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Diffstat (limited to 'talk/media/webrtc/fakewebrtccall.h')
| -rw-r--r-- | talk/media/webrtc/fakewebrtccall.h | 258 |
1 files changed, 258 insertions, 0 deletions
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h new file mode 100644 index 0000000000..88edc60d78 --- /dev/null +++ b/talk/media/webrtc/fakewebrtccall.h @@ -0,0 +1,258 @@ +/* + * libjingle + * Copyright 2015 Google Inc. + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * 1. Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * 3. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED + * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF + * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO + * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; + * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, + * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR + * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF + * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + */ + +// 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 TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_ +#define TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_ + +#include <vector> + +#include "webrtc/call.h" +#include "webrtc/audio_receive_stream.h" +#include "webrtc/audio_send_stream.h" +#include "webrtc/video_frame.h" +#include "webrtc/video_receive_stream.h" +#include "webrtc/video_send_stream.h" + +namespace cricket { + +class FakeAudioSendStream : public webrtc::AudioSendStream { + public: + explicit FakeAudioSendStream( + const webrtc::AudioSendStream::Config& config); + + const webrtc::AudioSendStream::Config& GetConfig() const; + void SetStats(const webrtc::AudioSendStream::Stats& stats); + + private: + // webrtc::SendStream implementation. + void Start() override {} + void Stop() override {} + void SignalNetworkState(webrtc::NetworkState state) override {} + bool DeliverRtcp(const uint8_t* packet, size_t length) override { + return true; + } + + // webrtc::AudioSendStream implementation. + webrtc::AudioSendStream::Stats GetStats() const override; + + webrtc::AudioSendStream::Config config_; + webrtc::AudioSendStream::Stats stats_; +}; + +class FakeAudioReceiveStream : 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_; } + void IncrementReceivedPackets(); + + private: + // webrtc::ReceiveStream implementation. + void Start() override {} + void Stop() override {} + void SignalNetworkState(webrtc::NetworkState state) override {} + bool DeliverRtcp(const uint8_t* packet, size_t length) override { + return true; + } + bool DeliverRtp(const uint8_t* packet, + size_t length, + const webrtc::PacketTime& packet_time) override { + return true; + } + + // webrtc::AudioReceiveStream implementation. + webrtc::AudioReceiveStream::Stats GetStats() const override; + + webrtc::AudioReceiveStream::Config config_; + webrtc::AudioReceiveStream::Stats stats_; + int received_packets_; +}; + +class FakeVideoSendStream : public webrtc::VideoSendStream, + public webrtc::VideoCaptureInput { + public: + FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, + const webrtc::VideoEncoderConfig& encoder_config); + webrtc::VideoSendStream::Config GetConfig() 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); + + private: + void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override; + + // webrtc::SendStream implementation. + void Start() override; + void Stop() override; + void SignalNetworkState(webrtc::NetworkState state) override {} + bool DeliverRtcp(const uint8_t* packet, size_t length) override { + return true; + } + + // webrtc::VideoSendStream implementation. + webrtc::VideoSendStream::Stats GetStats() override; + bool ReconfigureVideoEncoder( + const webrtc::VideoEncoderConfig& config) override; + webrtc::VideoCaptureInput* Input() 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_; + int num_swapped_frames_; + webrtc::VideoFrame last_frame_; + webrtc::VideoSendStream::Stats stats_; +}; + +class FakeVideoReceiveStream : public webrtc::VideoReceiveStream { + public: + explicit FakeVideoReceiveStream( + const webrtc::VideoReceiveStream::Config& config); + + webrtc::VideoReceiveStream::Config GetConfig(); + + bool IsReceiving() const; + + void InjectFrame(const webrtc::VideoFrame& frame, int time_to_render_ms); + + void SetStats(const webrtc::VideoReceiveStream::Stats& stats); + + private: + // webrtc::ReceiveStream implementation. + void Start() override; + void Stop() override; + void SignalNetworkState(webrtc::NetworkState state) override {} + bool DeliverRtcp(const uint8_t* packet, size_t length) override { + return true; + } + bool DeliverRtp(const uint8_t* packet, + size_t length, + const webrtc::PacketTime& packet_time) override { + return true; + } + + // webrtc::VideoReceiveStream implementation. + webrtc::VideoReceiveStream::Stats GetStats() const override; + + webrtc::VideoReceiveStream::Config config_; + bool receiving_; + webrtc::VideoReceiveStream::Stats stats_; +}; + +class FakeCall : 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_; } + webrtc::NetworkState GetNetworkState() 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( + const webrtc::VideoSendStream::Config& config, + const webrtc::VideoEncoderConfig& encoder_config) override; + void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; + + webrtc::VideoReceiveStream* CreateVideoReceiveStream( + const 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 SignalNetworkState(webrtc::NetworkState state) override; + void OnSentPacket(const rtc::SentPacket& sent_packet) override; + + webrtc::Call::Config config_; + webrtc::NetworkState network_state_; + rtc::SentPacket last_sent_packet_; + 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_ |