24     const webrtc::AudioSendStream::Config& config)  in FakeAudioSendStream()
28     const webrtc::AudioSendStream::Config& config) {  in Reconfigure()
32 const webrtc::AudioSendStream::Config& FakeAudioSendStream::GetConfig() const {  in GetConfig()
37     const webrtc::AudioSendStream::Stats& stats) {  in SetStats()
61 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {  in GetStats()
65 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats(  in GetStats()
72     const webrtc::AudioReceiveStream::Config& config)  in FakeAudioReceiveStream()
75 const webrtc::AudioReceiveStream::Config& FakeAudioReceiveStream::GetConfig()  in GetConfig()
81     const webrtc::AudioReceiveStream::Stats& stats) {  in SetStats()
99     const webrtc::AudioReceiveStream::Config& config) {  in Reconfigure()
103 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {  in GetStats()
107 void FakeAudioReceiveStream::SetSink(webrtc::AudioSinkInterface* sink) {  in SetSink()
116     webrtc::VideoSendStream::Config config,  in FakeVideoSendStream()
117     webrtc::VideoEncoderConfig encoder_config)  in FakeVideoSendStream()
135 const webrtc::VideoSendStream::Config& FakeVideoSendStream::GetConfig() const {  in GetConfig()
139 const webrtc::VideoEncoderConfig& FakeVideoSendStream::GetEncoderConfig()  in GetEncoderConfig()
144 const std::vector<webrtc::VideoStream>& FakeVideoSendStream::GetVideoStreams()  in GetVideoStreams()
154     webrtc::VideoCodecVP8* settings) const {  in GetVp8Settings()
164     webrtc::VideoCodecVP9* settings) const {  in GetVp9Settings()
174     webrtc::VideoCodecH264* settings) const {  in GetH264Settings()
200 void FakeVideoSendStream::OnFrame(const webrtc::VideoFrame& frame) {  in OnFrame()
212     const webrtc::VideoSendStream::Stats& stats) {  in SetStats()
216 webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {  in GetStats()
221     webrtc::VideoEncoderConfig config) {  in ReconfigureVideoEncoder()
283     rtc::scoped_refptr<webrtc::Resource> resource) {}  in AddAdaptationResource()
285 std::vector<rtc::scoped_refptr<webrtc::Resource>>
291     rtc::VideoSourceInterface<webrtc::VideoFrame>* source,  in SetSource()
292     const webrtc::DegradationPreference& degradation_preference) {  in SetSource()
297     case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:  in SetSource()
301     case webrtc::DegradationPreference::MAINTAIN_RESOLUTION:  in SetSource()
305     case webrtc::DegradationPreference::BALANCED:  in SetSource()
309     case webrtc::DegradationPreference::DISABLED:  in SetSource()
327     webrtc::VideoReceiveStream::Config config)  in FakeVideoReceiveStream()
333 const webrtc::VideoReceiveStream::Config& FakeVideoReceiveStream::GetConfig()  in GetConfig()
342 void FakeVideoReceiveStream::InjectFrame(const webrtc::VideoFrame& frame) {  in InjectFrame()
346 webrtc::VideoReceiveStream::Stats FakeVideoReceiveStream::GetStats() const {  in GetStats()
359     const webrtc::VideoReceiveStream::Stats& stats) {  in SetStats()
364     webrtc::RtpPacketSinkInterface* sink) {  in AddSecondarySink()
369     const webrtc::RtpPacketSinkInterface* sink) {  in RemoveSecondarySink()
382     const webrtc::FlexfecReceiveStream::Config& config)  in FakeFlexfecReceiveStream()
385 const webrtc::FlexfecReceiveStream::Config&
391 webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const {  in GetStats()
392   return webrtc::FlexfecReceiveStream::Stats();  in GetStats()
395 void FakeFlexfecReceiveStream::OnRtpPacket(const webrtc::RtpPacketReceived&) {  in OnRtpPacket()
400     : audio_network_state_(webrtc::kNetworkUp),  in FakeCall()
401       video_network_state_(webrtc::kNetworkUp),  in FakeCall()
460 webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {  in GetNetworkState()
462     case webrtc::MediaType::AUDIO:  in GetNetworkState()
464     case webrtc::MediaType::VIDEO:  in GetNetworkState()
466     case webrtc::MediaType::DATA:  in GetNetworkState()
467     case webrtc::MediaType::ANY:  in GetNetworkState()
469       return webrtc::kNetworkDown;  in GetNetworkState()
475   return webrtc::kNetworkDown;  in GetNetworkState()
478 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(  in CreateAudioSendStream()
479     const webrtc::AudioSendStream::Config& config) {  in CreateAudioSendStream()
487 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {  in DestroyAudioSendStream()
498 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(  in CreateAudioReceiveStream()
499     const webrtc::AudioReceiveStream::Config& config) {  in CreateAudioReceiveStream()
507     webrtc::AudioReceiveStream* receive_stream) {  in DestroyAudioReceiveStream()
518 webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(  in CreateVideoSendStream()
519     webrtc::VideoSendStream::Config config,  in CreateVideoSendStream()
520     webrtc::VideoEncoderConfig encoder_config) {  in CreateVideoSendStream()
528 void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {  in DestroyVideoSendStream()
539 webrtc::VideoReceiveStream* FakeCall::CreateVideoReceiveStream(  in CreateVideoReceiveStream()
540     webrtc::VideoReceiveStream::Config config) {  in CreateVideoReceiveStream()
548     webrtc::VideoReceiveStream* receive_stream) {  in DestroyVideoReceiveStream()
559 webrtc::FlexfecReceiveStream* FakeCall::CreateFlexfecReceiveStream(  in CreateFlexfecReceiveStream()
560     const webrtc::FlexfecReceiveStream::Config& config) {  in CreateFlexfecReceiveStream()
568     webrtc::FlexfecReceiveStream* receive_stream) {  in DestroyFlexfecReceiveStream()
582     rtc::scoped_refptr<webrtc::Resource> resource) {}  in AddAdaptationResource()
584 webrtc::PacketReceiver* FakeCall::Receiver() {  in Receiver()
588 FakeCall::DeliveryStatus FakeCall::DeliverPacket(webrtc::MediaType media_type,  in DeliverPacket()
592   RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||  in DeliverPacket()
593              media_type == webrtc::MediaType::VIDEO);  in DeliverPacket()
599   if (media_type == webrtc::MediaType::VIDEO) {  in DeliverPacket()
605   if (media_type == webrtc::MediaType::AUDIO) {  in DeliverPacket()
616 void FakeCall::SetStats(const webrtc::Call::Stats& stats) {  in SetStats()
628 webrtc::Call::Stats FakeCall::GetStats() const {  in GetStats()
632 void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,  in SignalChannelNetworkState()
633                                          webrtc::NetworkState state) {  in SignalChannelNetworkState()
635     case webrtc::MediaType::AUDIO:  in SignalChannelNetworkState()
638     case webrtc::MediaType::VIDEO:  in SignalChannelNetworkState()
641     case webrtc::MediaType::DATA:  in SignalChannelNetworkState()
642     case webrtc::MediaType::ANY:  in SignalChannelNetworkState()