| /* |
| * Copyright (c) 2004 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. |
| */ |
| |
| #ifndef MEDIA_BASE_FAKE_MEDIA_ENGINE_H_ |
| #define MEDIA_BASE_FAKE_MEDIA_ENGINE_H_ |
| |
| #include <atomic> |
| #include <list> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <tuple> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/functional/any_invocable.h" |
| #include "api/call/audio_sink.h" |
| #include "api/media_types.h" |
| #include "media/base/audio_source.h" |
| #include "media/base/media_channel.h" |
| #include "media/base/media_channel_impl.h" |
| #include "media/base/media_engine.h" |
| #include "media/base/rtp_utils.h" |
| #include "media/base/stream_params.h" |
| #include "media/engine/webrtc_video_engine.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/network_route.h" |
| #include "rtc_base/thread.h" |
| |
| using webrtc::RtpExtension; |
| |
| namespace cricket { |
| |
| class FakeMediaEngine; |
| class FakeVideoEngine; |
| class FakeVoiceEngine; |
| |
| // A common helper class that handles sending and receiving RTP/RTCP packets. |
| template <class Base> |
| class RtpReceiveChannelHelper : public Base, public MediaChannelUtil { |
| public: |
| explicit RtpReceiveChannelHelper(webrtc::TaskQueueBase* network_thread) |
| : MediaChannelUtil(network_thread), |
| playout_(false), |
| fail_set_recv_codecs_(false), |
| transport_overhead_per_packet_(0), |
| num_network_route_changes_(0) {} |
| virtual ~RtpReceiveChannelHelper() = default; |
| const std::vector<RtpExtension>& recv_extensions() { |
| return recv_extensions_; |
| } |
| bool playout() const { return playout_; } |
| const std::list<std::string>& rtp_packets() const { return rtp_packets_; } |
| const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; } |
| |
| bool SendRtcp(const void* data, size_t len) { |
| rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len, |
| kMaxRtpPacketLen); |
| return Base::SendRtcp(&packet, rtc::PacketOptions()); |
| } |
| |
| bool CheckRtp(const void* data, size_t len) { |
| bool success = !rtp_packets_.empty(); |
| if (success) { |
| std::string packet = rtp_packets_.front(); |
| rtp_packets_.pop_front(); |
| success = (packet == std::string(static_cast<const char*>(data), len)); |
| } |
| return success; |
| } |
| bool CheckRtcp(const void* data, size_t len) { |
| bool success = !rtcp_packets_.empty(); |
| if (success) { |
| std::string packet = rtcp_packets_.front(); |
| rtcp_packets_.pop_front(); |
| success = (packet == std::string(static_cast<const char*>(data), len)); |
| } |
| return success; |
| } |
| bool CheckNoRtp() { return rtp_packets_.empty(); } |
| bool CheckNoRtcp() { return rtcp_packets_.empty(); } |
| void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; } |
| void ResetUnsignaledRecvStream() override {} |
| absl::optional<uint32_t> GetUnsignaledSsrc() const override { |
| return absl::nullopt; |
| } |
| void ChooseReceiverReportSsrc(const std::set<uint32_t>& choices) override {} |
| |
| virtual bool SetLocalSsrc(const StreamParams& sp) { return true; } |
| void OnDemuxerCriteriaUpdatePending() override {} |
| void OnDemuxerCriteriaUpdateComplete() override {} |
| |
| bool AddRecvStream(const StreamParams& sp) override { |
| if (absl::c_linear_search(receive_streams_, sp)) { |
| return false; |
| } |
| receive_streams_.push_back(sp); |
| rtp_receive_parameters_[sp.first_ssrc()] = |
| CreateRtpParametersWithEncodings(sp); |
| return true; |
| } |
| bool RemoveRecvStream(uint32_t ssrc) override { |
| auto parameters_iterator = rtp_receive_parameters_.find(ssrc); |
| if (parameters_iterator != rtp_receive_parameters_.end()) { |
| rtp_receive_parameters_.erase(parameters_iterator); |
| } |
| return RemoveStreamBySsrc(&receive_streams_, ssrc); |
| } |
| |
| webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override { |
| auto parameters_iterator = rtp_receive_parameters_.find(ssrc); |
| if (parameters_iterator != rtp_receive_parameters_.end()) { |
| return parameters_iterator->second; |
| } |
| return webrtc::RtpParameters(); |
| } |
| webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override { |
| return webrtc::RtpParameters(); |
| } |
| |
| const std::vector<StreamParams>& recv_streams() const { |
| return receive_streams_; |
| } |
| bool HasRecvStream(uint32_t ssrc) const { |
| return GetStreamBySsrc(receive_streams_, ssrc) != nullptr; |
| } |
| |
| const RtcpParameters& recv_rtcp_parameters() { return recv_rtcp_parameters_; } |
| |
| int transport_overhead_per_packet() const { |
| return transport_overhead_per_packet_; |
| } |
| |
| rtc::NetworkRoute last_network_route() const { return last_network_route_; } |
| int num_network_route_changes() const { return num_network_route_changes_; } |
| void set_num_network_route_changes(int changes) { |
| num_network_route_changes_ = changes; |
| } |
| |
| void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet, |
| int64_t packet_time_us) { |
| rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size())); |
| } |
| |
| void SetFrameDecryptor(uint32_t ssrc, |
| rtc::scoped_refptr<webrtc::FrameDecryptorInterface> |
| frame_decryptor) override {} |
| |
| void SetDepacketizerToDecoderFrameTransformer( |
| uint32_t ssrc, |
| rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) |
| override {} |
| |
| void SetInterface(MediaChannelNetworkInterface* iface) override { |
| network_interface_ = iface; |
| MediaChannelUtil::SetInterface(iface); |
| } |
| |
| protected: |
| void set_playout(bool playout) { playout_ = playout; } |
| bool SetRecvRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) { |
| recv_extensions_ = extensions; |
| return true; |
| } |
| void set_recv_rtcp_parameters(const RtcpParameters& params) { |
| recv_rtcp_parameters_ = params; |
| } |
| void OnPacketReceived(const webrtc::RtpPacketReceived& packet) override { |
| rtp_packets_.push_back( |
| std::string(packet.Buffer().cdata<char>(), packet.size())); |
| } |
| bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; } |
| |
| private: |
| bool playout_; |
| std::vector<RtpExtension> recv_extensions_; |
| std::list<std::string> rtp_packets_; |
| std::list<std::string> rtcp_packets_; |
| std::vector<StreamParams> receive_streams_; |
| RtcpParameters recv_rtcp_parameters_; |
| std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_; |
| bool fail_set_recv_codecs_; |
| std::string rtcp_cname_; |
| int transport_overhead_per_packet_; |
| rtc::NetworkRoute last_network_route_; |
| int num_network_route_changes_; |
| MediaChannelNetworkInterface* network_interface_ = nullptr; |
| }; |
| |
| // A common helper class that handles sending and receiving RTP/RTCP packets. |
| template <class Base> |
| class RtpSendChannelHelper : public Base, public MediaChannelUtil { |
| public: |
| explicit RtpSendChannelHelper(webrtc::TaskQueueBase* network_thread) |
| : MediaChannelUtil(network_thread), |
| sending_(false), |
| fail_set_send_codecs_(false), |
| send_ssrc_(0), |
| ready_to_send_(false), |
| transport_overhead_per_packet_(0), |
| num_network_route_changes_(0) {} |
| virtual ~RtpSendChannelHelper() = default; |
| const std::vector<RtpExtension>& send_extensions() { |
| return send_extensions_; |
| } |
| bool sending() const { return sending_; } |
| const std::list<std::string>& rtp_packets() const { return rtp_packets_; } |
| const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; } |
| |
| bool SendPacket(const void* data, |
| size_t len, |
| const rtc::PacketOptions& options) { |
| if (!sending_) { |
| return false; |
| } |
| rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len, |
| kMaxRtpPacketLen); |
| return MediaChannelUtil::SendPacket(&packet, options); |
| } |
| bool SendRtcp(const void* data, size_t len) { |
| rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len, |
| kMaxRtpPacketLen); |
| return MediaChannelUtil::SendRtcp(&packet, rtc::PacketOptions()); |
| } |
| |
| bool CheckRtp(const void* data, size_t len) { |
| bool success = !rtp_packets_.empty(); |
| if (success) { |
| std::string packet = rtp_packets_.front(); |
| rtp_packets_.pop_front(); |
| success = (packet == std::string(static_cast<const char*>(data), len)); |
| } |
| return success; |
| } |
| bool CheckRtcp(const void* data, size_t len) { |
| bool success = !rtcp_packets_.empty(); |
| if (success) { |
| std::string packet = rtcp_packets_.front(); |
| rtcp_packets_.pop_front(); |
| success = (packet == std::string(static_cast<const char*>(data), len)); |
| } |
| return success; |
| } |
| bool CheckNoRtp() { return rtp_packets_.empty(); } |
| bool CheckNoRtcp() { return rtcp_packets_.empty(); } |
| void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; } |
| bool AddSendStream(const StreamParams& sp) override { |
| if (absl::c_linear_search(send_streams_, sp)) { |
| return false; |
| } |
| send_streams_.push_back(sp); |
| rtp_send_parameters_[sp.first_ssrc()] = |
| CreateRtpParametersWithEncodings(sp); |
| |
| if (ssrc_list_changed_callback_) { |
| std::set<uint32_t> ssrcs_in_use; |
| for (const auto& send_stream : send_streams_) { |
| ssrcs_in_use.insert(send_stream.first_ssrc()); |
| } |
| ssrc_list_changed_callback_(ssrcs_in_use); |
| } |
| |
| return true; |
| } |
| bool RemoveSendStream(uint32_t ssrc) override { |
| auto parameters_iterator = rtp_send_parameters_.find(ssrc); |
| if (parameters_iterator != rtp_send_parameters_.end()) { |
| rtp_send_parameters_.erase(parameters_iterator); |
| } |
| return RemoveStreamBySsrc(&send_streams_, ssrc); |
| } |
| void SetSsrcListChangedCallback( |
| absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override { |
| ssrc_list_changed_callback_ = std::move(callback); |
| } |
| |
| void SetExtmapAllowMixed(bool extmap_allow_mixed) override { |
| return MediaChannelUtil::SetExtmapAllowMixed(extmap_allow_mixed); |
| } |
| bool ExtmapAllowMixed() const override { |
| return MediaChannelUtil::ExtmapAllowMixed(); |
| } |
| |
| webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override { |
| auto parameters_iterator = rtp_send_parameters_.find(ssrc); |
| if (parameters_iterator != rtp_send_parameters_.end()) { |
| return parameters_iterator->second; |
| } |
| return webrtc::RtpParameters(); |
| } |
| webrtc::RTCError SetRtpSendParameters( |
| uint32_t ssrc, |
| const webrtc::RtpParameters& parameters, |
| webrtc::SetParametersCallback callback) override { |
| auto parameters_iterator = rtp_send_parameters_.find(ssrc); |
| if (parameters_iterator != rtp_send_parameters_.end()) { |
| auto result = CheckRtpParametersInvalidModificationAndValues( |
| parameters_iterator->second, parameters); |
| if (!result.ok()) { |
| return webrtc::InvokeSetParametersCallback(callback, result); |
| } |
| |
| parameters_iterator->second = parameters; |
| |
| return webrtc::InvokeSetParametersCallback(callback, |
| webrtc::RTCError::OK()); |
| } |
| // Replicate the behavior of the real media channel: return false |
| // when setting parameters for unknown SSRCs. |
| return InvokeSetParametersCallback( |
| callback, webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR)); |
| } |
| |
| bool IsStreamMuted(uint32_t ssrc) const { |
| bool ret = muted_streams_.find(ssrc) != muted_streams_.end(); |
| // If |ssrc = 0| check if the first send stream is muted. |
| if (!ret && ssrc == 0 && !send_streams_.empty()) { |
| return muted_streams_.find(send_streams_[0].first_ssrc()) != |
| muted_streams_.end(); |
| } |
| return ret; |
| } |
| const std::vector<StreamParams>& send_streams() const { |
| return send_streams_; |
| } |
| bool HasSendStream(uint32_t ssrc) const { |
| return GetStreamBySsrc(send_streams_, ssrc) != nullptr; |
| } |
| // TODO(perkj): This is to support legacy unit test that only check one |
| // sending stream. |
| uint32_t send_ssrc() const { |
| if (send_streams_.empty()) |
| return 0; |
| return send_streams_[0].first_ssrc(); |
| } |
| |
| const RtcpParameters& send_rtcp_parameters() { return send_rtcp_parameters_; } |
| |
| bool ready_to_send() const { return ready_to_send_; } |
| |
| int transport_overhead_per_packet() const { |
| return transport_overhead_per_packet_; |
| } |
| |
| rtc::NetworkRoute last_network_route() const { return last_network_route_; } |
| int num_network_route_changes() const { return num_network_route_changes_; } |
| void set_num_network_route_changes(int changes) { |
| num_network_route_changes_ = changes; |
| } |
| |
| void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet, |
| int64_t packet_time_us) { |
| rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size())); |
| } |
| |
| // Stuff that deals with encryptors, transformers and the like |
| void SetFrameEncryptor(uint32_t ssrc, |
| rtc::scoped_refptr<webrtc::FrameEncryptorInterface> |
| frame_encryptor) override {} |
| void SetEncoderToPacketizerFrameTransformer( |
| uint32_t ssrc, |
| rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) |
| override {} |
| |
| void SetInterface(MediaChannelNetworkInterface* iface) override { |
| network_interface_ = iface; |
| MediaChannelUtil::SetInterface(iface); |
| } |
| bool HasNetworkInterface() const override { |
| return network_interface_ != nullptr; |
| } |
| |
| protected: |
| bool MuteStream(uint32_t ssrc, bool mute) { |
| if (!HasSendStream(ssrc) && ssrc != 0) { |
| return false; |
| } |
| if (mute) { |
| muted_streams_.insert(ssrc); |
| } else { |
| muted_streams_.erase(ssrc); |
| } |
| return true; |
| } |
| bool set_sending(bool send) { |
| sending_ = send; |
| return true; |
| } |
| bool SetSendRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) { |
| send_extensions_ = extensions; |
| return true; |
| } |
| void set_send_rtcp_parameters(const RtcpParameters& params) { |
| send_rtcp_parameters_ = params; |
| } |
| void OnPacketSent(const rtc::SentPacket& sent_packet) override {} |
| void OnReadyToSend(bool ready) override { ready_to_send_ = ready; } |
| void OnNetworkRouteChanged(absl::string_view transport_name, |
| const rtc::NetworkRoute& network_route) override { |
| last_network_route_ = network_route; |
| ++num_network_route_changes_; |
| transport_overhead_per_packet_ = network_route.packet_overhead; |
| } |
| bool fail_set_send_codecs() const { return fail_set_send_codecs_; } |
| |
| private: |
| // TODO(bugs.webrtc.org/12783): This flag is used from more than one thread. |
| // As a workaround for tsan, it's currently std::atomic but that might not |
| // be the appropriate fix. |
| std::atomic<bool> sending_; |
| std::vector<RtpExtension> send_extensions_; |
| std::list<std::string> rtp_packets_; |
| std::list<std::string> rtcp_packets_; |
| std::vector<StreamParams> send_streams_; |
| RtcpParameters send_rtcp_parameters_; |
| std::set<uint32_t> muted_streams_; |
| std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_; |
| bool fail_set_send_codecs_; |
| uint32_t send_ssrc_; |
| std::string rtcp_cname_; |
| bool ready_to_send_; |
| int transport_overhead_per_packet_; |
| rtc::NetworkRoute last_network_route_; |
| int num_network_route_changes_; |
| MediaChannelNetworkInterface* network_interface_ = nullptr; |
| absl::AnyInvocable<void(const std::set<uint32_t>&)> |
| ssrc_list_changed_callback_ = nullptr; |
| }; |
| |
| class FakeVoiceMediaReceiveChannel |
| : public RtpReceiveChannelHelper<VoiceMediaReceiveChannelInterface> { |
| public: |
| struct DtmfInfo { |
| DtmfInfo(uint32_t ssrc, int event_code, int duration); |
| uint32_t ssrc; |
| int event_code; |
| int duration; |
| }; |
| FakeVoiceMediaReceiveChannel(const AudioOptions& options, |
| webrtc::TaskQueueBase* network_thread); |
| virtual ~FakeVoiceMediaReceiveChannel(); |
| |
| // Test methods |
| const std::vector<AudioCodec>& recv_codecs() const; |
| const std::vector<DtmfInfo>& dtmf_info_queue() const; |
| const AudioOptions& options() const; |
| int max_bps() const; |
| bool HasSource(uint32_t ssrc) const; |
| |
| // Overrides |
| VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override { |
| return nullptr; |
| } |
| VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override { |
| return this; |
| } |
| cricket::MediaType media_type() const override { |
| return cricket::MEDIA_TYPE_AUDIO; |
| } |
| |
| bool SetRecvParameters(const AudioReceiverParameters& params) override; |
| void SetPlayout(bool playout) override; |
| |
| bool AddRecvStream(const StreamParams& sp) override; |
| bool RemoveRecvStream(uint32_t ssrc) override; |
| |
| bool SetOutputVolume(uint32_t ssrc, double volume) override; |
| bool SetDefaultOutputVolume(double volume) override; |
| |
| bool GetOutputVolume(uint32_t ssrc, double* volume); |
| |
| bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override; |
| absl::optional<int> GetBaseMinimumPlayoutDelayMs( |
| uint32_t ssrc) const override; |
| |
| bool GetStats(VoiceMediaReceiveInfo* info, |
| bool get_and_clear_legacy_stats) override; |
| |
| void SetRawAudioSink( |
| uint32_t ssrc, |
| std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
| void SetDefaultRawAudioSink( |
| std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
| |
| std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override; |
| void SetReceiveNackEnabled(bool enabled) override {} |
| void SetReceiveNonSenderRttEnabled(bool enabled) override {} |
| |
| private: |
| class VoiceChannelAudioSink : public AudioSource::Sink { |
| public: |
| explicit VoiceChannelAudioSink(AudioSource* source); |
| ~VoiceChannelAudioSink() override; |
| void 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) override; |
| void OnClose() override; |
| int NumPreferredChannels() const override { return -1; } |
| AudioSource* source() const; |
| |
| private: |
| AudioSource* source_; |
| }; |
| |
| bool SetRecvCodecs(const std::vector<AudioCodec>& codecs); |
| bool SetMaxSendBandwidth(int bps); |
| bool SetOptions(const AudioOptions& options); |
| |
| std::vector<AudioCodec> recv_codecs_; |
| std::map<uint32_t, double> output_scalings_; |
| std::map<uint32_t, int> output_delays_; |
| std::vector<DtmfInfo> dtmf_info_queue_; |
| AudioOptions options_; |
| std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_; |
| std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
| int max_bps_; |
| }; |
| |
| class FakeVoiceMediaSendChannel |
| : public RtpSendChannelHelper<VoiceMediaSendChannelInterface> { |
| public: |
| struct DtmfInfo { |
| DtmfInfo(uint32_t ssrc, int event_code, int duration); |
| uint32_t ssrc; |
| int event_code; |
| int duration; |
| }; |
| FakeVoiceMediaSendChannel(const AudioOptions& options, |
| webrtc::TaskQueueBase* network_thread); |
| ~FakeVoiceMediaSendChannel() override; |
| |
| const std::vector<AudioCodec>& send_codecs() const; |
| const std::vector<DtmfInfo>& dtmf_info_queue() const; |
| const AudioOptions& options() const; |
| int max_bps() const; |
| bool HasSource(uint32_t ssrc) const; |
| bool GetOutputVolume(uint32_t ssrc, double* volume); |
| |
| // Overrides |
| VideoMediaSendChannelInterface* AsVideoSendChannel() override { |
| return nullptr; |
| } |
| VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { return this; } |
| cricket::MediaType media_type() const override { |
| return cricket::MEDIA_TYPE_AUDIO; |
| } |
| |
| bool SetSendParameters(const AudioSenderParameter& params) override; |
| void SetSend(bool send) override; |
| bool SetAudioSend(uint32_t ssrc, |
| bool enable, |
| const AudioOptions* options, |
| AudioSource* source) override; |
| |
| bool CanInsertDtmf() override; |
| bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override; |
| |
| bool SenderNackEnabled() const override { return false; } |
| bool SenderNonSenderRttEnabled() const override { return false; } |
| void SetReceiveNackEnabled(bool enabled) {} |
| void SetReceiveNonSenderRttEnabled(bool enabled) {} |
| bool SendCodecHasNack() const override { return false; } |
| void SetSendCodecChangedCallback( |
| absl::AnyInvocable<void()> callback) override {} |
| absl::optional<Codec> GetSendCodec() const override; |
| |
| bool GetStats(VoiceMediaSendInfo* stats) override; |
| |
| private: |
| class VoiceChannelAudioSink : public AudioSource::Sink { |
| public: |
| explicit VoiceChannelAudioSink(AudioSource* source); |
| ~VoiceChannelAudioSink() override; |
| void 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) override; |
| void OnClose() override; |
| int NumPreferredChannels() const override { return -1; } |
| AudioSource* source() const; |
| |
| private: |
| AudioSource* source_; |
| }; |
| |
| bool SetSendCodecs(const std::vector<AudioCodec>& codecs); |
| bool SetMaxSendBandwidth(int bps); |
| bool SetOptions(const AudioOptions& options); |
| bool SetLocalSource(uint32_t ssrc, AudioSource* source); |
| |
| std::vector<AudioCodec> send_codecs_; |
| std::map<uint32_t, double> output_scalings_; |
| std::map<uint32_t, int> output_delays_; |
| std::vector<DtmfInfo> dtmf_info_queue_; |
| AudioOptions options_; |
| std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_; |
| int max_bps_; |
| }; |
| |
| // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo. |
| bool CompareDtmfInfo(const FakeVoiceMediaSendChannel::DtmfInfo& info, |
| uint32_t ssrc, |
| int event_code, |
| int duration); |
| |
| class FakeVideoMediaReceiveChannel |
| : public RtpReceiveChannelHelper<VideoMediaReceiveChannelInterface> { |
| public: |
| FakeVideoMediaReceiveChannel(const VideoOptions& options, |
| webrtc::TaskQueueBase* network_thread); |
| |
| virtual ~FakeVideoMediaReceiveChannel(); |
| |
| VideoMediaReceiveChannelInterface* AsVideoReceiveChannel() override { |
| return this; |
| } |
| VoiceMediaReceiveChannelInterface* AsVoiceReceiveChannel() override { |
| return nullptr; |
| } |
| cricket::MediaType media_type() const override { |
| return cricket::MEDIA_TYPE_VIDEO; |
| } |
| |
| const std::vector<VideoCodec>& recv_codecs() const; |
| const std::vector<VideoCodec>& send_codecs() const; |
| bool rendering() const; |
| const VideoOptions& options() const; |
| const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>& |
| sinks() const; |
| int max_bps() const; |
| bool SetRecvParameters(const VideoReceiverParameters& params) override; |
| |
| bool SetSink(uint32_t ssrc, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override; |
| void SetDefaultSink( |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override; |
| bool HasSink(uint32_t ssrc) const; |
| |
| void SetReceive(bool receive) override {} |
| |
| bool HasSource(uint32_t ssrc) const; |
| bool AddRecvStream(const StreamParams& sp) override; |
| bool RemoveRecvStream(uint32_t ssrc) override; |
| |
| std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override; |
| |
| bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override; |
| absl::optional<int> GetBaseMinimumPlayoutDelayMs( |
| uint32_t ssrc) const override; |
| |
| void SetRecordableEncodedFrameCallback( |
| uint32_t ssrc, |
| std::function<void(const webrtc::RecordableEncodedFrame&)> callback) |
| override; |
| void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override; |
| void RequestRecvKeyFrame(uint32_t ssrc) override; |
| void SetReceiverFeedbackParameters(bool lntf_enabled, |
| bool nack_enabled, |
| webrtc::RtcpMode rtcp_mode, |
| absl::optional<int> rtx_time) override {} |
| bool GetStats(VideoMediaReceiveInfo* info) override; |
| |
| bool AddDefaultRecvStreamForTesting(const StreamParams& sp) override { |
| RTC_CHECK_NOTREACHED(); |
| return false; |
| } |
| |
| private: |
| bool SetRecvCodecs(const std::vector<VideoCodec>& codecs); |
| bool SetSendCodecs(const std::vector<VideoCodec>& codecs); |
| bool SetOptions(const VideoOptions& options); |
| bool SetMaxSendBandwidth(int bps); |
| |
| std::vector<VideoCodec> recv_codecs_; |
| std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*> sinks_; |
| std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_; |
| std::map<uint32_t, int> output_delays_; |
| VideoOptions options_; |
| int max_bps_; |
| }; |
| |
| class FakeVideoMediaSendChannel |
| : public RtpSendChannelHelper<VideoMediaSendChannelInterface> { |
| public: |
| FakeVideoMediaSendChannel(const VideoOptions& options, |
| webrtc::TaskQueueBase* network_thread); |
| |
| virtual ~FakeVideoMediaSendChannel(); |
| |
| VideoMediaSendChannelInterface* AsVideoSendChannel() override { return this; } |
| VoiceMediaSendChannelInterface* AsVoiceSendChannel() override { |
| return nullptr; |
| } |
| cricket::MediaType media_type() const override { |
| return cricket::MEDIA_TYPE_VIDEO; |
| } |
| |
| const std::vector<VideoCodec>& send_codecs() const; |
| const std::vector<VideoCodec>& codecs() const; |
| const VideoOptions& options() const; |
| const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>& |
| sinks() const; |
| int max_bps() const; |
| bool SetSendParameters(const VideoSenderParameters& params) override; |
| |
| absl::optional<Codec> GetSendCodec() const override; |
| |
| bool SetSend(bool send) override; |
| bool SetVideoSend( |
| uint32_t ssrc, |
| const VideoOptions* options, |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override; |
| |
| bool HasSource(uint32_t ssrc) const; |
| |
| void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override; |
| |
| void GenerateSendKeyFrame(uint32_t ssrc, |
| const std::vector<std::string>& rids) override; |
| webrtc::RtcpMode SendCodecRtcpMode() const override { |
| return webrtc::RtcpMode::kCompound; |
| } |
| void SetSendCodecChangedCallback( |
| absl::AnyInvocable<void()> callback) override {} |
| void SetSsrcListChangedCallback( |
| absl::AnyInvocable<void(const std::set<uint32_t>&)> callback) override {} |
| |
| void SetVideoCodecSwitchingEnabled(bool enabled) override {} |
| bool SendCodecHasLntf() const override { return false; } |
| bool SendCodecHasNack() const override { return false; } |
| absl::optional<int> SendCodecRtxTime() const override { |
| return absl::nullopt; |
| } |
| bool GetStats(VideoMediaSendInfo* info) override; |
| |
| private: |
| bool SetSendCodecs(const std::vector<VideoCodec>& codecs); |
| bool SetOptions(const VideoOptions& options); |
| bool SetMaxSendBandwidth(int bps); |
| |
| std::vector<VideoCodec> send_codecs_; |
| std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_; |
| VideoOptions options_; |
| int max_bps_; |
| }; |
| |
| class FakeVoiceEngine : public VoiceEngineInterface { |
| public: |
| FakeVoiceEngine(); |
| void Init() override; |
| rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override; |
| |
| std::unique_ptr<VoiceMediaSendChannelInterface> CreateSendChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const AudioOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::AudioCodecPairId codec_pair_id) override; |
| std::unique_ptr<VoiceMediaReceiveChannelInterface> CreateReceiveChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const AudioOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::AudioCodecPairId codec_pair_id) override; |
| |
| // TODO(ossu): For proper testing, These should either individually settable |
| // or the voice engine should reference mockable factories. |
| const std::vector<AudioCodec>& send_codecs() const override; |
| const std::vector<AudioCodec>& recv_codecs() const override; |
| void SetCodecs(const std::vector<AudioCodec>& codecs); |
| void SetRecvCodecs(const std::vector<AudioCodec>& codecs); |
| void SetSendCodecs(const std::vector<AudioCodec>& codecs); |
| int GetInputLevel(); |
| bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override; |
| void StopAecDump() override; |
| absl::optional<webrtc::AudioDeviceModule::Stats> GetAudioDeviceStats() |
| override; |
| std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions() |
| const override; |
| void SetRtpHeaderExtensions( |
| std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions); |
| |
| private: |
| std::vector<AudioCodec> recv_codecs_; |
| std::vector<AudioCodec> send_codecs_; |
| bool fail_create_channel_; |
| std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_; |
| |
| friend class FakeMediaEngine; |
| }; |
| |
| class FakeVideoEngine : public VideoEngineInterface { |
| public: |
| FakeVideoEngine(); |
| bool SetOptions(const VideoOptions& options); |
| std::unique_ptr<VideoMediaSendChannelInterface> CreateSendChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) |
| override; |
| std::unique_ptr<VideoMediaReceiveChannelInterface> CreateReceiveChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const VideoOptions& options, |
| const webrtc::CryptoOptions& crypto_options) override; |
| FakeVideoMediaSendChannel* GetSendChannel(size_t index); |
| FakeVideoMediaReceiveChannel* GetReceiveChannel(size_t index); |
| |
| std::vector<VideoCodec> send_codecs() const override { |
| return send_codecs(true); |
| } |
| std::vector<VideoCodec> recv_codecs() const override { |
| return recv_codecs(true); |
| } |
| std::vector<VideoCodec> send_codecs(bool include_rtx) const override; |
| std::vector<VideoCodec> recv_codecs(bool include_rtx) const override; |
| void SetSendCodecs(const std::vector<VideoCodec>& codecs); |
| void SetRecvCodecs(const std::vector<VideoCodec>& codecs); |
| bool SetCapture(bool capture); |
| std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions() |
| const override; |
| void SetRtpHeaderExtensions( |
| std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions); |
| |
| private: |
| std::vector<VideoCodec> send_codecs_; |
| std::vector<VideoCodec> recv_codecs_; |
| bool capture_; |
| VideoOptions options_; |
| bool fail_create_channel_; |
| std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_; |
| |
| friend class FakeMediaEngine; |
| }; |
| |
| class FakeMediaEngine : public CompositeMediaEngine { |
| public: |
| FakeMediaEngine(); |
| |
| ~FakeMediaEngine() override; |
| |
| void SetAudioCodecs(const std::vector<AudioCodec>& codecs); |
| void SetAudioRecvCodecs(const std::vector<AudioCodec>& codecs); |
| void SetAudioSendCodecs(const std::vector<AudioCodec>& codecs); |
| void SetVideoCodecs(const std::vector<VideoCodec>& codecs); |
| |
| void set_fail_create_channel(bool fail); |
| |
| FakeVoiceEngine* fake_voice_engine() { return voice_; } |
| FakeVideoEngine* fake_video_engine() { return video_; } |
| |
| private: |
| FakeVoiceEngine* const voice_; |
| FakeVideoEngine* const video_; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // MEDIA_BASE_FAKE_MEDIA_ENGINE_H_ |
