| Index: webrtc/api/rtpsenderreceiver_unittest.cc
|
| diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc
|
| index 8c91e148bef38daff9050d1d266baf803ca22469..a08e1b4c634a34083e01b451fd8080560504d30b 100644
|
| --- a/webrtc/api/rtpsenderreceiver_unittest.cc
|
| +++ b/webrtc/api/rtpsenderreceiver_unittest.cc
|
| @@ -15,8 +15,6 @@
|
| #include "testing/gmock/include/gmock/gmock.h"
|
| #include "testing/gtest/include/gtest/gtest.h"
|
| #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"
|
| @@ -27,10 +25,6 @@
|
| #include "webrtc/api/videotrack.h"
|
| #include "webrtc/base/gunit.h"
|
| #include "webrtc/media/base/mediachannel.h"
|
| -#include "webrtc/media/base/fakemediaengine.h"
|
| -#include "webrtc/media/engine/fakewebrtccall.h"
|
| -#include "webrtc/p2p/base/faketransportcontroller.h"
|
| -#include "webrtc/pc/channelmanager.h"
|
|
|
| using ::testing::_;
|
| using ::testing::Exactly;
|
| @@ -47,56 +41,69 @@
|
|
|
| namespace webrtc {
|
|
|
| +// Helper class to test RtpSender/RtpReceiver.
|
| +class MockAudioProvider : public AudioProviderInterface {
|
| + public:
|
| + // TODO(nisse): Valid overrides commented out, because the gmock
|
| + // methods don't use any override declarations, and we want to avoid
|
| + // warnings from -Winconsistent-missing-override. See
|
| + // http://crbug.com/428099.
|
| + ~MockAudioProvider() /* override */ {}
|
| +
|
| + MOCK_METHOD2(SetAudioPlayout,
|
| + void(uint32_t ssrc,
|
| + bool enable));
|
| + MOCK_METHOD4(SetAudioSend,
|
| + void(uint32_t ssrc,
|
| + bool enable,
|
| + const cricket::AudioOptions& options,
|
| + cricket::AudioSource* source));
|
| + MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
|
| + MOCK_CONST_METHOD1(GetAudioRtpSendParameters, RtpParameters(uint32_t ssrc));
|
| + MOCK_METHOD2(SetAudioRtpSendParameters,
|
| + bool(uint32_t ssrc, const RtpParameters&));
|
| + MOCK_CONST_METHOD1(GetAudioRtpReceiveParameters,
|
| + RtpParameters(uint32_t ssrc));
|
| + MOCK_METHOD2(SetAudioRtpReceiveParameters,
|
| + bool(uint32_t ssrc, const RtpParameters&));
|
| +
|
| + void SetRawAudioSink(
|
| + uint32_t, std::unique_ptr<AudioSinkInterface> sink) /* override */ {
|
| + sink_ = std::move(sink);
|
| + }
|
| +
|
| + private:
|
| + std::unique_ptr<AudioSinkInterface> sink_;
|
| +};
|
| +
|
| +// Helper class to test RtpSender/RtpReceiver.
|
| +class MockVideoProvider : public VideoProviderInterface {
|
| + public:
|
| + virtual ~MockVideoProvider() {}
|
| + MOCK_METHOD3(SetVideoPlayout,
|
| + void(uint32_t ssrc,
|
| + bool enable,
|
| + rtc::VideoSinkInterface<cricket::VideoFrame>* sink));
|
| + MOCK_METHOD4(SetVideoSend,
|
| + void(uint32_t ssrc,
|
| + bool enable,
|
| + const cricket::VideoOptions* options,
|
| + rtc::VideoSourceInterface<cricket::VideoFrame>* source));
|
| +
|
| + MOCK_CONST_METHOD1(GetVideoRtpSendParameters, RtpParameters(uint32_t ssrc));
|
| + MOCK_METHOD2(SetVideoRtpSendParameters,
|
| + bool(uint32_t ssrc, const RtpParameters&));
|
| + MOCK_CONST_METHOD1(GetVideoRtpReceiveParameters,
|
| + RtpParameters(uint32_t ssrc));
|
| + MOCK_METHOD2(SetVideoRtpReceiveParameters,
|
| + bool(uint32_t ssrc, const RtpParameters&));
|
| +};
|
| +
|
| 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_(webrtc::Call::Config()),
|
| - fake_media_controller_(&channel_manager_, &fake_call_),
|
| - stream_(MediaStream::Create(kStreamLabel1)) {
|
| - // Create channels to be used by the RtpSenders and RtpReceivers.
|
| - channel_manager_.Init();
|
| - voice_channel_ = channel_manager_.CreateVoiceChannel(
|
| - &fake_media_controller_, &fake_transport_controller_, cricket::CN_AUDIO,
|
| - nullptr, false, cricket::AudioOptions());
|
| - video_channel_ = channel_manager_.CreateVideoChannel(
|
| - &fake_media_controller_, &fake_transport_controller_, cricket::CN_VIDEO,
|
| - nullptr, false, 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(); }
|
| + virtual void SetUp() {
|
| + stream_ = MediaStream::Create(kStreamLabel1);
|
| + }
|
|
|
| void AddVideoTrack() {
|
| rtc::scoped_refptr<VideoTrackSourceInterface> source(
|
| @@ -105,128 +112,68 @@
|
| EXPECT_TRUE(stream_->AddTrack(video_track_));
|
| }
|
|
|
| - void CreateAudioRtpSender() { CreateAudioRtpSender(nullptr); }
|
| -
|
| - void CreateAudioRtpSender(rtc::scoped_refptr<LocalAudioSource> source) {
|
| - audio_track_ = AudioTrack::Create(kAudioTrackId, source);
|
| + void CreateAudioRtpSender() {
|
| + audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
|
| EXPECT_TRUE(stream_->AddTrack(audio_track_));
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
|
| audio_rtp_sender_ =
|
| new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
|
| - voice_channel_, nullptr);
|
| + &audio_provider_, nullptr);
|
| audio_rtp_sender_->SetSsrc(kAudioSsrc);
|
| - VerifyVoiceChannelInput();
|
| }
|
|
|
| void CreateVideoRtpSender() {
|
| AddVideoTrack();
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
|
| video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
|
| - stream_->label(), video_channel_);
|
| + stream_->label(), &video_provider_);
|
| video_rtp_sender_->SetSsrc(kVideoSsrc);
|
| - VerifyVideoChannelInput();
|
| }
|
|
|
| void DestroyAudioRtpSender() {
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _))
|
| + .Times(1);
|
| audio_rtp_sender_ = nullptr;
|
| - VerifyVoiceChannelNoInput();
|
| }
|
|
|
| void DestroyVideoRtpSender() {
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
|
| + .Times(1);
|
| video_rtp_sender_ = nullptr;
|
| - VerifyVideoChannelNoInput();
|
| }
|
|
|
| void CreateAudioRtpReceiver() {
|
| audio_track_ = AudioTrack::Create(
|
| kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
|
| EXPECT_TRUE(stream_->AddTrack(audio_track_));
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
|
| audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId,
|
| - kAudioSsrc, voice_channel_);
|
| + kAudioSsrc, &audio_provider_);
|
| audio_track_ = audio_rtp_receiver_->audio_track();
|
| - VerifyVoiceChannelOutput();
|
| }
|
|
|
| void CreateVideoRtpReceiver() {
|
| + EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, true, _));
|
| video_rtp_receiver_ =
|
| new VideoRtpReceiver(stream_, kVideoTrackId, rtc::Thread::Current(),
|
| - kVideoSsrc, video_channel_);
|
| + kVideoSsrc, &video_provider_);
|
| video_track_ = video_rtp_receiver_->video_track();
|
| - VerifyVideoChannelOutput();
|
| }
|
|
|
| void DestroyAudioRtpReceiver() {
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false));
|
| audio_rtp_receiver_ = nullptr;
|
| - VerifyVoiceChannelNoOutput();
|
| }
|
|
|
| void DestroyVideoRtpReceiver() {
|
| + EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, false, NULL));
|
| 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:
|
| - 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_;
|
| + MockAudioProvider audio_provider_;
|
| + MockVideoProvider video_provider_;
|
| rtc::scoped_refptr<AudioRtpSender> audio_rtp_sender_;
|
| rtc::scoped_refptr<VideoRtpSender> video_rtp_sender_;
|
| rtc::scoped_refptr<AudioRtpReceiver> audio_rtp_receiver_;
|
| @@ -236,96 +183,72 @@
|
| rtc::scoped_refptr<AudioTrackInterface> audio_track_;
|
| };
|
|
|
| -// Test that |voice_channel_| is updated when an audio track is associated
|
| +// Test that |audio_provider_| is notified 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
|
| +// Test that |video_provider_| is notified 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
|
| +// Test that |audio_provider_| is notified when a remote audio and 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 that |video_provider_| is notified 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();
|
|
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _));
|
| audio_track_->set_enabled(false);
|
| - EXPECT_TRUE(voice_media_channel_->IsStreamMuted(kAudioSsrc));
|
| -
|
| +
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
|
| 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);
|
| -
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false));
|
| audio_track_->set_enabled(false);
|
| - EXPECT_TRUE(voice_media_channel_->GetOutputVolume(kAudioSsrc, &volume));
|
| - EXPECT_EQ(0, volume);
|
| -
|
| +
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
|
| 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();
|
|
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc, false, _, video_track_.get()));
|
| video_track_->set_enabled(false);
|
| +
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
|
| 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();
|
|
|
| @@ -340,223 +263,257 @@
|
| 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);
|
| + double volume = 0.5;
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, volume));
|
| + audio_track_->GetSource()->SetVolume(volume);
|
|
|
| // Disable the audio track, this should prevent setting the volume.
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, false));
|
| 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_->GetSource()->SetVolume(1.0);
|
| +
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
|
| 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);
|
| +
|
| + double new_volume = 0.8;
|
| + EXPECT_CALL(audio_provider_, SetAudioPlayoutVolume(kAudioSsrc, new_volume));
|
| + audio_track_->GetSource()->SetVolume(new_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 that provider methods aren't called without both a track and an SSRC.
|
| TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
|
| - audio_rtp_sender_ = new AudioRtpSender(voice_channel_, nullptr);
|
| + rtc::scoped_refptr<AudioRtpSender> sender =
|
| + new AudioRtpSender(&audio_provider_, 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.
|
| + EXPECT_TRUE(sender->SetTrack(track));
|
| + EXPECT_TRUE(sender->SetTrack(nullptr));
|
| + sender->SetSsrc(kAudioSsrc);
|
| + sender->SetSsrc(0);
|
| + // Just let it get destroyed and make sure it doesn't call any methods on the
|
| + // provider interface.
|
| +}
|
| +
|
| +// Test that provider methods aren't called without both a track and an 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.
|
| + rtc::scoped_refptr<VideoRtpSender> sender =
|
| + new VideoRtpSender(&video_provider_);
|
| + EXPECT_TRUE(sender->SetTrack(video_track_));
|
| + EXPECT_TRUE(sender->SetTrack(nullptr));
|
| + sender->SetSsrc(kVideoSsrc);
|
| + sender->SetSsrc(0);
|
| + // Just let it get destroyed and make sure it doesn't call any methods on the
|
| + // provider interface.
|
| +}
|
| +
|
| +// Test that an audio sender calls the expected methods on the provider once
|
| +// it 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<AudioRtpSender> sender =
|
| + new AudioRtpSender(&audio_provider_, 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.
|
| + sender->SetSsrc(kAudioSsrc);
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
|
| + sender->SetTrack(track);
|
| +
|
| + // Calls expected from destructor.
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
|
| +}
|
| +
|
| +// Test that an audio sender calls the expected methods on the provider once
|
| +// it 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<AudioRtpSender> sender =
|
| + new AudioRtpSender(&audio_provider_, 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.
|
| + sender->SetTrack(track);
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
|
| + sender->SetSsrc(kAudioSsrc);
|
| +
|
| + // Calls expected from destructor.
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
|
| +}
|
| +
|
| +// Test that a video sender calls the expected methods on the provider once
|
| +// it 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.
|
| + rtc::scoped_refptr<VideoRtpSender> sender =
|
| + new VideoRtpSender(&video_provider_);
|
| + sender->SetSsrc(kVideoSsrc);
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
|
| + sender->SetTrack(video_track_);
|
| +
|
| + // Calls expected from destructor.
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
|
| + .Times(1);
|
| +}
|
| +
|
| +// Test that a video sender calls the expected methods on the provider once
|
| +// it 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.
|
| + rtc::scoped_refptr<VideoRtpSender> sender =
|
| + new VideoRtpSender(&video_provider_);
|
| + sender->SetTrack(video_track_);
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
|
| + sender->SetSsrc(kVideoSsrc);
|
| +
|
| + // Calls expected from destructor.
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
|
| + .Times(1);
|
| +}
|
| +
|
| +// Test that the sender is disconnected from the provider when its 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.
|
| + rtc::scoped_refptr<AudioTrackInterface> track =
|
| + AudioTrack::Create(kAudioTrackId, nullptr);
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
|
| + rtc::scoped_refptr<AudioRtpSender> sender =
|
| + new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
|
| + sender->SetSsrc(kAudioSsrc);
|
| +
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
|
| + sender->SetSsrc(0);
|
| +
|
| + // Make sure it's SetSsrc that called methods on the provider, and not the
|
| + // destructor.
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
|
| +}
|
| +
|
| +// Test that the sender is disconnected from the provider when its 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.
|
| + AddVideoTrack();
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
|
| + rtc::scoped_refptr<VideoRtpSender> sender =
|
| + new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
|
| + sender->SetSsrc(kVideoSsrc);
|
| +
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
|
| + .Times(1);
|
| + sender->SetSsrc(0);
|
| +
|
| + // Make sure it's SetSsrc that called methods on the provider, and not the
|
| + // destructor.
|
| + EXPECT_CALL(video_provider_, SetVideoSend(_, _, _, _)).Times(0);
|
| +}
|
| +
|
| 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.
|
| + rtc::scoped_refptr<AudioTrackInterface> track =
|
| + AudioTrack::Create(kAudioTrackId, nullptr);
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
|
| + rtc::scoped_refptr<AudioRtpSender> sender =
|
| + new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
|
| + sender->SetSsrc(kAudioSsrc);
|
| +
|
| + // Expect that SetAudioSend will be called before the reference to the track
|
| + // is released.
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, nullptr))
|
| + .Times(1)
|
| + .WillOnce(InvokeWithoutArgs([&track] {
|
| + EXPECT_LT(2, track->AddRef());
|
| + track->Release();
|
| + }));
|
| + EXPECT_TRUE(sender->SetTrack(nullptr));
|
| +
|
| + // Make sure it's SetTrack that called methods on the provider, and not the
|
| + // destructor.
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
|
| +}
|
| +
|
| 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.
|
| + rtc::scoped_refptr<VideoTrackSourceInterface> source(
|
| + FakeVideoTrackSource::Create());
|
| + rtc::scoped_refptr<VideoTrackInterface> track =
|
| + VideoTrack::Create(kVideoTrackId, source);
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _, track.get()));
|
| + rtc::scoped_refptr<VideoRtpSender> sender =
|
| + new VideoRtpSender(track, kStreamLabel1, &video_provider_);
|
| + sender->SetSsrc(kVideoSsrc);
|
| +
|
| + // Expect that SetVideoSend will be called before the reference to the track
|
| + // is released.
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
|
| + .Times(1)
|
| + .WillOnce(InvokeWithoutArgs([&track] {
|
| + EXPECT_LT(2, track->AddRef());
|
| + track->Release();
|
| + }));
|
| + EXPECT_TRUE(sender->SetTrack(nullptr));
|
| +
|
| + // Make sure it's SetTrack that called methods on the provider, and not the
|
| + // destructor.
|
| + EXPECT_CALL(video_provider_, SetVideoSend(_, _, _, _)).Times(0);
|
| +}
|
| +
|
| 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.
|
| + AddVideoTrack();
|
| + rtc::scoped_refptr<AudioTrackInterface> track =
|
| + AudioTrack::Create(kAudioTrackId, nullptr);
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
|
| + rtc::scoped_refptr<AudioRtpSender> sender =
|
| + new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
|
| + sender->SetSsrc(kAudioSsrc);
|
| +
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, true, _, _)).Times(1);
|
| + sender->SetSsrc(kAudioSsrc2);
|
| +
|
| + // Calls expected from destructor.
|
| + EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, false, _, _)).Times(1);
|
| +}
|
| +
|
| TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
|
| - CreateVideoRtpSender();
|
| -
|
| - video_rtp_sender_->SetSsrc(kVideoSsrc2);
|
| - VerifyVideoChannelNoInput(kVideoSsrc);
|
| - VerifyVideoChannelInput(kVideoSsrc2);
|
| -
|
| - video_rtp_sender_ = nullptr;
|
| - VerifyVideoChannelNoInput(kVideoSsrc2);
|
| + AddVideoTrack();
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc, true, _, video_track_.get()));
|
| + rtc::scoped_refptr<VideoRtpSender> sender =
|
| + new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
|
| + sender->SetSsrc(kVideoSsrc);
|
| +
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _, nullptr))
|
| + .Times(1);
|
| + EXPECT_CALL(video_provider_,
|
| + SetVideoSend(kVideoSsrc2, true, _, video_track_.get()))
|
| + .Times(1);
|
| + sender->SetSsrc(kVideoSsrc2);
|
| +
|
| + // Calls expected from destructor.
|
| + EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _, nullptr))
|
| + .Times(1);
|
| }
|
|
|
| TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
|
| CreateAudioRtpSender();
|
|
|
| + EXPECT_CALL(audio_provider_, GetAudioRtpSendParameters(kAudioSsrc))
|
| + .WillOnce(Return(RtpParameters()));
|
| + EXPECT_CALL(audio_provider_, SetAudioRtpSendParameters(kAudioSsrc, _))
|
| + .WillOnce(Return(true));
|
| 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();
|
| }
|
| @@ -564,35 +521,12 @@
|
| TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
|
| CreateVideoRtpSender();
|
|
|
| + EXPECT_CALL(video_provider_, GetVideoRtpSendParameters(kVideoSsrc))
|
| + .WillOnce(Return(RtpParameters()));
|
| + EXPECT_CALL(video_provider_, SetVideoRtpSendParameters(kVideoSsrc, _))
|
| + .WillOnce(Return(true));
|
| 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();
|
| }
|
| @@ -600,8 +534,11 @@
|
| TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
|
| CreateAudioRtpReceiver();
|
|
|
| + EXPECT_CALL(audio_provider_, GetAudioRtpReceiveParameters(kAudioSsrc))
|
| + .WillOnce(Return(RtpParameters()));
|
| + EXPECT_CALL(audio_provider_, SetAudioRtpReceiveParameters(kAudioSsrc, _))
|
| + .WillOnce(Return(true));
|
| RtpParameters params = audio_rtp_receiver_->GetParameters();
|
| - EXPECT_EQ(1u, params.encodings.size());
|
| EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
|
|
|
| DestroyAudioRtpReceiver();
|
| @@ -610,8 +547,11 @@
|
| TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
|
| CreateVideoRtpReceiver();
|
|
|
| + EXPECT_CALL(video_provider_, GetVideoRtpReceiveParameters(kVideoSsrc))
|
| + .WillOnce(Return(RtpParameters()));
|
| + EXPECT_CALL(video_provider_, SetVideoRtpReceiveParameters(kVideoSsrc, _))
|
| + .WillOnce(Return(true));
|
| RtpParameters params = video_rtp_receiver_->GetParameters();
|
| - EXPECT_EQ(1u, params.encodings.size());
|
| EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
|
|
|
| DestroyVideoRtpReceiver();
|
|
|