| /* |
| * Copyright 2018 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 "media/base/fake_media_engine.h" |
| |
| #include <memory> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/strings/match.h" |
| #include "absl/types/optional.h" |
| #include "rtc_base/checks.h" |
| |
| namespace cricket { |
| using webrtc::TaskQueueBase; |
| |
| FakeVoiceMediaChannel::DtmfInfo::DtmfInfo(uint32_t ssrc, |
| int event_code, |
| int duration) |
| : ssrc(ssrc), event_code(event_code), duration(duration) {} |
| |
| FakeVoiceMediaChannel::VoiceChannelAudioSink::VoiceChannelAudioSink( |
| AudioSource* source) |
| : source_(source) { |
| source_->SetSink(this); |
| } |
| FakeVoiceMediaChannel::VoiceChannelAudioSink::~VoiceChannelAudioSink() { |
| if (source_) { |
| source_->SetSink(nullptr); |
| } |
| } |
| void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnData( |
| const void* audio_data, |
| int bits_per_sample, |
| int sample_rate, |
| size_t number_of_channels, |
| size_t number_of_frames, |
| absl::optional<int64_t> absolute_capture_timestamp_ms) {} |
| void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnClose() { |
| source_ = nullptr; |
| } |
| AudioSource* FakeVoiceMediaChannel::VoiceChannelAudioSink::source() const { |
| return source_; |
| } |
| |
| FakeVoiceMediaChannel::FakeVoiceMediaChannel(MediaChannel::Role role, |
| FakeVoiceEngine* engine, |
| const AudioOptions& options, |
| TaskQueueBase* network_thread) |
| : RtpHelper<VoiceMediaChannel>(role, network_thread), |
| engine_(engine), |
| max_bps_(-1) { |
| output_scalings_[0] = 1.0; // For default channel. |
| SetOptions(options); |
| } |
| FakeVoiceMediaChannel::~FakeVoiceMediaChannel() { |
| if (engine_) { |
| engine_->UnregisterChannel(this); |
| } |
| } |
| const std::vector<AudioCodec>& FakeVoiceMediaChannel::recv_codecs() const { |
| return recv_codecs_; |
| } |
| const std::vector<AudioCodec>& FakeVoiceMediaChannel::send_codecs() const { |
| return send_codecs_; |
| } |
| const std::vector<AudioCodec>& FakeVoiceMediaChannel::codecs() const { |
| return send_codecs(); |
| } |
| const std::vector<FakeVoiceMediaChannel::DtmfInfo>& |
| FakeVoiceMediaChannel::dtmf_info_queue() const { |
| return dtmf_info_queue_; |
| } |
| const AudioOptions& FakeVoiceMediaChannel::options() const { |
| return options_; |
| } |
| int FakeVoiceMediaChannel::max_bps() const { |
| return max_bps_; |
| } |
| bool FakeVoiceMediaChannel::SetSendParameters( |
| const AudioSendParameters& params) { |
| set_send_rtcp_parameters(params.rtcp); |
| return (SetSendCodecs(params.codecs) && |
| SetSendExtmapAllowMixed(params.extmap_allow_mixed) && |
| SetSendRtpHeaderExtensions(params.extensions) && |
| SetMaxSendBandwidth(params.max_bandwidth_bps) && |
| SetOptions(params.options)); |
| } |
| bool FakeVoiceMediaChannel::SetRecvParameters( |
| const AudioRecvParameters& params) { |
| set_recv_rtcp_parameters(params.rtcp); |
| return (SetRecvCodecs(params.codecs) && |
| SetRecvRtpHeaderExtensions(params.extensions)); |
| } |
| void FakeVoiceMediaChannel::SetPlayout(bool playout) { |
| set_playout(playout); |
| } |
| void FakeVoiceMediaChannel::SetSend(bool send) { |
| set_sending(send); |
| } |
| bool FakeVoiceMediaChannel::SetAudioSend(uint32_t ssrc, |
| bool enable, |
| const AudioOptions* options, |
| AudioSource* source) { |
| if (!SetLocalSource(ssrc, source)) { |
| return false; |
| } |
| if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) { |
| return false; |
| } |
| if (enable && options) { |
| return SetOptions(*options); |
| } |
| return true; |
| } |
| bool FakeVoiceMediaChannel::HasSource(uint32_t ssrc) const { |
| return local_sinks_.find(ssrc) != local_sinks_.end(); |
| } |
| bool FakeVoiceMediaChannel::AddRecvStream(const StreamParams& sp) { |
| if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp)) |
| return false; |
| output_scalings_[sp.first_ssrc()] = 1.0; |
| output_delays_[sp.first_ssrc()] = 0; |
| return true; |
| } |
| bool FakeVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { |
| if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc)) |
| return false; |
| output_scalings_.erase(ssrc); |
| output_delays_.erase(ssrc); |
| return true; |
| } |
| bool FakeVoiceMediaChannel::CanInsertDtmf() { |
| for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin(); |
| it != send_codecs_.end(); ++it) { |
| // Find the DTMF telephone event "codec". |
| if (absl::EqualsIgnoreCase(it->name, "telephone-event")) { |
| return true; |
| } |
| } |
| return false; |
| } |
| bool FakeVoiceMediaChannel::InsertDtmf(uint32_t ssrc, |
| int event_code, |
| int duration) { |
| dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration)); |
| return true; |
| } |
| bool FakeVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { |
| if (output_scalings_.find(ssrc) != output_scalings_.end()) { |
| output_scalings_[ssrc] = volume; |
| return true; |
| } |
| return false; |
| } |
| bool FakeVoiceMediaChannel::SetDefaultOutputVolume(double volume) { |
| for (auto& entry : output_scalings_) { |
| entry.second = volume; |
| } |
| return true; |
| } |
| bool FakeVoiceMediaChannel::GetOutputVolume(uint32_t ssrc, double* volume) { |
| if (output_scalings_.find(ssrc) == output_scalings_.end()) |
| return false; |
| *volume = output_scalings_[ssrc]; |
| return true; |
| } |
| bool FakeVoiceMediaChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, |
| int delay_ms) { |
| if (output_delays_.find(ssrc) == output_delays_.end()) { |
| return false; |
| } else { |
| output_delays_[ssrc] = delay_ms; |
| return true; |
| } |
| } |
| absl::optional<int> FakeVoiceMediaChannel::GetBaseMinimumPlayoutDelayMs( |
| uint32_t ssrc) const { |
| const auto it = output_delays_.find(ssrc); |
| if (it != output_delays_.end()) { |
| return it->second; |
| } |
| return absl::nullopt; |
| } |
| bool FakeVoiceMediaChannel::GetSendStats(VoiceMediaSendInfo* info) { |
| return false; |
| } |
| bool FakeVoiceMediaChannel::GetReceiveStats(VoiceMediaReceiveInfo* info, |
| bool get_and_clear_legacy_stats) { |
| return false; |
| } |
| void FakeVoiceMediaChannel::SetRawAudioSink( |
| uint32_t ssrc, |
| std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
| sink_ = std::move(sink); |
| } |
| void FakeVoiceMediaChannel::SetDefaultRawAudioSink( |
| std::unique_ptr<webrtc::AudioSinkInterface> sink) { |
| sink_ = std::move(sink); |
| } |
| std::vector<webrtc::RtpSource> FakeVoiceMediaChannel::GetSources( |
| uint32_t ssrc) const { |
| return std::vector<webrtc::RtpSource>(); |
| } |
| bool FakeVoiceMediaChannel::SetRecvCodecs( |
| const std::vector<AudioCodec>& codecs) { |
| if (fail_set_recv_codecs()) { |
| // Fake the failure in SetRecvCodecs. |
| return false; |
| } |
| recv_codecs_ = codecs; |
| return true; |
| } |
| bool FakeVoiceMediaChannel::SetSendCodecs( |
| const std::vector<AudioCodec>& codecs) { |
| if (fail_set_send_codecs()) { |
| // Fake the failure in SetSendCodecs. |
| return false; |
| } |
| send_codecs_ = codecs; |
| return true; |
| } |
| bool FakeVoiceMediaChannel::SetMaxSendBandwidth(int bps) { |
| max_bps_ = bps; |
| return true; |
| } |
| bool FakeVoiceMediaChannel::SetOptions(const AudioOptions& options) { |
| // Does a "merge" of current options and set options. |
| options_.SetAll(options); |
| return true; |
| } |
| bool FakeVoiceMediaChannel::SetLocalSource(uint32_t ssrc, AudioSource* source) { |
| auto it = local_sinks_.find(ssrc); |
| if (source) { |
| if (it != local_sinks_.end()) { |
| RTC_CHECK(it->second->source() == source); |
| } else { |
| local_sinks_.insert(std::make_pair( |
| ssrc, std::make_unique<VoiceChannelAudioSink>(source))); |
| } |
| } else { |
| if (it != local_sinks_.end()) { |
| local_sinks_.erase(it); |
| } |
| } |
| return true; |
| } |
| |
| bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info, |
| uint32_t ssrc, |
| int event_code, |
| int duration) { |
| return (info.duration == duration && info.event_code == event_code && |
| info.ssrc == ssrc); |
| } |
| |
| FakeVideoMediaChannel::FakeVideoMediaChannel(MediaChannel::Role role, |
| FakeVideoEngine* engine, |
| const VideoOptions& options, |
| TaskQueueBase* network_thread) |
| : RtpHelper<VideoMediaChannel>(role, network_thread), |
| engine_(engine), |
| max_bps_(-1) { |
| SetOptions(options); |
| } |
| FakeVideoMediaChannel::~FakeVideoMediaChannel() { |
| if (engine_) { |
| engine_->UnregisterChannel(this); |
| } |
| } |
| const std::vector<VideoCodec>& FakeVideoMediaChannel::recv_codecs() const { |
| return recv_codecs_; |
| } |
| const std::vector<VideoCodec>& FakeVideoMediaChannel::send_codecs() const { |
| return send_codecs_; |
| } |
| const std::vector<VideoCodec>& FakeVideoMediaChannel::codecs() const { |
| return send_codecs(); |
| } |
| bool FakeVideoMediaChannel::rendering() const { |
| return playout(); |
| } |
| const VideoOptions& FakeVideoMediaChannel::options() const { |
| return options_; |
| } |
| const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>& |
| FakeVideoMediaChannel::sinks() const { |
| return sinks_; |
| } |
| int FakeVideoMediaChannel::max_bps() const { |
| return max_bps_; |
| } |
| bool FakeVideoMediaChannel::SetSendParameters( |
| const VideoSendParameters& params) { |
| set_send_rtcp_parameters(params.rtcp); |
| return (SetSendCodecs(params.codecs) && |
| SetSendExtmapAllowMixed(params.extmap_allow_mixed) && |
| SetSendRtpHeaderExtensions(params.extensions) && |
| SetMaxSendBandwidth(params.max_bandwidth_bps)); |
| } |
| bool FakeVideoMediaChannel::SetRecvParameters( |
| const VideoRecvParameters& params) { |
| set_recv_rtcp_parameters(params.rtcp); |
| return (SetRecvCodecs(params.codecs) && |
| SetRecvRtpHeaderExtensions(params.extensions)); |
| } |
| bool FakeVideoMediaChannel::AddSendStream(const StreamParams& sp) { |
| return RtpHelper<VideoMediaChannel>::AddSendStream(sp); |
| } |
| bool FakeVideoMediaChannel::RemoveSendStream(uint32_t ssrc) { |
| return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc); |
| } |
| bool FakeVideoMediaChannel::GetSendCodec(VideoCodec* send_codec) { |
| if (send_codecs_.empty()) { |
| return false; |
| } |
| *send_codec = send_codecs_[0]; |
| return true; |
| } |
| bool FakeVideoMediaChannel::SetSink( |
| uint32_t ssrc, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) { |
| auto it = sinks_.find(ssrc); |
| if (it == sinks_.end()) { |
| return false; |
| } |
| it->second = sink; |
| return true; |
| } |
| void FakeVideoMediaChannel::SetDefaultSink( |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {} |
| bool FakeVideoMediaChannel::HasSink(uint32_t ssrc) const { |
| return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr; |
| } |
| bool FakeVideoMediaChannel::SetSend(bool send) { |
| return set_sending(send); |
| } |
| bool FakeVideoMediaChannel::SetVideoSend( |
| uint32_t ssrc, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) { |
| if (options) { |
| if (!SetOptions(*options)) { |
| return false; |
| } |
| } |
| sources_[ssrc] = source; |
| return true; |
| } |
| bool FakeVideoMediaChannel::HasSource(uint32_t ssrc) const { |
| return sources_.find(ssrc) != sources_.end() && sources_.at(ssrc) != nullptr; |
| } |
| bool FakeVideoMediaChannel::AddRecvStream(const StreamParams& sp) { |
| if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp)) |
| return false; |
| sinks_[sp.first_ssrc()] = NULL; |
| output_delays_[sp.first_ssrc()] = 0; |
| return true; |
| } |
| bool FakeVideoMediaChannel::RemoveRecvStream(uint32_t ssrc) { |
| if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc)) |
| return false; |
| sinks_.erase(ssrc); |
| output_delays_.erase(ssrc); |
| return true; |
| } |
| void FakeVideoMediaChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) { |
| } |
| bool FakeVideoMediaChannel::GetSendStats(VideoMediaSendInfo* info) { |
| return false; |
| } |
| bool FakeVideoMediaChannel::GetReceiveStats(VideoMediaReceiveInfo* info) { |
| return false; |
| } |
| std::vector<webrtc::RtpSource> FakeVideoMediaChannel::GetSources( |
| uint32_t ssrc) const { |
| return {}; |
| } |
| bool FakeVideoMediaChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, |
| int delay_ms) { |
| if (output_delays_.find(ssrc) == output_delays_.end()) { |
| return false; |
| } else { |
| output_delays_[ssrc] = delay_ms; |
| return true; |
| } |
| } |
| absl::optional<int> FakeVideoMediaChannel::GetBaseMinimumPlayoutDelayMs( |
| uint32_t ssrc) const { |
| const auto it = output_delays_.find(ssrc); |
| if (it != output_delays_.end()) { |
| return it->second; |
| } |
| return absl::nullopt; |
| } |
| bool FakeVideoMediaChannel::SetRecvCodecs( |
| const std::vector<VideoCodec>& codecs) { |
| if (fail_set_recv_codecs()) { |
| // Fake the failure in SetRecvCodecs. |
| return false; |
| } |
| recv_codecs_ = codecs; |
| return true; |
| } |
| bool FakeVideoMediaChannel::SetSendCodecs( |
| const std::vector<VideoCodec>& codecs) { |
| if (fail_set_send_codecs()) { |
| // Fake the failure in SetSendCodecs. |
| return false; |
| } |
| send_codecs_ = codecs; |
| |
| return true; |
| } |
| bool FakeVideoMediaChannel::SetOptions(const VideoOptions& options) { |
| options_ = options; |
| return true; |
| } |
| |
| bool FakeVideoMediaChannel::SetMaxSendBandwidth(int bps) { |
| max_bps_ = bps; |
| return true; |
| } |
| |
| void FakeVideoMediaChannel::SetRecordableEncodedFrameCallback( |
| uint32_t ssrc, |
| std::function<void(const webrtc::RecordableEncodedFrame&)> callback) {} |
| |
| void FakeVideoMediaChannel::ClearRecordableEncodedFrameCallback(uint32_t ssrc) { |
| } |
| |
| void FakeVideoMediaChannel::RequestRecvKeyFrame(uint32_t ssrc) {} |
| void FakeVideoMediaChannel::GenerateSendKeyFrame( |
| uint32_t ssrc, |
| const std::vector<std::string>& rids) {} |
| |
| FakeVoiceEngine::FakeVoiceEngine() : fail_create_channel_(false) { |
| // Add a fake audio codec. Note that the name must not be "" as there are |
| // sanity checks against that. |
| SetCodecs({AudioCodec(101, "fake_audio_codec", 0, 0, 1)}); |
| } |
| void FakeVoiceEngine::Init() {} |
| rtc::scoped_refptr<webrtc::AudioState> FakeVoiceEngine::GetAudioState() const { |
| return rtc::scoped_refptr<webrtc::AudioState>(); |
| } |
| VoiceMediaChannel* FakeVoiceEngine::CreateMediaChannel( |
| MediaChannel::Role role, |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const AudioOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::AudioCodecPairId codec_pair_id) { |
| if (fail_create_channel_) { |
| return nullptr; |
| } |
| |
| FakeVoiceMediaChannel* ch = |
| new FakeVoiceMediaChannel(role, this, options, call->network_thread()); |
| switch (role) { |
| case MediaChannel::Role::kSend: |
| send_channels_.push_back(ch); |
| break; |
| case MediaChannel::Role::kReceive: |
| receive_channels_.push_back(ch); |
| break; |
| case MediaChannel::Role::kBoth: |
| send_channels_.push_back(ch); |
| receive_channels_.push_back(ch); |
| break; |
| default: |
| RTC_CHECK_NOTREACHED(); |
| } |
| return ch; |
| } |
| FakeVoiceMediaChannel* FakeVoiceEngine::GetSendChannel(size_t index) { |
| return (send_channels_.size() > index) ? send_channels_[index] : NULL; |
| } |
| FakeVoiceMediaChannel* FakeVoiceEngine::GetReceiveChannel(size_t index) { |
| return (receive_channels_.size() > index) ? receive_channels_[index] : NULL; |
| } |
| void FakeVoiceEngine::UnregisterChannel(VoiceMediaChannel* channel) { |
| switch (channel->role()) { |
| case MediaChannel::Role::kSend: |
| send_channels_.erase(absl::c_find(send_channels_, channel)); |
| break; |
| case MediaChannel::Role::kReceive: |
| receive_channels_.erase(absl::c_find(receive_channels_, channel)); |
| break; |
| case MediaChannel::Role::kBoth: |
| send_channels_.erase(absl::c_find(send_channels_, channel)); |
| receive_channels_.erase(absl::c_find(receive_channels_, channel)); |
| break; |
| default: |
| RTC_CHECK_NOTREACHED(); |
| } |
| } |
| const std::vector<AudioCodec>& FakeVoiceEngine::send_codecs() const { |
| return send_codecs_; |
| } |
| const std::vector<AudioCodec>& FakeVoiceEngine::recv_codecs() const { |
| return recv_codecs_; |
| } |
| void FakeVoiceEngine::SetCodecs(const std::vector<AudioCodec>& codecs) { |
| send_codecs_ = codecs; |
| recv_codecs_ = codecs; |
| } |
| void FakeVoiceEngine::SetRecvCodecs(const std::vector<AudioCodec>& codecs) { |
| recv_codecs_ = codecs; |
| } |
| void FakeVoiceEngine::SetSendCodecs(const std::vector<AudioCodec>& codecs) { |
| send_codecs_ = codecs; |
| } |
| int FakeVoiceEngine::GetInputLevel() { |
| return 0; |
| } |
| bool FakeVoiceEngine::StartAecDump(webrtc::FileWrapper file, |
| int64_t max_size_bytes) { |
| return false; |
| } |
| absl::optional<webrtc::AudioDeviceModule::Stats> |
| FakeVoiceEngine::GetAudioDeviceStats() { |
| return absl::nullopt; |
| } |
| void FakeVoiceEngine::StopAecDump() {} |
| |
| std::vector<webrtc::RtpHeaderExtensionCapability> |
| FakeVoiceEngine::GetRtpHeaderExtensions() const { |
| return header_extensions_; |
| } |
| |
| void FakeVoiceEngine::SetRtpHeaderExtensions( |
| std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions) { |
| header_extensions_ = std::move(header_extensions); |
| } |
| |
| FakeVideoEngine::FakeVideoEngine() |
| : capture_(false), fail_create_channel_(false) { |
| // Add a fake video codec. Note that the name must not be "" as there are |
| // sanity checks against that. |
| send_codecs_.push_back(VideoCodec(0, "fake_video_codec")); |
| recv_codecs_.push_back(VideoCodec(0, "fake_video_codec")); |
| } |
| bool FakeVideoEngine::SetOptions(const VideoOptions& options) { |
| options_ = options; |
| return true; |
| } |
| VideoMediaChannel* FakeVideoEngine::CreateMediaChannel( |
| MediaChannel::Role role, |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) { |
| if (fail_create_channel_) { |
| return nullptr; |
| } |
| |
| FakeVideoMediaChannel* ch = |
| new FakeVideoMediaChannel(role, this, options, call->network_thread()); |
| switch (role) { |
| case MediaChannel::Role::kSend: |
| send_channels_.emplace_back(ch); |
| break; |
| case MediaChannel::Role::kReceive: |
| receive_channels_.emplace_back(ch); |
| break; |
| case MediaChannel::Role::kBoth: |
| send_channels_.push_back(ch); |
| receive_channels_.push_back(ch); |
| break; |
| default: |
| RTC_CHECK_NOTREACHED(); |
| } |
| return ch; |
| } |
| FakeVideoMediaChannel* FakeVideoEngine::GetSendChannel(size_t index) { |
| return (send_channels_.size() > index) ? send_channels_[index] : nullptr; |
| } |
| FakeVideoMediaChannel* FakeVideoEngine::GetReceiveChannel(size_t index) { |
| return (receive_channels_.size() > index) ? receive_channels_[index] |
| : nullptr; |
| } |
| void FakeVideoEngine::UnregisterChannel(VideoMediaChannel* channel) { |
| switch (channel->role()) { |
| case MediaChannel::Role::kSend: { |
| auto it = absl::c_find(send_channels_, channel); |
| RTC_DCHECK(it != send_channels_.end()); |
| send_channels_.erase(it); |
| } break; |
| case MediaChannel::Role::kReceive: { |
| auto it = absl::c_find(receive_channels_, channel); |
| RTC_DCHECK(it != receive_channels_.end()); |
| receive_channels_.erase(it); |
| } break; |
| case MediaChannel::Role::kBoth: { |
| auto it = absl::c_find(send_channels_, channel); |
| RTC_DCHECK(it != send_channels_.end()); |
| send_channels_.erase(it); |
| it = absl::c_find(receive_channels_, channel); |
| RTC_DCHECK(it != receive_channels_.end()); |
| receive_channels_.erase(it); |
| } break; |
| default: |
| RTC_CHECK_NOTREACHED(); |
| } |
| } |
| std::vector<VideoCodec> FakeVideoEngine::send_codecs(bool use_rtx) const { |
| return send_codecs_; |
| } |
| |
| std::vector<VideoCodec> FakeVideoEngine::recv_codecs(bool use_rtx) const { |
| return recv_codecs_; |
| } |
| |
| void FakeVideoEngine::SetSendCodecs(const std::vector<VideoCodec>& codecs) { |
| send_codecs_ = codecs; |
| } |
| |
| void FakeVideoEngine::SetRecvCodecs(const std::vector<VideoCodec>& codecs) { |
| recv_codecs_ = codecs; |
| } |
| |
| bool FakeVideoEngine::SetCapture(bool capture) { |
| capture_ = capture; |
| return true; |
| } |
| std::vector<webrtc::RtpHeaderExtensionCapability> |
| FakeVideoEngine::GetRtpHeaderExtensions() const { |
| return header_extensions_; |
| } |
| void FakeVideoEngine::SetRtpHeaderExtensions( |
| std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions) { |
| header_extensions_ = std::move(header_extensions); |
| } |
| |
| FakeMediaEngine::FakeMediaEngine() |
| : CompositeMediaEngine(std::make_unique<FakeVoiceEngine>(), |
| std::make_unique<FakeVideoEngine>()), |
| voice_(static_cast<FakeVoiceEngine*>(&voice())), |
| video_(static_cast<FakeVideoEngine*>(&video())) {} |
| FakeMediaEngine::~FakeMediaEngine() {} |
| void FakeMediaEngine::SetAudioCodecs(const std::vector<AudioCodec>& codecs) { |
| voice_->SetCodecs(codecs); |
| } |
| void FakeMediaEngine::SetAudioRecvCodecs( |
| const std::vector<AudioCodec>& codecs) { |
| voice_->SetRecvCodecs(codecs); |
| } |
| void FakeMediaEngine::SetAudioSendCodecs( |
| const std::vector<AudioCodec>& codecs) { |
| voice_->SetSendCodecs(codecs); |
| } |
| void FakeMediaEngine::SetVideoCodecs(const std::vector<VideoCodec>& codecs) { |
| video_->SetSendCodecs(codecs); |
| video_->SetRecvCodecs(codecs); |
| } |
| |
| FakeVoiceMediaChannel* FakeMediaEngine::GetVoiceSendChannel(size_t index) { |
| return voice_->GetSendChannel(index); |
| } |
| FakeVideoMediaChannel* FakeMediaEngine::GetVideoSendChannel(size_t index) { |
| return video_->GetSendChannel(index); |
| } |
| FakeVoiceMediaChannel* FakeMediaEngine::GetVoiceReceiveChannel(size_t index) { |
| return voice_->GetReceiveChannel(index); |
| } |
| FakeVideoMediaChannel* FakeMediaEngine::GetVideoReceiveChannel(size_t index) { |
| return video_->GetReceiveChannel(index); |
| } |
| |
| void FakeMediaEngine::set_fail_create_channel(bool fail) { |
| voice_->fail_create_channel_ = fail; |
| video_->fail_create_channel_ = fail; |
| } |
| |
| } // namespace cricket |