| Index: webrtc/api/webrtcsession_unittest.cc
|
| diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
|
| index 1907b8410ccac03b0504bbe5e2187823dfc7bfa6..a09e9302c4dad3396fa41fb343af486568ca2349 100644
|
| --- a/webrtc/api/webrtcsession_unittest.cc
|
| +++ b/webrtc/api/webrtcsession_unittest.cc
|
| @@ -253,11 +253,6 @@ class WebRtcSessionForTest : public webrtc::WebRtcSession {
|
| 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) {
|
| if (!ch) {
|
| @@ -3390,163 +3385,6 @@ TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
|
| 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);
|
| }
|
|
|