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/*
* 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 MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
#define MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_frame.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
#include "call/flexfec_receive_stream.h"
#include "call/test/mock_rtp_transport_controller_send.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/buffer.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; }
void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
}
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;
webrtc::AudioSendStream::Stats GetStats(
bool has_remote_tracks) 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_; }
float gain() const { return gain_; }
bool DeliverRtp(const uint8_t* packet, size_t length, int64_t packet_time_us);
bool started() const { return started_; }
int base_mininum_playout_delay_ms() const {
return base_mininum_playout_delay_ms_;
}
void SetLocalSsrc(uint32_t local_ssrc) {
config_.rtp.local_ssrc = local_ssrc;
}
void SetSyncGroup(const std::string& sync_group) {
config_.sync_group = sync_group;
}
private:
const webrtc::ReceiveStream::RtpConfig& rtp_config() const override {
return config_.rtp;
}
void Start() override { started_ = true; }
void Stop() override { started_ = false; }
bool IsRunning() const override { return started_; }
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
void SetDecoderMap(
std::map<int, webrtc::SdpAudioFormat> decoder_map) override;
void SetUseTransportCcAndNackHistory(bool use_transport_cc,
int history_ms) override;
void SetNonSenderRttMeasurement(bool enabled) override;
void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override;
void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
webrtc::AudioReceiveStream::Stats GetStats(
bool get_and_clear_legacy_stats) const override;
void SetSink(webrtc::AudioSinkInterface* sink) override;
void SetGain(float gain) override;
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {
base_mininum_playout_delay_ms_ = delay_ms;
return true;
}
int GetBaseMinimumPlayoutDelayMs() const override {
return base_mininum_playout_delay_ms_;
}
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;
webrtc::AudioSinkInterface* sink_ = nullptr;
float gain_ = 1.0f;
rtc::Buffer last_packet_;
bool started_ = false;
int base_mininum_playout_delay_ms_ = 0;
};
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;
bool GetH264Settings(webrtc::VideoCodecH264* 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_;
}
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 UpdateActiveSimulcastLayers(
const std::vector<bool> active_layers) override;
void Start() override;
void Stop() override;
bool started() override { return IsSending(); }
void AddAdaptationResource(
rtc::scoped_refptr<webrtc::Resource> resource) override;
std::vector<rtc::scoped_refptr<webrtc::Resource>> GetAdaptationResources()
override;
void SetSource(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const webrtc::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 CodecSpecificSettings {
webrtc::VideoCodecVP8 vp8;
webrtc::VideoCodecVP9 vp9;
webrtc::VideoCodecH264 h264;
} codec_specific_settings_;
bool resolution_scaling_enabled_;
bool framerate_scaling_enabled_;
rtc::VideoSourceInterface<webrtc::VideoFrame>* source_;
int num_swapped_frames_;
absl::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);
std::vector<webrtc::RtpSource> GetSources() const override {
return std::vector<webrtc::RtpSource>();
}
int base_mininum_playout_delay_ms() const {
return base_mininum_playout_delay_ms_;
}
void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override {}
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override {}
RecordingState SetAndGetRecordingState(RecordingState state,
bool generate_key_frame) override {
return RecordingState();
}
void GenerateKeyFrame() override {}
private:
// webrtc::VideoReceiveStream implementation.
void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
const webrtc::ReceiveStream::RtpConfig& rtp_config() const override {
return config_.rtp;
}
void Start() override;
void Stop() override;
webrtc::VideoReceiveStream::Stats GetStats() const override;
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override {
base_mininum_playout_delay_ms_ = delay_ms;
return true;
}
int GetBaseMinimumPlayoutDelayMs() const override {
return base_mininum_playout_delay_ms_;
}
webrtc::VideoReceiveStream::Config config_;
bool receiving_;
webrtc::VideoReceiveStream::Stats stats_;
int base_mininum_playout_delay_ms_ = 0;
};
class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
public:
explicit FakeFlexfecReceiveStream(
const webrtc::FlexfecReceiveStream::Config& config);
void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
const webrtc::ReceiveStream::RtpConfig& rtp_config() const override {
return config_.rtp;
}
const webrtc::FlexfecReceiveStream::Config& GetConfig() const;
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:
FakeCall();
FakeCall(webrtc::TaskQueueBase* worker_thread,
webrtc::TaskQueueBase* network_thread);
~FakeCall() override;
webrtc::MockRtpTransportControllerSend* GetMockTransportControllerSend() {
return &transport_controller_send_;
}
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 FakeVideoReceiveStream* GetVideoReceiveStream(uint32_t ssrc);
const std::vector<FakeFlexfecReceiveStream*>& GetFlexfecReceiveStreams();
rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
size_t GetDeliveredPacketsForSsrc(uint32_t ssrc) const {
auto it = delivered_packets_by_ssrc_.find(ssrc);
return it != delivered_packets_by_ssrc_.end() ? it->second : 0u;
}
// 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);
void SetClientBitratePreferences(
const webrtc::BitrateSettings& preferences) override {}
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;
void AddAdaptationResource(
rtc::scoped_refptr<webrtc::Resource> resource) override;
webrtc::PacketReceiver* Receiver() override;
DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
webrtc::RtpTransportControllerSendInterface* GetTransportControllerSend()
override {
return &transport_controller_send_;
}
webrtc::Call::Stats GetStats() const override;
const webrtc::WebRtcKeyValueConfig& trials() const override {
return trials_;
}
webrtc::TaskQueueBase* network_thread() const override;
webrtc::TaskQueueBase* worker_thread() const override;
void SignalChannelNetworkState(webrtc::MediaType media,
webrtc::NetworkState state) override;
void OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) override;
void OnLocalSsrcUpdated(webrtc::AudioReceiveStream& stream,
uint32_t local_ssrc) override;
void OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream,
const std::string& sync_group) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
webrtc::TaskQueueBase* const network_thread_;
webrtc::TaskQueueBase* const worker_thread_;
::testing::NiceMock<webrtc::MockRtpTransportControllerSend>
transport_controller_send_;
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_;
std::map<uint32_t, size_t> delivered_packets_by_ssrc_;
int num_created_send_streams_;
int num_created_receive_streams_;
webrtc::FieldTrialBasedConfig trials_;
};
} // namespace cricket
#endif // MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_