/*
* 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::AudioReceiveStreamInterface
// webrtc::VideoSendStream
// webrtc::VideoReceiveStreamInterface
#ifndef MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
#define MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
#include <cstddef>
#include <cstdint>
#include <map>
#include <memory>
#include <optional>
#include <span>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/adaptation/resource.h"
#include "api/audio/audio_frame.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_format.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/environment/environment.h"
#include "api/fec_controller.h"
#include "api/frame_transformer_interface.h"
#include "api/media_types.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video/video_stream_encoder_settings.h"
#include "api/video_codecs/video_codec.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
#include "call/fake_payload_type_suggester.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
#include "call/payload_type.h"
#include "call/rtp_transport_controller_send_interface.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/include/receive_statistics.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/buffer.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "test/gmock.h"
#include "video/config/video_encoder_config.h"
namespace webrtc {
class FakeAudioSendStream final : public 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 AudioSendStream::Config& config);
int id() const { return id_; }
const AudioSendStream::Config& GetConfig() const override;
void SetStats(const AudioSendStream::Stats& stats);
TelephoneEvent GetLatestTelephoneEvent() const ;
bool IsSending() const { return sending_; }
bool muted() const { return muted_; }
private :
// AudioSendStream implementation.
void Reconfigure(const AudioSendStream::Config& config,
SetParametersCallback callback) override;
void Start() override { sending_ = true ; }
void Stop() override { sending_ = false; }
void SendAudioData(std::unique_ptr<AudioFrame> /* audio_frame */) override {}
bool SendTelephoneEvent(int payload_type,
int payload_frequency,
int event,
int duration_ms) override;
void SetMuted(bool muted) override;
AudioSendStream::Stats GetStats() const override;
AudioSendStream::Stats GetStats(bool has_remote_tracks) const override;
int id_ = -1 ;
TelephoneEvent latest_telephone_event_;
AudioSendStream::Config config_;
AudioSendStream::Stats stats_;
bool sending_ = false;
bool muted_ = false;
};
class FakeAudioReceiveStream final : public AudioReceiveStreamInterface {
public :
explicit FakeAudioReceiveStream(
int id,
const AudioReceiveStreamInterface::Config& config);
int id() const { return id_; }
const AudioReceiveStreamInterface::Config& GetConfig() const ;
void SetStats(const AudioReceiveStreamInterface::Stats& stats);
int received_packets() const { return received_packets_; }
bool VerifyLastPacket(std::span<const uint8_t> data) const ;
const AudioSinkInterface* sink() const { return sink_; }
float gain() const { return gain_; }
bool DeliverRtp(std::span<const uint8_t> packet, int64_t packet_time_us);
bool started() const { return started_; }
int base_mininum_playout_delay_ms() const {
return base_mininum_playout_delay_ms_;
}
void SetSyncGroup(absl::string_view sync_group) {
config_.sync_group = std::string(sync_group);
}
uint32_t remote_ssrc() const override { return config_.rtp.remote_ssrc; }
void Start() override { started_ = true ; }
void Stop() override { started_ = false; }
bool IsRunning() const override { return started_; }
void SetDepacketizerToDecoderFrameTransformer(
scoped_refptr<FrameTransformerInterface> frame_transformer) override;
void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
void SetNackHistory(int history_ms) override;
void SetRtcpMode(RtcpMode mode) override;
void SetNonSenderRttMeasurement(bool enabled) override;
void SetFrameDecryptor(
scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
AudioReceiveStreamInterface::Stats GetStats(
bool get_and_clear_legacy_stats) const override;
void SetSink(AudioSinkInterface* sink) override;
void SetGain(float gain) override;
void SetJitterBufferMaxPackets(size_t max_packets) override {}
void SetJitterBufferFastAccelerate(bool fast_accelerate) 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<RtpSource> GetSources() const override {
return std::vector<RtpSource>();
}
AudioMixer::Source* source() override {
// TODO(b/397376626): Add a Fake AudioMixer::Source
return nullptr;
}
private :
int id_ = -1 ;
AudioReceiveStreamInterface::Config config_;
AudioReceiveStreamInterface::Stats stats_;
int received_packets_ = 0 ;
AudioSinkInterface* sink_ = nullptr;
float gain_ = 1 .0 f;
Buffer last_packet_;
bool started_ = false;
int base_mininum_playout_delay_ms_ = 0 ;
};
class FakeVideoSendStream final : public VideoSendStream,
public VideoSinkInterface<VideoFrame> {
public :
FakeVideoSendStream(const Environment& env,
VideoSendStream::Config config,
VideoEncoderConfig encoder_config);
~FakeVideoSendStream() override;
const VideoSendStream::Config& GetConfig() const ;
const VideoEncoderConfig& GetEncoderConfig() const ;
const std::vector<VideoStream>& GetVideoStreams() const ;
bool IsSending() const ;
bool GetVp8Settings(VideoCodecVP8* settings) const ;
bool GetVp9Settings(VideoCodecVP9* settings) const ;
bool GetH264Settings(VideoCodecH264* settings) const ;
bool GetAv1Settings(VideoCodecAV1* settings) const ;
int GetNumberOfSwappedFrames() const ;
int GetLastWidth() const ;
int GetLastHeight() const ;
int64_t GetLastTimestamp() const ;
void SetStats(const VideoSendStream::Stats& stats) override;
void SetCsrcs(std::span<const uint32_t> csrcs) override;
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 VideoSinkWants& wants);
VideoSourceInterface<VideoFrame>* source() const { return source_; }
void GenerateKeyFrame(const std::vector<std::string>& rids) override;
const std::vector<std::string>& GetKeyFramesRequested() const {
return keyframes_requested_by_rid_;
}
private :
// VideoSinkInterface<VideoFrame> implementation.
void OnFrame(const VideoFrame& frame) override;
// VideoSendStream implementation.
void Start() override;
void Stop() override;
bool started() override { return IsSending(); }
void AddAdaptationResource(scoped_refptr<Resource> resource) override;
std::vector<scoped_refptr<Resource>> GetAdaptationResources() override;
void SetSource(VideoSourceInterface<VideoFrame>* source,
const DegradationPreference& degradation_preference) override;
VideoSendStream::Stats GetStats() override;
void ReconfigureVideoEncoder(VideoEncoderConfig config) override;
void ReconfigureVideoEncoder(VideoEncoderConfig config,
SetParametersCallback callback) override;
const Environment env_;
bool sending_;
VideoSendStream::Config config_;
VideoEncoderConfig encoder_config_;
std::vector<VideoStream> video_streams_;
VideoSinkWants sink_wants_;
bool codec_settings_set_;
union CodecSpecificSettings {
VideoCodecVP8 vp8;
VideoCodecVP9 vp9;
VideoCodecH264 h264;
VideoCodecAV1 av1;
} codec_specific_settings_;
bool resolution_scaling_enabled_;
bool framerate_scaling_enabled_;
VideoSourceInterface<VideoFrame>* source_;
int num_swapped_frames_;
std::optional<VideoFrame> last_frame_;
VideoSendStream::Stats stats_;
int num_encoder_reconfigurations_ = 0 ;
std::vector<std::string> keyframes_requested_by_rid_;
};
class FakeVideoReceiveStream final : public VideoReceiveStreamInterface {
public :
explicit FakeVideoReceiveStream(VideoReceiveStreamInterface::Config config);
const VideoReceiveStreamInterface::Config& GetConfig() const ;
bool IsReceiving() const ;
void InjectFrame(const VideoFrame& frame);
void SetStats(const VideoReceiveStreamInterface::Stats& stats);
std::vector<RtpSource> GetSources() const override {
return std::vector<RtpSource>();
}
int base_mininum_playout_delay_ms() const {
return base_mininum_playout_delay_ms_;
}
void UpdateRtxSsrc(uint32_t ssrc) override { config_.rtp.rtx_ssrc = ssrc; }
void SetFrameDecryptor(scoped_refptr<FrameDecryptorInterface>
/* frame_decryptor */) override {}
void SetDepacketizerToDecoderFrameTransformer(
scoped_refptr<FrameTransformerInterface> /* frame_transformer */)
override {}
RecordingState SetAndGetRecordingState(
RecordingState /* state */,
bool /* generate_key_frame */) override {
return RecordingState();
}
void GenerateKeyFrame() override {}
void SetRtcpMode(RtcpMode mode) override { config_.rtp.rtcp_mode = mode; }
void SetFlexFecProtection(RtpPacketSinkInterface* sink) override {
config_.rtp.packet_sink_ = sink;
config_.rtp.protected_by_flexfec = (sink != nullptr);
}
void SetLossNotificationEnabled(bool enabled) override {
config_.rtp.lntf.enabled = enabled;
}
void SetNackHistory(TimeDelta history) override {
config_.rtp.nack.rtp_history_ms = history.ms();
}
void SetProtectionPayloadTypes(int red_payload_type,
int ulpfec_payload_type) override {
config_.rtp.red_payload_type = red_payload_type;
config_.rtp.ulpfec_payload_type = ulpfec_payload_type;
}
void SetRtcpXr(Config::Rtp::RtcpXr rtcp_xr) override {
config_.rtp.rtcp_xr = rtcp_xr;
}
void SetAssociatedPayloadTypes(
std::map<int, int> associated_payload_types) override {
config_.rtp.rtx_associated_payload_types =
std::move(associated_payload_types);
}
void Start() override;
void Stop() override;
VideoReceiveStreamInterface::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_;
}
private :
VideoReceiveStreamInterface::Config config_;
bool receiving_;
VideoReceiveStreamInterface::Stats stats_;
int base_mininum_playout_delay_ms_ = 0 ;
};
class FakeFlexfecReceiveStream final : public FlexfecReceiveStream {
public :
explicit FakeFlexfecReceiveStream(const FlexfecReceiveStream::Config config);
void SetRtcpMode(RtcpMode mode) override { config_.rtcp_mode = mode; }
int payload_type() const override { return config_.payload_type; }
void SetPayloadType(int payload_type) override {
config_.payload_type = payload_type;
}
const FlexfecReceiveStream::Config& GetConfig() const ;
uint32_t remote_ssrc() const { return config_.remote_ssrc; }
const ReceiveStatistics* GetStats() const override { return nullptr; }
private :
void OnRtpPacket(const RtpPacketReceived& packet) override;
FlexfecReceiveStream::Config config_;
};
class FakeCall final : public Call, public PacketReceiver {
public :
explicit FakeCall(const Environment& env);
FakeCall(const Environment& env,
TaskQueueBase* worker_thread,
TaskQueueBase* network_thread);
~FakeCall() override;
PayloadTypeSuggester* GetPayloadTypeSuggester() { return &pt_suggester_; }
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();
SentPacketInfo last_sent_packet() const { return last_sent_packet_; }
const RtpPacketReceived& last_received_rtp_packet() const {
return last_received_rtp_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 : 0 u;
}
// 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_;
}
NetworkState GetNetworkState(MediaType media) const ;
int GetNumCreatedSendStreams() const ;
int GetNumCreatedReceiveStreams() const ;
void SetStats(const Call::Stats& stats);
void SetClientBitratePreferences(
const BitrateSettings& /* preferences */) override {}
void SetPreferredRtcpCcAckType(
RtcpFeedbackType preferred_rtcp_cc_ack_type) override {}
std::optional<int> FeedbackAccordingToRfc8888Count() override { return 0 ; }
std::optional<int> FeedbackAccordingToTransportCcCount() override {
return 0 ;
}
private :
AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) override;
void DestroyAudioSendStream(AudioSendStream* send_stream) override;
AudioReceiveStreamInterface* CreateAudioReceiveStream(
const AudioReceiveStreamInterface::Config& config) override;
void DestroyAudioReceiveStream(
AudioReceiveStreamInterface* receive_stream) override;
VideoSendStream* CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
EncoderSwitchRequestCallback encoder_switch_request_callback =
nullptr) override;
VideoSendStream* CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
EncoderSwitchRequestCallback encoder_switch_request_callback,
std::unique_ptr<FecController> fec_controller) override;
void DestroyVideoSendStream(VideoSendStream* send_stream) override;
VideoReceiveStreamInterface* CreateVideoReceiveStream(
VideoReceiveStreamInterface::Config config) override;
void DestroyVideoReceiveStream(
VideoReceiveStreamInterface* receive_stream) override;
FlexfecReceiveStream* CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config config) override;
void DestroyFlexfecReceiveStream(
FlexfecReceiveStream* receive_stream) override;
void AddAdaptationResource(scoped_refptr<Resource> resource) override;
PacketReceiver* Receiver() override;
void DeliverRtcpPacket(CopyOnWriteBuffer /* packet */) override {}
void DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
OnUndemuxablePacketHandler un_demuxable_packet_handler) override;
bool DeliverPacketInternal(MediaType media_type,
uint32_t ssrc,
const CopyOnWriteBuffer& packet,
Timestamp arrival_time);
RtpTransportControllerSendInterface* GetTransportControllerSend() override {
return &transport_controller_send_;
}
Call::Stats GetStats() const override;
TaskQueueBase* network_thread() const override;
TaskQueueBase* worker_thread() const override;
void SignalChannelNetworkState(MediaType media, NetworkState state) override;
void OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) override;
void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream,
absl::string_view sync_group) override;
void OnSentPacket(const SentPacketInfo& sent_packet) override;
const Environment env_;
TaskQueueBase* const network_thread_;
TaskQueueBase* const worker_thread_;
::testing::NiceMock<MockRtpTransportControllerSend>
transport_controller_send_;
NetworkState audio_network_state_;
NetworkState video_network_state_;
SentPacketInfo last_sent_packet_;
RtpPacketReceived last_received_rtp_packet_;
int last_sent_nonnegative_packet_id_ = -1 ;
int next_stream_id_ = 665 ;
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_;
FakePayloadTypeSuggester pt_suggester_;
};
} // namespace webrtc
#endif // MEDIA_ENGINE_FAKE_WEBRTC_CALL_H_
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