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