blob: 91585f69a95c9bb3d3392b34f4b3a62af01c904a [file] [log] [blame]
/*
* 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 <string>
#include <vector>
#include "webrtc/api/video/video_frame.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
#include "webrtc/call/call.h"
#include "webrtc/call/flexfec_receive_stream.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/rtc_base/buffer.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 payload_frequency = -1;
int event_code = 0;
int duration_ms = 0;
};
explicit FakeAudioSendStream(
int id, const webrtc::AudioSendStream::Config& config);
int id() const { return id_; }
const webrtc::AudioSendStream::Config& GetConfig() const override;
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 Reconfigure(const webrtc::AudioSendStream::Config& config) override;
void Start() override { sending_ = true; }
void Stop() override { sending_ = false; }
bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
int duration_ms) override;
void SetMuted(bool muted) override;
webrtc::AudioSendStream::Stats GetStats() const override;
int id_ = -1;
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(
int id, const webrtc::AudioReceiveStream::Config& config);
int id() const { return id_; }
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;
int GetOutputLevel() const override { return 0; }
void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
void SetGain(float gain) override;
std::vector<webrtc::RtpSource> GetSources() const override {
return std::vector<webrtc::RtpSource>();
}
int id_ = -1;
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;
const std::vector<webrtc::VideoStream>& GetVideoStreams() const;
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_;
}
void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
size_t byte_limit) override;
bool resolution_scaling_enabled() const {
return resolution_scaling_enabled_;
}
bool framerate_scaling_enabled() const { return framerate_scaling_enabled_; }
void InjectVideoSinkWants(const rtc::VideoSinkWants& wants);
rtc::VideoSourceInterface<webrtc::VideoFrame>* source() const {
return source_;
}
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,
const webrtc::VideoSendStream::DegradationPreference&
degradation_preference) override;
webrtc::VideoSendStream::Stats GetStats() override;
void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override;
bool sending_;
webrtc::VideoSendStream::Config config_;
webrtc::VideoEncoderConfig encoder_config_;
std::vector<webrtc::VideoStream> video_streams_;
rtc::VideoSinkWants sink_wants_;
bool codec_settings_set_;
union VpxSettings {
webrtc::VideoCodecVP8 vp8;
webrtc::VideoCodecVP9 vp9;
} vpx_settings_;
bool resolution_scaling_enabled_;
bool framerate_scaling_enabled_;
rtc::VideoSourceInterface<webrtc::VideoFrame>* source_;
int num_swapped_frames_;
rtc::Optional<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() const;
bool IsReceiving() const;
void InjectFrame(const webrtc::VideoFrame& frame);
void SetStats(const webrtc::VideoReceiveStream::Stats& stats);
void EnableEncodedFrameRecording(rtc::PlatformFile file,
size_t byte_limit) override;
void AddSecondarySink(webrtc::RtpPacketSinkInterface* sink) override;
void RemoveSecondarySink(const webrtc::RtpPacketSinkInterface* sink) override;
private:
// webrtc::VideoReceiveStream implementation.
void Start() override;
void Stop() override;
webrtc::VideoReceiveStream::Stats GetStats() const override;
rtc::Optional<webrtc::TimingFrameInfo> GetAndResetTimingFrameInfo() override;
webrtc::VideoReceiveStream::Config config_;
bool receiving_;
webrtc::VideoReceiveStream::Stats stats_;
};
class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
public:
explicit FakeFlexfecReceiveStream(
const webrtc::FlexfecReceiveStream::Config& config);
const webrtc::FlexfecReceiveStream::Config& GetConfig() const override;
private:
webrtc::FlexfecReceiveStream::Stats GetStats() const override;
void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
webrtc::FlexfecReceiveStream::Config config_;
};
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);
const std::vector<FakeFlexfecReceiveStream*>& GetFlexfecReceiveStreams();
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::FlexfecReceiveStream* CreateFlexfecReceiveStream(
const webrtc::FlexfecReceiveStream::Config& config) override;
void DestroyFlexfecReceiveStream(
webrtc::FlexfecReceiveStream* 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 SetBitrateConfigMask(
const webrtc::Call::Config::BitrateConfigMask& mask) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override {}
void SignalChannelNetworkState(webrtc::MediaType media,
webrtc::NetworkState state) override;
void OnTransportOverheadChanged(webrtc::MediaType media,
int transport_overhead_per_packet) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) 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;
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_;
int audio_transport_overhead_;
int video_transport_overhead_;
};
} // namespace cricket
#endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_