Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/webrtcsession_unittest.cc

Index: webrtc/api/webrtcsession_unittest.cc
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index ee1d19b23adaee4cf899e28666b294dafc0aa127..fab5e1a56fcc4cd1c314fbfe84f8f5aeca06f1eb 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -252,6 +252,11 @@
   cricket::TransportChannel* data_rtcp_transport_channel() {
     return rtcp_transport_channel(data_channel());
   }
+
+  using webrtc::WebRtcSession::SetAudioPlayout;
+  using webrtc::WebRtcSession::SetAudioSend;
+  using webrtc::WebRtcSession::SetVideoPlayout;
+  using webrtc::WebRtcSession::SetVideoSend;
 
  private:
   cricket::TransportChannel* rtp_transport_channel(cricket::BaseChannel* ch) {
@@ -3387,6 +3392,163 @@
   SetLocalDescriptionWithoutError(offer);
 }
 
+TEST_F(WebRtcSessionTest, SetAudioPlayout) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->recv_streams().size());
+  uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc();
+  double volume;
+  EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
+  EXPECT_EQ(1, volume);
+  session_->SetAudioPlayout(receive_ssrc, false);
+  EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
+  EXPECT_EQ(0, volume);
+  session_->SetAudioPlayout(receive_ssrc, true);
+  EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
+  EXPECT_EQ(1, volume);
+}
+
+TEST_F(WebRtcSessionTest, SetAudioMaxSendBitrate) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_EQ(-1, channel->max_bps());
+  webrtc::RtpParameters params = session_->GetAudioRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetAudioRtpSendParameters(send_ssrc, params));
+
+  // Read back the parameters and verify they have been changed.
+  params = session_->GetAudioRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the audio channel received the new parameters.
+  params = channel->GetRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, channel->max_bps());
+}
+
+TEST_F(WebRtcSessionTest, SetAudioSend) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->send_streams().size());
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+
+  cricket::AudioOptions options;
+  options.echo_cancellation = rtc::Optional<bool>(true);
+
+  std::unique_ptr<FakeAudioSource> source(new FakeAudioSource());
+  session_->SetAudioSend(send_ssrc, false, options, source.get());
+  EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
+  EXPECT_EQ(rtc::Optional<bool>(), channel->options().echo_cancellation);
+  EXPECT_TRUE(source->sink() != nullptr);
+
+  // This will trigger SetSink(nullptr) to the |source|.
+  session_->SetAudioSend(send_ssrc, true, options, nullptr);
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+  EXPECT_EQ(rtc::Optional<bool>(true), channel->options().echo_cancellation);
+  EXPECT_TRUE(source->sink() == nullptr);
+}
+
+TEST_F(WebRtcSessionTest, AudioSourceForLocalStream) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->send_streams().size());
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+
+  std::unique_ptr<FakeAudioSource> source(new FakeAudioSource());
+  cricket::AudioOptions options;
+  session_->SetAudioSend(send_ssrc, true, options, source.get());
+  EXPECT_TRUE(source->sink() != nullptr);
+
+  // Delete the |source| and it will trigger OnClose() to the sink, and this
+  // will invalidate the |source_| pointer in the sink and prevent getting a
+  // SetSink(nullptr) callback afterwards.
+  source.reset();
+
+  // This will trigger SetSink(nullptr) if no OnClose() callback.
+  session_->SetAudioSend(send_ssrc, true, options, nullptr);
+}
+
+TEST_F(WebRtcSessionTest, SetVideoPlayout) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_LT(0u, channel->sinks().size());
+  EXPECT_TRUE(channel->sinks().begin()->second == NULL);
+  ASSERT_EQ(1u, channel->recv_streams().size());
+  uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc();
+  cricket::FakeVideoRenderer renderer;
+  session_->SetVideoPlayout(receive_ssrc, true, &renderer);
+  EXPECT_TRUE(channel->sinks().begin()->second == &renderer);
+  session_->SetVideoPlayout(receive_ssrc, false, &renderer);
+  EXPECT_TRUE(channel->sinks().begin()->second == NULL);
+}
+
+TEST_F(WebRtcSessionTest, SetVideoMaxSendBitrate) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_EQ(-1, channel->max_bps());
+  webrtc::RtpParameters params = session_->GetVideoRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetVideoRtpSendParameters(send_ssrc, params));
+
+  // Read back the parameters and verify they have been changed.
+  params = session_->GetVideoRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the video channel received the new parameters.
+  params = channel->GetRtpSendParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, channel->max_bps());
+}
+
+TEST_F(WebRtcSessionTest, SetVideoSend) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  ASSERT_EQ(1u, channel->send_streams().size());
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+  cricket::VideoOptions* options = NULL;
+  session_->SetVideoSend(send_ssrc, false, options, nullptr);
+  EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
+  session_->SetVideoSend(send_ssrc, true, options, nullptr);
+  EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
+}
+
 TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
   TestCanInsertDtmf(false);
 }