391 lines
14 KiB
C
391 lines
14 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 MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
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#define MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/video/video_frame.h"
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#include "call/audio_receive_stream.h"
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#include "call/audio_send_stream.h"
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#include "call/call.h"
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#include "call/flexfec_receive_stream.h"
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#include "call/test/mock_rtp_transport_controller_send.h"
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/buffer.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 payload_frequency = -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(int id,
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const webrtc::AudioSendStream::Config& config);
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int id() const { return id_; }
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const webrtc::AudioSendStream::Config& GetConfig() const override;
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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 Reconfigure(const webrtc::AudioSendStream::Config& config) override;
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void Start() override { sending_ = true; }
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void Stop() override { sending_ = false; }
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void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
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}
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bool SendTelephoneEvent(int payload_type,
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int payload_frequency,
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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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webrtc::AudioSendStream::Stats GetStats(
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bool has_remote_tracks) const override;
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int id_ = -1;
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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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int id,
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const webrtc::AudioReceiveStream::Config& config);
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int id() const { return id_; }
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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_; }
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float gain() const { return gain_; }
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bool DeliverRtp(const uint8_t* packet, size_t length, int64_t packet_time_us);
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bool started() const { return started_; }
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int base_mininum_playout_delay_ms() const {
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return base_mininum_playout_delay_ms_;
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}
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private:
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// webrtc::AudioReceiveStream implementation.
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void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override;
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void Start() override { started_ = true; }
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void Stop() override { started_ = false; }
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webrtc::AudioReceiveStream::Stats GetStats() const override;
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void SetSink(webrtc::AudioSinkInterface* sink) override;
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void SetGain(float gain) override;
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bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {
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base_mininum_playout_delay_ms_ = delay_ms;
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return true;
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}
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int GetBaseMinimumPlayoutDelayMs() const override {
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return base_mininum_playout_delay_ms_;
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}
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std::vector<webrtc::RtpSource> GetSources() const override {
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return std::vector<webrtc::RtpSource>();
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}
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int id_ = -1;
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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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webrtc::AudioSinkInterface* sink_ = nullptr;
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float gain_ = 1.0f;
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rtc::Buffer last_packet_;
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bool started_ = false;
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int base_mininum_playout_delay_ms_ = 0;
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};
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class FakeVideoSendStream final
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: public webrtc::VideoSendStream,
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public rtc::VideoSinkInterface<webrtc::VideoFrame> {
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public:
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FakeVideoSendStream(webrtc::VideoSendStream::Config config,
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webrtc::VideoEncoderConfig encoder_config);
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~FakeVideoSendStream() override;
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const webrtc::VideoSendStream::Config& GetConfig() const;
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const webrtc::VideoEncoderConfig& GetEncoderConfig() const;
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const std::vector<webrtc::VideoStream>& GetVideoStreams() const;
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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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bool GetH264Settings(webrtc::VideoCodecH264* 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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bool resolution_scaling_enabled() const {
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return resolution_scaling_enabled_;
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}
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bool framerate_scaling_enabled() const { return framerate_scaling_enabled_; }
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void InjectVideoSinkWants(const rtc::VideoSinkWants& wants);
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rtc::VideoSourceInterface<webrtc::VideoFrame>* source() const {
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return source_;
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}
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private:
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// rtc::VideoSinkInterface<VideoFrame> implementation.
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void OnFrame(const webrtc::VideoFrame& frame) override;
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// webrtc::VideoSendStream implementation.
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void UpdateActiveSimulcastLayers(
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const std::vector<bool> active_layers) override;
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void Start() override;
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void Stop() override;
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void AddAdaptationResource(
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rtc::scoped_refptr<webrtc::Resource> resource) override;
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std::vector<rtc::scoped_refptr<webrtc::Resource>> GetAdaptationResources()
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override;
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void SetSource(
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rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
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const webrtc::DegradationPreference& degradation_preference) override;
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webrtc::VideoSendStream::Stats GetStats() override;
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void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) 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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std::vector<webrtc::VideoStream> video_streams_;
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rtc::VideoSinkWants sink_wants_;
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bool codec_settings_set_;
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union CodecSpecificSettings {
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webrtc::VideoCodecVP8 vp8;
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webrtc::VideoCodecVP9 vp9;
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webrtc::VideoCodecH264 h264;
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} codec_specific_settings_;
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bool resolution_scaling_enabled_;
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bool framerate_scaling_enabled_;
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rtc::VideoSourceInterface<webrtc::VideoFrame>* source_;
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int num_swapped_frames_;
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absl::optional<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() const;
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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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void AddSecondarySink(webrtc::RtpPacketSinkInterface* sink) override;
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void RemoveSecondarySink(const webrtc::RtpPacketSinkInterface* sink) override;
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int GetNumAddedSecondarySinks() const;
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int GetNumRemovedSecondarySinks() const;
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std::vector<webrtc::RtpSource> GetSources() const override {
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return std::vector<webrtc::RtpSource>();
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}
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int base_mininum_playout_delay_ms() const {
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return base_mininum_playout_delay_ms_;
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}
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void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
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frame_decryptor) override {}
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void SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
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override {}
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RecordingState SetAndGetRecordingState(RecordingState state,
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bool generate_key_frame) override {
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return RecordingState();
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}
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void GenerateKeyFrame() override {}
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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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bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {
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base_mininum_playout_delay_ms_ = delay_ms;
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return true;
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}
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int GetBaseMinimumPlayoutDelayMs() const override {
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return base_mininum_playout_delay_ms_;
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}
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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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int base_mininum_playout_delay_ms_ = 0;
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int num_added_secondary_sinks_;
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int num_removed_secondary_sinks_;
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};
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class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
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public:
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explicit FakeFlexfecReceiveStream(
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const webrtc::FlexfecReceiveStream::Config& config);
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const webrtc::FlexfecReceiveStream::Config& GetConfig() const override;
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private:
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webrtc::FlexfecReceiveStream::Stats GetStats() const override;
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void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
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webrtc::FlexfecReceiveStream::Config config_;
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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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FakeCall();
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~FakeCall() override;
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webrtc::MockRtpTransportControllerSend* GetMockTransportControllerSend() {
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return &transport_controller_send_;
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}
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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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const FakeVideoReceiveStream* GetVideoReceiveStream(uint32_t ssrc);
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const std::vector<FakeFlexfecReceiveStream*>& GetFlexfecReceiveStreams();
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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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void SetClientBitratePreferences(
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const webrtc::BitrateSettings& preferences) override {}
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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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webrtc::VideoSendStream::Config config,
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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::FlexfecReceiveStream* CreateFlexfecReceiveStream(
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const webrtc::FlexfecReceiveStream::Config& config) override;
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void DestroyFlexfecReceiveStream(
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webrtc::FlexfecReceiveStream* receive_stream) override;
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void AddAdaptationResource(
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rtc::scoped_refptr<webrtc::Resource> resource) override;
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webrtc::PacketReceiver* Receiver() override;
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DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us) override;
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webrtc::RtpTransportControllerSendInterface* GetTransportControllerSend()
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override {
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return &transport_controller_send_;
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}
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webrtc::Call::Stats GetStats() const override;
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||
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void SignalChannelNetworkState(webrtc::MediaType media,
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webrtc::NetworkState state) override;
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void OnAudioTransportOverheadChanged(
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int transport_overhead_per_packet) override;
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||
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void OnSentPacket(const rtc::SentPacket& sent_packet) override;
|
||
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|
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||
|
|
::testing::NiceMock<webrtc::MockRtpTransportControllerSend>
|
||
|
|
transport_controller_send_;
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||
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|
|
||
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|
webrtc::NetworkState audio_network_state_;
|
||
|
|
webrtc::NetworkState video_network_state_;
|
||
|
|
rtc::SentPacket last_sent_packet_;
|
||
|
|
int last_sent_nonnegative_packet_id_ = -1;
|
||
|
|
int next_stream_id_ = 665;
|
||
|
|
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_;
|
||
|
|
std::vector<FakeFlexfecReceiveStream*> flexfec_receive_streams_;
|
||
|
|
|
||
|
|
int num_created_send_streams_;
|
||
|
|
int num_created_receive_streams_;
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace cricket
|
||
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|
#endif // MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
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