Index: webrtc/api/rtpsenderreceiver_unittest.cc |
diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc |
deleted file mode 100644 |
index fc1b748034cd45ab5d1506b449870ec4216f9ec3..0000000000000000000000000000000000000000 |
--- a/webrtc/api/rtpsenderreceiver_unittest.cc |
+++ /dev/null |
@@ -1,724 +0,0 @@ |
-/* |
- * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
- * |
- * Use of this source code is governed by a BSD-style license |
- * that can be found in the LICENSE file in the root of the source |
- * tree. An additional intellectual property rights grant can be found |
- * in the file PATENTS. All contributing project authors may |
- * be found in the AUTHORS file in the root of the source tree. |
- */ |
- |
-#include <memory> |
-#include <string> |
-#include <utility> |
- |
-#include "webrtc/api/audiotrack.h" |
-#include "webrtc/api/fakemediacontroller.h" |
-#include "webrtc/api/localaudiosource.h" |
-#include "webrtc/api/mediastream.h" |
-#include "webrtc/api/remoteaudiosource.h" |
-#include "webrtc/api/rtpreceiver.h" |
-#include "webrtc/api/rtpsender.h" |
-#include "webrtc/api/streamcollection.h" |
-#include "webrtc/api/test/fakevideotracksource.h" |
-#include "webrtc/api/videotrack.h" |
-#include "webrtc/api/videotracksource.h" |
-#include "webrtc/base/gunit.h" |
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
-#include "webrtc/media/base/fakemediaengine.h" |
-#include "webrtc/media/base/mediachannel.h" |
-#include "webrtc/media/engine/fakewebrtccall.h" |
-#include "webrtc/p2p/base/faketransportcontroller.h" |
-#include "webrtc/pc/channelmanager.h" |
-#include "webrtc/test/gmock.h" |
-#include "webrtc/test/gtest.h" |
- |
-using ::testing::_; |
-using ::testing::Exactly; |
-using ::testing::InvokeWithoutArgs; |
-using ::testing::Return; |
- |
-static const char kStreamLabel1[] = "local_stream_1"; |
-static const char kVideoTrackId[] = "video_1"; |
-static const char kAudioTrackId[] = "audio_1"; |
-static const uint32_t kVideoSsrc = 98; |
-static const uint32_t kVideoSsrc2 = 100; |
-static const uint32_t kAudioSsrc = 99; |
-static const uint32_t kAudioSsrc2 = 101; |
- |
-namespace webrtc { |
- |
-class RtpSenderReceiverTest : public testing::Test { |
- public: |
- RtpSenderReceiverTest() |
- : // Create fake media engine/etc. so we can create channels to use to |
- // test RtpSenders/RtpReceivers. |
- media_engine_(new cricket::FakeMediaEngine()), |
- channel_manager_(media_engine_, |
- rtc::Thread::Current(), |
- rtc::Thread::Current()), |
- fake_call_(Call::Config(&event_log_)), |
- fake_media_controller_(&channel_manager_, &fake_call_), |
- stream_(MediaStream::Create(kStreamLabel1)) { |
- // Create channels to be used by the RtpSenders and RtpReceivers. |
- channel_manager_.Init(); |
- bool rtcp_mux_required = true; |
- bool srtp_required = true; |
- cricket::DtlsTransportInternal* rtp_transport = |
- fake_transport_controller_.CreateDtlsTransport( |
- cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP); |
- voice_channel_ = channel_manager_.CreateVoiceChannel( |
- &fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(), |
- cricket::CN_AUDIO, nullptr, rtcp_mux_required, srtp_required, |
- cricket::AudioOptions()); |
- video_channel_ = channel_manager_.CreateVideoChannel( |
- &fake_media_controller_, rtp_transport, nullptr, rtc::Thread::Current(), |
- cricket::CN_VIDEO, nullptr, rtcp_mux_required, srtp_required, |
- cricket::VideoOptions()); |
- voice_media_channel_ = media_engine_->GetVoiceChannel(0); |
- video_media_channel_ = media_engine_->GetVideoChannel(0); |
- RTC_CHECK(voice_channel_); |
- RTC_CHECK(video_channel_); |
- RTC_CHECK(voice_media_channel_); |
- RTC_CHECK(video_media_channel_); |
- |
- // Create streams for predefined SSRCs. Streams need to exist in order |
- // for the senders and receievers to apply parameters to them. |
- // Normally these would be created by SetLocalDescription and |
- // SetRemoteDescription. |
- voice_media_channel_->AddSendStream( |
- cricket::StreamParams::CreateLegacy(kAudioSsrc)); |
- voice_media_channel_->AddRecvStream( |
- cricket::StreamParams::CreateLegacy(kAudioSsrc)); |
- voice_media_channel_->AddSendStream( |
- cricket::StreamParams::CreateLegacy(kAudioSsrc2)); |
- voice_media_channel_->AddRecvStream( |
- cricket::StreamParams::CreateLegacy(kAudioSsrc2)); |
- video_media_channel_->AddSendStream( |
- cricket::StreamParams::CreateLegacy(kVideoSsrc)); |
- video_media_channel_->AddRecvStream( |
- cricket::StreamParams::CreateLegacy(kVideoSsrc)); |
- video_media_channel_->AddSendStream( |
- cricket::StreamParams::CreateLegacy(kVideoSsrc2)); |
- video_media_channel_->AddRecvStream( |
- cricket::StreamParams::CreateLegacy(kVideoSsrc2)); |
- } |
- |
- void TearDown() override { channel_manager_.Terminate(); } |
- |
- void AddVideoTrack() { AddVideoTrack(false); } |
- |
- void AddVideoTrack(bool is_screencast) { |
- rtc::scoped_refptr<VideoTrackSourceInterface> source( |
- FakeVideoTrackSource::Create(is_screencast)); |
- video_track_ = VideoTrack::Create(kVideoTrackId, source); |
- EXPECT_TRUE(stream_->AddTrack(video_track_)); |
- } |
- |
- void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); } |
- |
- void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) { |
- audio_track_ = AudioTrack::Create(kAudioTrackId, source); |
- EXPECT_TRUE(stream_->AddTrack(audio_track_)); |
- audio_rtp_sender_ = |
- new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(), |
- voice_channel_, nullptr); |
- audio_rtp_sender_->SetSsrc(kAudioSsrc); |
- VerifyVoiceChannelInput(); |
- } |
- |
- void CreateVideoRtpSender() { CreateVideoRtpSender(false); } |
- |
- void CreateVideoRtpSender(bool is_screencast) { |
- AddVideoTrack(is_screencast); |
- video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0], |
- stream_->label(), video_channel_); |
- video_rtp_sender_->SetSsrc(kVideoSsrc); |
- VerifyVideoChannelInput(); |
- } |
- |
- void DestroyAudioRtpSender() { |
- audio_rtp_sender_ = nullptr; |
- VerifyVoiceChannelNoInput(); |
- } |
- |
- void DestroyVideoRtpSender() { |
- video_rtp_sender_ = nullptr; |
- VerifyVideoChannelNoInput(); |
- } |
- |
- void CreateAudioRtpReceiver() { |
- audio_track_ = AudioTrack::Create( |
- kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL)); |
- EXPECT_TRUE(stream_->AddTrack(audio_track_)); |
- audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId, |
- kAudioSsrc, voice_channel_); |
- audio_track_ = audio_rtp_receiver_->audio_track(); |
- VerifyVoiceChannelOutput(); |
- } |
- |
- void CreateVideoRtpReceiver() { |
- video_rtp_receiver_ = |
- new VideoRtpReceiver(stream_, kVideoTrackId, rtc::Thread::Current(), |
- kVideoSsrc, video_channel_); |
- video_track_ = video_rtp_receiver_->video_track(); |
- VerifyVideoChannelOutput(); |
- } |
- |
- void DestroyAudioRtpReceiver() { |
- audio_rtp_receiver_ = nullptr; |
- VerifyVoiceChannelNoOutput(); |
- } |
- |
- void DestroyVideoRtpReceiver() { |
- video_rtp_receiver_ = nullptr; |
- VerifyVideoChannelNoOutput(); |
- } |
- |
- void VerifyVoiceChannelInput() { VerifyVoiceChannelInput(kAudioSsrc); } |
- |
- void VerifyVoiceChannelInput(uint32_t ssrc) { |
- // Verify that the media channel has an audio source, and the stream isn't |
- // muted. |
- EXPECT_TRUE(voice_media_channel_->HasSource(ssrc)); |
- EXPECT_FALSE(voice_media_channel_->IsStreamMuted(ssrc)); |
- } |
- |
- void VerifyVideoChannelInput() { VerifyVideoChannelInput(kVideoSsrc); } |
- |
- void VerifyVideoChannelInput(uint32_t ssrc) { |
- // Verify that the media channel has a video source, |
- EXPECT_TRUE(video_media_channel_->HasSource(ssrc)); |
- } |
- |
- void VerifyVoiceChannelNoInput() { VerifyVoiceChannelNoInput(kAudioSsrc); } |
- |
- void VerifyVoiceChannelNoInput(uint32_t ssrc) { |
- // Verify that the media channel's source is reset. |
- EXPECT_FALSE(voice_media_channel_->HasSource(ssrc)); |
- } |
- |
- void VerifyVideoChannelNoInput() { VerifyVideoChannelNoInput(kVideoSsrc); } |
- |
- void VerifyVideoChannelNoInput(uint32_t ssrc) { |
- // Verify that the media channel's source is reset. |
- EXPECT_FALSE(video_media_channel_->HasSource(ssrc)); |
- } |
- |
- void VerifyVoiceChannelOutput() { |
- // Verify that the volume is initialized to 1. |
- double volume; |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(1, volume); |
- } |
- |
- void VerifyVideoChannelOutput() { |
- // Verify that the media channel has a sink. |
- EXPECT_TRUE(video_media_channel_->HasSink(kVideoSsrc)); |
- } |
- |
- void VerifyVoiceChannelNoOutput() { |
- // Verify that the volume is reset to 0. |
- double volume; |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(0, volume); |
- } |
- |
- void VerifyVideoChannelNoOutput() { |
- // Verify that the media channel's sink is reset. |
- EXPECT_FALSE(video_media_channel_->HasSink(kVideoSsrc)); |
- } |
- |
- protected: |
- webrtc::RtcEventLogNullImpl event_log_; |
- cricket::FakeMediaEngine* media_engine_; |
- cricket::FakeTransportController fake_transport_controller_; |
- cricket::ChannelManager channel_manager_; |
- cricket::FakeCall fake_call_; |
- cricket::FakeMediaController fake_media_controller_; |
- cricket::VoiceChannel* voice_channel_; |
- cricket::VideoChannel* video_channel_; |
- cricket::FakeVoiceMediaChannel* voice_media_channel_; |
- cricket::FakeVideoMediaChannel* video_media_channel_; |
- rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_; |
- rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_; |
- rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_; |
- rtc::scoped_refptr<VideoRtpReceiver> video_rtp_receiver_; |
- rtc::scoped_refptr<MediaStreamInterface> stream_; |
- rtc::scoped_refptr<VideoTrackInterface> video_track_; |
- rtc::scoped_refptr<AudioTrackInterface> audio_track_; |
-}; |
- |
-// Test that |voice_channel_| is updated when an audio track is associated |
-// and disassociated with an AudioRtpSender. |
-TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpSender) { |
- CreateAudioRtpSender(); |
- DestroyAudioRtpSender(); |
-} |
- |
-// Test that |video_channel_| is updated when a video track is associated and |
-// disassociated with a VideoRtpSender. |
-TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpSender) { |
- CreateVideoRtpSender(); |
- DestroyVideoRtpSender(); |
-} |
- |
-// Test that |voice_channel_| is updated when a remote audio track is |
-// associated and disassociated with an AudioRtpReceiver. |
-TEST_F(RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver) { |
- CreateAudioRtpReceiver(); |
- DestroyAudioRtpReceiver(); |
-} |
- |
-// Test that |video_channel_| is updated when a remote video track is |
-// associated and disassociated with a VideoRtpReceiver. |
-TEST_F(RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver) { |
- CreateVideoRtpReceiver(); |
- DestroyVideoRtpReceiver(); |
-} |
- |
-// Test that the AudioRtpSender applies options from the local audio source. |
-TEST_F(RtpSenderReceiverTest, LocalAudioSourceOptionsApplied) { |
- cricket::AudioOptions options; |
- options.echo_cancellation = rtc::Optional<bool>(true); |
- auto source = LocalAudioSource::Create( |
- PeerConnectionFactoryInterface::Options(), &options); |
- CreateAudioRtpSender(source.get()); |
- |
- EXPECT_EQ(rtc::Optional<bool>(true), |
- voice_media_channel_->options().echo_cancellation); |
- |
- DestroyAudioRtpSender(); |
-} |
- |
-// Test that the stream is muted when the track is disabled, and unmuted when |
-// the track is enabled. |
-TEST_F(RtpSenderReceiverTest, LocalAudioTrackDisable) { |
- CreateAudioRtpSender(); |
- |
- audio_track_->set_enabled(false); |
- EXPECT_TRUE(voice_media_channel_->IsStreamMuted(kAudioSsrc)); |
- |
- audio_track_->set_enabled(true); |
- EXPECT_FALSE(voice_media_channel_->IsStreamMuted(kAudioSsrc)); |
- |
- DestroyAudioRtpSender(); |
-} |
- |
-// Test that the volume is set to 0 when the track is disabled, and back to |
-// 1 when the track is enabled. |
-TEST_F(RtpSenderReceiverTest, RemoteAudioTrackDisable) { |
- CreateAudioRtpReceiver(); |
- |
- double volume; |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(1, volume); |
- |
- audio_track_->set_enabled(false); |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(0, volume); |
- |
- audio_track_->set_enabled(true); |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(1, volume); |
- |
- DestroyAudioRtpReceiver(); |
-} |
- |
-// Currently no action is taken when a remote video track is disabled or |
-// enabled, so there's nothing to test here, other than what is normally |
-// verified in DestroyVideoRtpSender. |
-TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) { |
- CreateVideoRtpSender(); |
- |
- video_track_->set_enabled(false); |
- video_track_->set_enabled(true); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-// Test that the state of the video track created by the VideoRtpReceiver is |
-// updated when the receiver is destroyed. |
-TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) { |
- CreateVideoRtpReceiver(); |
- |
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state()); |
- EXPECT_EQ(webrtc::MediaSourceInterface::kLive, |
- video_track_->GetSource()->state()); |
- |
- DestroyVideoRtpReceiver(); |
- |
- EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state()); |
- EXPECT_EQ(webrtc::MediaSourceInterface::kEnded, |
- video_track_->GetSource()->state()); |
-} |
- |
-// Currently no action is taken when a remote video track is disabled or |
-// enabled, so there's nothing to test here, other than what is normally |
-// verified in DestroyVideoRtpReceiver. |
-TEST_F(RtpSenderReceiverTest, RemoteVideoTrackDisable) { |
- CreateVideoRtpReceiver(); |
- |
- video_track_->set_enabled(false); |
- video_track_->set_enabled(true); |
- |
- DestroyVideoRtpReceiver(); |
-} |
- |
-// Test that the AudioRtpReceiver applies volume changes from the track source |
-// to the media channel. |
-TEST_F(RtpSenderReceiverTest, RemoteAudioTrackSetVolume) { |
- CreateAudioRtpReceiver(); |
- |
- double volume; |
- audio_track_->GetSource()->SetVolume(0.5); |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(0.5, volume); |
- |
- // Disable the audio track, this should prevent setting the volume. |
- audio_track_->set_enabled(false); |
- audio_track_->GetSource()->SetVolume(0.8); |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(0, volume); |
- |
- // When the track is enabled, the previously set volume should take effect. |
- audio_track_->set_enabled(true); |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(0.8, volume); |
- |
- // Try changing volume one more time. |
- audio_track_->GetSource()->SetVolume(0.9); |
- EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume)); |
- EXPECT_EQ(0.9, volume); |
- |
- DestroyAudioRtpReceiver(); |
-} |
- |
-// Test that the media channel isn't enabled for sending if the audio sender |
-// doesn't have both a track and SSRC. |
-TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) { |
- audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); |
- rtc::scoped_refptr<AudioTrackInterface> track = |
- AudioTrack::Create(kAudioTrackId, nullptr); |
- |
- // Track but no SSRC. |
- EXPECT_TRUE(audio_rtp_sender_->SetTrack(track)); |
- VerifyVoiceChannelNoInput(); |
- |
- // SSRC but no track. |
- EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr)); |
- audio_rtp_sender_->SetSsrc(kAudioSsrc); |
- VerifyVoiceChannelNoInput(); |
-} |
- |
-// Test that the media channel isn't enabled for sending if the video sender |
-// doesn't have both a track and SSRC. |
-TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) { |
- video_rtp_sender_ = new VideoRtpSender(video_channel_); |
- |
- // Track but no SSRC. |
- EXPECT_TRUE(video_rtp_sender_->SetTrack(video_track_)); |
- VerifyVideoChannelNoInput(); |
- |
- // SSRC but no track. |
- EXPECT_TRUE(video_rtp_sender_->SetTrack(nullptr)); |
- video_rtp_sender_->SetSsrc(kVideoSsrc); |
- VerifyVideoChannelNoInput(); |
-} |
- |
-// Test that the media channel is enabled for sending when the audio sender |
-// has a track and SSRC, when the SSRC is set first. |
-TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) { |
- audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); |
- rtc::scoped_refptr<AudioTrackInterface> track = |
- AudioTrack::Create(kAudioTrackId, nullptr); |
- audio_rtp_sender_->SetSsrc(kAudioSsrc); |
- audio_rtp_sender_->SetTrack(track); |
- VerifyVoiceChannelInput(); |
- |
- DestroyAudioRtpSender(); |
-} |
- |
-// Test that the media channel is enabled for sending when the audio sender |
-// has a track and SSRC, when the SSRC is set last. |
-TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) { |
- audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr); |
- rtc::scoped_refptr<AudioTrackInterface> track = |
- AudioTrack::Create(kAudioTrackId, nullptr); |
- audio_rtp_sender_->SetTrack(track); |
- audio_rtp_sender_->SetSsrc(kAudioSsrc); |
- VerifyVoiceChannelInput(); |
- |
- DestroyAudioRtpSender(); |
-} |
- |
-// Test that the media channel is enabled for sending when the video sender |
-// has a track and SSRC, when the SSRC is set first. |
-TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) { |
- AddVideoTrack(); |
- video_rtp_sender_ = new VideoRtpSender(video_channel_); |
- video_rtp_sender_->SetSsrc(kVideoSsrc); |
- video_rtp_sender_->SetTrack(video_track_); |
- VerifyVideoChannelInput(); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-// Test that the media channel is enabled for sending when the video sender |
-// has a track and SSRC, when the SSRC is set last. |
-TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) { |
- AddVideoTrack(); |
- video_rtp_sender_ = new VideoRtpSender(video_channel_); |
- video_rtp_sender_->SetTrack(video_track_); |
- video_rtp_sender_->SetSsrc(kVideoSsrc); |
- VerifyVideoChannelInput(); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-// Test that the media channel stops sending when the audio sender's SSRC is set |
-// to 0. |
-TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) { |
- CreateAudioRtpSender(); |
- |
- audio_rtp_sender_->SetSsrc(0); |
- VerifyVoiceChannelNoInput(); |
-} |
- |
-// Test that the media channel stops sending when the video sender's SSRC is set |
-// to 0. |
-TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) { |
- CreateAudioRtpSender(); |
- |
- audio_rtp_sender_->SetSsrc(0); |
- VerifyVideoChannelNoInput(); |
-} |
- |
-// Test that the media channel stops sending when the audio sender's track is |
-// set to null. |
-TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) { |
- CreateAudioRtpSender(); |
- |
- EXPECT_TRUE(audio_rtp_sender_->SetTrack(nullptr)); |
- VerifyVoiceChannelNoInput(); |
-} |
- |
-// Test that the media channel stops sending when the video sender's track is |
-// set to null. |
-TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) { |
- CreateVideoRtpSender(); |
- |
- video_rtp_sender_->SetSsrc(0); |
- VerifyVideoChannelNoInput(); |
-} |
- |
-// Test that when the audio sender's SSRC is changed, the media channel stops |
-// sending with the old SSRC and starts sending with the new one. |
-TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) { |
- CreateAudioRtpSender(); |
- |
- audio_rtp_sender_->SetSsrc(kAudioSsrc2); |
- VerifyVoiceChannelNoInput(kAudioSsrc); |
- VerifyVoiceChannelInput(kAudioSsrc2); |
- |
- audio_rtp_sender_ = nullptr; |
- VerifyVoiceChannelNoInput(kAudioSsrc2); |
-} |
- |
-// Test that when the audio sender's SSRC is changed, the media channel stops |
-// sending with the old SSRC and starts sending with the new one. |
-TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) { |
- CreateVideoRtpSender(); |
- |
- video_rtp_sender_->SetSsrc(kVideoSsrc2); |
- VerifyVideoChannelNoInput(kVideoSsrc); |
- VerifyVideoChannelInput(kVideoSsrc2); |
- |
- video_rtp_sender_ = nullptr; |
- VerifyVideoChannelNoInput(kVideoSsrc2); |
-} |
- |
-TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) { |
- CreateAudioRtpSender(); |
- |
- RtpParameters params = audio_rtp_sender_->GetParameters(); |
- EXPECT_EQ(1u, params.encodings.size()); |
- EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); |
- |
- DestroyAudioRtpSender(); |
-} |
- |
-TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { |
- CreateAudioRtpSender(); |
- |
- EXPECT_EQ(-1, voice_media_channel_->max_bps()); |
- webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); |
- EXPECT_EQ(1, params.encodings.size()); |
- EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); |
- params.encodings[0].max_bitrate_bps = 1000; |
- EXPECT_TRUE(audio_rtp_sender_->SetParameters(params)); |
- |
- // Read back the parameters and verify they have been changed. |
- params = audio_rtp_sender_->GetParameters(); |
- 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 = voice_media_channel_->GetRtpSendParameters(kAudioSsrc); |
- 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, voice_media_channel_->max_bps()); |
- |
- DestroyAudioRtpSender(); |
-} |
- |
-TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) { |
- CreateVideoRtpSender(); |
- |
- RtpParameters params = video_rtp_sender_->GetParameters(); |
- EXPECT_EQ(1u, params.encodings.size()); |
- EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) { |
- CreateVideoRtpSender(); |
- |
- EXPECT_EQ(-1, video_media_channel_->max_bps()); |
- webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); |
- EXPECT_EQ(1, params.encodings.size()); |
- EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps); |
- params.encodings[0].max_bitrate_bps = 1000; |
- EXPECT_TRUE(video_rtp_sender_->SetParameters(params)); |
- |
- // Read back the parameters and verify they have been changed. |
- params = video_rtp_sender_->GetParameters(); |
- 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 = video_media_channel_->GetRtpSendParameters(kVideoSsrc); |
- 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, video_media_channel_->max_bps()); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) { |
- CreateAudioRtpReceiver(); |
- |
- RtpParameters params = audio_rtp_receiver_->GetParameters(); |
- EXPECT_EQ(1u, params.encodings.size()); |
- EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params)); |
- |
- DestroyAudioRtpReceiver(); |
-} |
- |
-TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) { |
- CreateVideoRtpReceiver(); |
- |
- RtpParameters params = video_rtp_receiver_->GetParameters(); |
- EXPECT_EQ(1u, params.encodings.size()); |
- EXPECT_TRUE(video_rtp_receiver_->SetParameters(params)); |
- |
- DestroyVideoRtpReceiver(); |
-} |
- |
-// Test that makes sure that a video track content hint translates to the proper |
-// value for sources that are not screencast. |
-TEST_F(RtpSenderReceiverTest, PropagatesVideoTrackContentHint) { |
- CreateVideoRtpSender(); |
- |
- video_track_->set_enabled(true); |
- |
- // |video_track_| is not screencast by default. |
- EXPECT_EQ(rtc::Optional<bool>(false), |
- video_media_channel_->options().is_screencast); |
- // No content hint should be set by default. |
- EXPECT_EQ(VideoTrackInterface::ContentHint::kNone, |
- video_track_->content_hint()); |
- // Setting detailed should turn a non-screencast source into screencast mode. |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed); |
- EXPECT_EQ(rtc::Optional<bool>(true), |
- video_media_channel_->options().is_screencast); |
- // Removing the content hint should turn the track back into non-screencast |
- // mode. |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone); |
- EXPECT_EQ(rtc::Optional<bool>(false), |
- video_media_channel_->options().is_screencast); |
- // Setting fluid should remain in non-screencast mode (its default). |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid); |
- EXPECT_EQ(rtc::Optional<bool>(false), |
- video_media_channel_->options().is_screencast); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-// Test that makes sure that a video track content hint translates to the proper |
-// value for screencast sources. |
-TEST_F(RtpSenderReceiverTest, |
- PropagatesVideoTrackContentHintForScreencastSource) { |
- CreateVideoRtpSender(true); |
- |
- video_track_->set_enabled(true); |
- |
- // |video_track_| with a screencast source should be screencast by default. |
- EXPECT_EQ(rtc::Optional<bool>(true), |
- video_media_channel_->options().is_screencast); |
- // No content hint should be set by default. |
- EXPECT_EQ(VideoTrackInterface::ContentHint::kNone, |
- video_track_->content_hint()); |
- // Setting fluid should turn a screencast source into non-screencast mode. |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid); |
- EXPECT_EQ(rtc::Optional<bool>(false), |
- video_media_channel_->options().is_screencast); |
- // Removing the content hint should turn the track back into screencast mode. |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone); |
- EXPECT_EQ(rtc::Optional<bool>(true), |
- video_media_channel_->options().is_screencast); |
- // Setting detailed should still remain in screencast mode (its default). |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed); |
- EXPECT_EQ(rtc::Optional<bool>(true), |
- video_media_channel_->options().is_screencast); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-// Test that makes sure any content hints that are set on a track before |
-// VideoRtpSender is ready to send are still applied when it gets ready to send. |
-TEST_F(RtpSenderReceiverTest, |
- PropagatesVideoTrackContentHintSetBeforeEnabling) { |
- AddVideoTrack(); |
- // Setting detailed overrides the default non-screencast mode. This should be |
- // applied even if the track is set on construction. |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed); |
- video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0], |
- stream_->label(), video_channel_); |
- video_track_->set_enabled(true); |
- |
- // Sender is not ready to send (no SSRC) so no option should have been set. |
- EXPECT_EQ(rtc::Optional<bool>(), |
- video_media_channel_->options().is_screencast); |
- |
- // Verify that the content hint is accounted for when video_rtp_sender_ does |
- // get enabled. |
- video_rtp_sender_->SetSsrc(kVideoSsrc); |
- EXPECT_EQ(rtc::Optional<bool>(true), |
- video_media_channel_->options().is_screencast); |
- |
- // And removing the hint should go back to false (to verify that false was |
- // default correctly). |
- video_track_->set_content_hint(VideoTrackInterface::ContentHint::kNone); |
- EXPECT_EQ(rtc::Optional<bool>(false), |
- video_media_channel_->options().is_screencast); |
- |
- DestroyVideoRtpSender(); |
-} |
- |
-} // namespace webrtc |