| /* |
| * Copyright (c) 2015 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. |
| */ |
| |
| // This file contains fake implementations, for use in unit tests, of the |
| // following classes: |
| // |
| // webrtc::Call |
| // webrtc::AudioSendStream |
| // webrtc::AudioReceiveStream |
| // webrtc::VideoSendStream |
| // webrtc::VideoReceiveStream |
| |
| #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ |
| #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ |
| |
| #include <memory> |
| #include <vector> |
| |
| #include "webrtc/api/call/audio_receive_stream.h" |
| #include "webrtc/api/call/audio_send_stream.h" |
| #include "webrtc/base/buffer.h" |
| #include "webrtc/call.h" |
| #include "webrtc/video_frame.h" |
| #include "webrtc/video_receive_stream.h" |
| #include "webrtc/video_send_stream.h" |
| |
| namespace cricket { |
| class FakeAudioSendStream final : public webrtc::AudioSendStream { |
| public: |
| struct TelephoneEvent { |
| int payload_type = -1; |
| int event_code = 0; |
| int duration_ms = 0; |
| }; |
| |
| explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config); |
| |
| const webrtc::AudioSendStream::Config& GetConfig() const; |
| void SetStats(const webrtc::AudioSendStream::Stats& stats); |
| TelephoneEvent GetLatestTelephoneEvent() const; |
| bool IsSending() const { return sending_; } |
| bool muted() const { return muted_; } |
| |
| private: |
| // webrtc::AudioSendStream implementation. |
| void Start() override { sending_ = true; } |
| void Stop() override { sending_ = false; } |
| |
| bool SendTelephoneEvent(int payload_type, int event, |
| int duration_ms) override; |
| void SetMuted(bool muted) override; |
| webrtc::AudioSendStream::Stats GetStats() const override; |
| |
| TelephoneEvent latest_telephone_event_; |
| webrtc::AudioSendStream::Config config_; |
| webrtc::AudioSendStream::Stats stats_; |
| bool sending_ = false; |
| bool muted_ = false; |
| }; |
| |
| class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |
| public: |
| explicit FakeAudioReceiveStream( |
| const webrtc::AudioReceiveStream::Config& config); |
| |
| const webrtc::AudioReceiveStream::Config& GetConfig() const; |
| void SetStats(const webrtc::AudioReceiveStream::Stats& stats); |
| int received_packets() const { return received_packets_; } |
| bool VerifyLastPacket(const uint8_t* data, size_t length) const; |
| const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } |
| float gain() const { return gain_; } |
| bool DeliverRtp(const uint8_t* packet, |
| size_t length, |
| const webrtc::PacketTime& packet_time); |
| bool started() const { return started_; } |
| |
| private: |
| // webrtc::AudioReceiveStream implementation. |
| void Start() override { started_ = true; } |
| void Stop() override { started_ = false; } |
| |
| webrtc::AudioReceiveStream::Stats GetStats() const override; |
| void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
| void SetGain(float gain) override; |
| |
| webrtc::AudioReceiveStream::Config config_; |
| webrtc::AudioReceiveStream::Stats stats_; |
| int received_packets_ = 0; |
| std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
| float gain_ = 1.0f; |
| rtc::Buffer last_packet_; |
| bool started_ = false; |
| }; |
| |
| class FakeVideoSendStream final |
| : public webrtc::VideoSendStream, |
| public rtc::VideoSinkInterface<webrtc::VideoFrame> { |
| public: |
| FakeVideoSendStream(webrtc::VideoSendStream::Config config, |
| webrtc::VideoEncoderConfig encoder_config); |
| ~FakeVideoSendStream() override; |
| const webrtc::VideoSendStream::Config& GetConfig() const; |
| const webrtc::VideoEncoderConfig& GetEncoderConfig() const; |
| std::vector<webrtc::VideoStream> GetVideoStreams(); |
| |
| bool IsSending() const; |
| bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const; |
| bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const; |
| |
| int GetNumberOfSwappedFrames() const; |
| int GetLastWidth() const; |
| int GetLastHeight() const; |
| int64_t GetLastTimestamp() const; |
| void SetStats(const webrtc::VideoSendStream::Stats& stats); |
| int num_encoder_reconfigurations() const { |
| return num_encoder_reconfigurations_; |
| } |
| |
| private: |
| // rtc::VideoSinkInterface<VideoFrame> implementation. |
| void OnFrame(const webrtc::VideoFrame& frame) override; |
| |
| // webrtc::VideoSendStream implementation. |
| void Start() override; |
| void Stop() override; |
| void SetSource( |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override; |
| webrtc::VideoSendStream::Stats GetStats() override; |
| void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override; |
| |
| bool sending_; |
| webrtc::VideoSendStream::Config config_; |
| webrtc::VideoEncoderConfig encoder_config_; |
| bool codec_settings_set_; |
| union VpxSettings { |
| webrtc::VideoCodecVP8 vp8; |
| webrtc::VideoCodecVP9 vp9; |
| } vpx_settings_; |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source_; |
| int num_swapped_frames_; |
| webrtc::VideoFrame last_frame_; |
| webrtc::VideoSendStream::Stats stats_; |
| int num_encoder_reconfigurations_ = 0; |
| }; |
| |
| class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream { |
| public: |
| explicit FakeVideoReceiveStream(webrtc::VideoReceiveStream::Config config); |
| |
| const webrtc::VideoReceiveStream::Config& GetConfig(); |
| |
| bool IsReceiving() const; |
| |
| void InjectFrame(const webrtc::VideoFrame& frame); |
| |
| void SetStats(const webrtc::VideoReceiveStream::Stats& stats); |
| |
| private: |
| // webrtc::VideoReceiveStream implementation. |
| void Start() override; |
| void Stop() override; |
| |
| webrtc::VideoReceiveStream::Stats GetStats() const override; |
| |
| webrtc::VideoReceiveStream::Config config_; |
| bool receiving_; |
| webrtc::VideoReceiveStream::Stats stats_; |
| }; |
| |
| class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { |
| public: |
| explicit FakeCall(const webrtc::Call::Config& config); |
| ~FakeCall() override; |
| |
| webrtc::Call::Config GetConfig() const; |
| const std::vector<FakeVideoSendStream*>& GetVideoSendStreams(); |
| const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams(); |
| |
| const std::vector<FakeAudioSendStream*>& GetAudioSendStreams(); |
| const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc); |
| const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams(); |
| const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc); |
| |
| rtc::SentPacket last_sent_packet() const { return last_sent_packet_; } |
| |
| // This is useful if we care about the last media packet (with id populated) |
| // but not the last ICE packet (with -1 ID). |
| int last_sent_nonnegative_packet_id() const { |
| return last_sent_nonnegative_packet_id_; |
| } |
| |
| webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; |
| int GetNumCreatedSendStreams() const; |
| int GetNumCreatedReceiveStreams() const; |
| void SetStats(const webrtc::Call::Stats& stats); |
| |
| private: |
| webrtc::AudioSendStream* CreateAudioSendStream( |
| const webrtc::AudioSendStream::Config& config) override; |
| void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; |
| |
| webrtc::AudioReceiveStream* CreateAudioReceiveStream( |
| const webrtc::AudioReceiveStream::Config& config) override; |
| void DestroyAudioReceiveStream( |
| webrtc::AudioReceiveStream* receive_stream) override; |
| |
| webrtc::VideoSendStream* CreateVideoSendStream( |
| webrtc::VideoSendStream::Config config, |
| webrtc::VideoEncoderConfig encoder_config) override; |
| void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; |
| |
| webrtc::VideoReceiveStream* CreateVideoReceiveStream( |
| webrtc::VideoReceiveStream::Config config) override; |
| void DestroyVideoReceiveStream( |
| webrtc::VideoReceiveStream* receive_stream) override; |
| webrtc::PacketReceiver* Receiver() override; |
| |
| DeliveryStatus DeliverPacket(webrtc::MediaType media_type, |
| const uint8_t* packet, |
| size_t length, |
| const webrtc::PacketTime& packet_time) override; |
| |
| webrtc::Call::Stats GetStats() const override; |
| |
| void SetBitrateConfig( |
| const webrtc::Call::Config::BitrateConfig& bitrate_config) override; |
| void OnNetworkRouteChanged(const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) override {} |
| void SignalChannelNetworkState(webrtc::MediaType media, |
| webrtc::NetworkState state) override; |
| void OnSentPacket(const rtc::SentPacket& sent_packet) override; |
| |
| bool StartEventLog(rtc::PlatformFile log_file, |
| int64_t max_size_bytes) override; |
| void StopEventLog() override; |
| |
| webrtc::Call::Config config_; |
| webrtc::NetworkState audio_network_state_; |
| webrtc::NetworkState video_network_state_; |
| rtc::SentPacket last_sent_packet_; |
| int last_sent_nonnegative_packet_id_ = -1; |
| webrtc::Call::Stats stats_; |
| std::vector<FakeVideoSendStream*> video_send_streams_; |
| std::vector<FakeAudioSendStream*> audio_send_streams_; |
| std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
| std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
| |
| int num_created_send_streams_; |
| int num_created_receive_streams_; |
| }; |
| |
| } // namespace cricket |
| #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |