_1 = NULL;
 }
 
 }  // namespace content