| /* |
| * Copyright (c) 2012 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. |
| */ |
| |
| #ifndef VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_ |
| #define VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/crypto/frame_decryptor_interface.h" |
| #include "api/sequence_checker.h" |
| #include "api/units/timestamp.h" |
| #include "api/video/color_space.h" |
| #include "api/video/video_codec_type.h" |
| #include "call/rtp_packet_sink_interface.h" |
| #include "call/syncable.h" |
| #include "call/video_receive_stream.h" |
| #include "modules/rtp_rtcp/include/receive_statistics.h" |
| #include "modules/rtp_rtcp/include/recovered_packet_receiver.h" |
| #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| #include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h" |
| #include "modules/rtp_rtcp/source/capture_clock_offset_updater.h" |
| #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" |
| #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" |
| #include "modules/rtp_rtcp/source/rtp_video_header.h" |
| #include "modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate.h" |
| #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h" |
| #include "modules/video_coding/h264_sps_pps_tracker.h" |
| #include "modules/video_coding/loss_notification_controller.h" |
| #include "modules/video_coding/nack_requester.h" |
| #include "modules/video_coding/packet_buffer.h" |
| #include "modules/video_coding/rtp_frame_reference_finder.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| #include "rtc_base/numerics/sequence_number_unwrapper.h" |
| #include "rtc_base/system/no_unique_address.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "video/buffered_frame_decryptor.h" |
| #include "video/unique_timestamp_counter.h" |
| |
| namespace webrtc { |
| |
| class NackRequester; |
| class PacketRouter; |
| class ReceiveStatistics; |
| class RtcpRttStats; |
| class RtpPacketReceived; |
| class Transport; |
| class UlpfecReceiver; |
| |
| class RtpVideoStreamReceiver2 : public LossNotificationSender, |
| public RecoveredPacketReceiver, |
| public RtpPacketSinkInterface, |
| public KeyFrameRequestSender, |
| public NackSender, |
| public OnDecryptedFrameCallback, |
| public OnDecryptionStatusChangeCallback, |
| public RtpVideoFrameReceiver { |
| public: |
| // A complete frame is a frame which has received all its packets and all its |
| // references are known. |
| class OnCompleteFrameCallback { |
| public: |
| virtual ~OnCompleteFrameCallback() {} |
| virtual void OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) = 0; |
| }; |
| |
| RtpVideoStreamReceiver2( |
| TaskQueueBase* current_queue, |
| Clock* clock, |
| Transport* transport, |
| RtcpRttStats* rtt_stats, |
| // The packet router is optional; if provided, the RtpRtcp module for this |
| // stream is registered as a candidate for sending REMB and transport |
| // feedback. |
| PacketRouter* packet_router, |
| const VideoReceiveStreamInterface::Config* config, |
| ReceiveStatistics* rtp_receive_statistics, |
| RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
| RtcpCnameCallback* rtcp_cname_callback, |
| NackPeriodicProcessor* nack_periodic_processor, |
| // The KeyFrameRequestSender is optional; if not provided, key frame |
| // requests are sent via the internal RtpRtcp module. |
| OnCompleteFrameCallback* complete_frame_callback, |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, |
| const FieldTrialsView& field_trials, |
| RtcEventLog* event_log); |
| ~RtpVideoStreamReceiver2() override; |
| |
| void AddReceiveCodec(uint8_t payload_type, |
| VideoCodecType video_codec, |
| const webrtc::CodecParameterMap& codec_params, |
| bool raw_payload); |
| void RemoveReceiveCodec(uint8_t payload_type); |
| |
| // Clears state for all receive codecs added via `AddReceiveCodec`. |
| void RemoveReceiveCodecs(); |
| |
| void StartReceive(); |
| void StopReceive(); |
| |
| // Produces the transport-related timestamps; current_delay_ms is left unset. |
| absl::optional<Syncable::Info> GetSyncInfo() const; |
| |
| bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length); |
| |
| void FrameContinuous(int64_t seq_num); |
| |
| void FrameDecoded(int64_t seq_num); |
| |
| void SignalNetworkState(NetworkState state); |
| |
| // Returns number of different frames seen. |
| int GetUniqueFramesSeen() const { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| return frame_counter_.GetUniqueSeen(); |
| } |
| |
| // Implements RtpPacketSinkInterface. |
| void OnRtpPacket(const RtpPacketReceived& packet) override; |
| |
| // Public only for tests. |
| // Returns true if the packet should be stashed and retried at a later stage. |
| bool OnReceivedPayloadData(rtc::CopyOnWriteBuffer codec_payload, |
| const RtpPacketReceived& rtp_packet, |
| const RTPVideoHeader& video, |
| int times_nacked); |
| |
| // Implements RecoveredPacketReceiver. |
| void OnRecoveredPacket(const RtpPacketReceived& packet) override; |
| |
| // Send an RTCP keyframe request. |
| void RequestKeyFrame() override; |
| |
| // Implements NackSender. |
| void SendNack(const std::vector<uint16_t>& sequence_numbers, |
| bool buffering_allowed) override; |
| |
| // Implements LossNotificationSender. |
| void SendLossNotification(uint16_t last_decoded_seq_num, |
| uint16_t last_received_seq_num, |
| bool decodability_flag, |
| bool buffering_allowed) override; |
| |
| // Returns true if a decryptor is attached and frames can be decrypted. |
| // Updated by OnDecryptionStatusChangeCallback. Note this refers to Frame |
| // Decryption not SRTP. |
| bool IsDecryptable() const; |
| |
| // Implements OnDecryptedFrameCallback. |
| void OnDecryptedFrame(std::unique_ptr<RtpFrameObject> frame) override; |
| |
| // Implements OnDecryptionStatusChangeCallback. |
| void OnDecryptionStatusChange( |
| FrameDecryptorInterface::Status status) override; |
| |
| // Optionally set a frame decryptor after a stream has started. This will not |
| // reset the decoder state. |
| void SetFrameDecryptor( |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor); |
| |
| // Sets a frame transformer after a stream has started, if no transformer |
| // has previously been set. Does not reset the decoder state. |
| void SetDepacketizerToDecoderFrameTransformer( |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer); |
| |
| // Called by VideoReceiveStreamInterface when stats are updated. |
| void UpdateRtt(int64_t max_rtt_ms); |
| |
| // Called when the local_ssrc is changed to match with a sender. |
| void OnLocalSsrcChange(uint32_t local_ssrc); |
| |
| // Forwards the call to set rtcp_sender_ to the RTCP mode of the rtcp sender. |
| void SetRtcpMode(RtcpMode mode); |
| |
| void SetReferenceTimeReport(bool enabled); |
| |
| // Sets or clears the callback sink that gets called for RTP packets. Used for |
| // packet handlers such as FlexFec. Must be called on the packet delivery |
| // thread (same context as `OnRtpPacket` is called on). |
| // TODO(bugs.webrtc.org/11993): Packet delivery thread today means `worker |
| // thread` but will be `network thread`. |
| void SetPacketSink(RtpPacketSinkInterface* packet_sink); |
| |
| // Turns on/off loss notifications. Must be called on the packet delivery |
| // thread. |
| void SetLossNotificationEnabled(bool enabled); |
| |
| void SetNackHistory(TimeDelta history); |
| |
| int ulpfec_payload_type() const; |
| int red_payload_type() const; |
| void SetProtectionPayloadTypes(int red_payload_type, int ulpfec_payload_type); |
| |
| absl::optional<int64_t> LastReceivedPacketMs() const; |
| absl::optional<uint32_t> LastReceivedFrameRtpTimestamp() const; |
| absl::optional<int64_t> LastReceivedKeyframePacketMs() const; |
| |
| private: |
| // Implements RtpVideoFrameReceiver. |
| void ManageFrame(std::unique_ptr<RtpFrameObject> frame) override; |
| |
| void OnCompleteFrames(RtpFrameReferenceFinder::ReturnVector frame) |
| RTC_RUN_ON(packet_sequence_checker_); |
| |
| // Used for buffering RTCP feedback messages and sending them all together. |
| // Note: |
| // 1. Key frame requests and NACKs are mutually exclusive, with the |
| // former taking precedence over the latter. |
| // 2. Loss notifications are orthogonal to either. (That is, may be sent |
| // alongside either.) |
| class RtcpFeedbackBuffer : public KeyFrameRequestSender, |
| public NackSender, |
| public LossNotificationSender { |
| public: |
| RtcpFeedbackBuffer(KeyFrameRequestSender* key_frame_request_sender, |
| NackSender* nack_sender, |
| LossNotificationSender* loss_notification_sender); |
| |
| ~RtcpFeedbackBuffer() override = default; |
| |
| // KeyFrameRequestSender implementation. |
| void RequestKeyFrame() override; |
| |
| // NackSender implementation. |
| void SendNack(const std::vector<uint16_t>& sequence_numbers, |
| bool buffering_allowed) override; |
| |
| // LossNotificationSender implementation. |
| void SendLossNotification(uint16_t last_decoded_seq_num, |
| uint16_t last_received_seq_num, |
| bool decodability_flag, |
| bool buffering_allowed) override; |
| |
| // Send all RTCP feedback messages buffered thus far. |
| void SendBufferedRtcpFeedback(); |
| |
| void ClearLossNotificationState(); |
| |
| private: |
| // LNTF-related state. |
| struct LossNotificationState { |
| LossNotificationState(uint16_t last_decoded_seq_num, |
| uint16_t last_received_seq_num, |
| bool decodability_flag) |
| : last_decoded_seq_num(last_decoded_seq_num), |
| last_received_seq_num(last_received_seq_num), |
| decodability_flag(decodability_flag) {} |
| |
| uint16_t last_decoded_seq_num; |
| uint16_t last_received_seq_num; |
| bool decodability_flag; |
| }; |
| |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_; |
| KeyFrameRequestSender* const key_frame_request_sender_; |
| NackSender* const nack_sender_; |
| LossNotificationSender* const loss_notification_sender_; |
| |
| // Key-frame-request-related state. |
| bool request_key_frame_ RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| // NACK-related state. |
| std::vector<uint16_t> nack_sequence_numbers_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| absl::optional<LossNotificationState> lntf_state_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| }; |
| enum ParseGenericDependenciesResult { |
| kStashPacket, |
| kDropPacket, |
| kHasGenericDescriptor, |
| kNoGenericDescriptor |
| }; |
| |
| // Entry point doing non-stats work for a received packet. Called |
| // for the same packet both before and after RED decapsulation. |
| void ReceivePacket(const RtpPacketReceived& packet) |
| RTC_RUN_ON(packet_sequence_checker_); |
| |
| // Parses and handles RED headers. |
| // This function assumes that it's being called from only one thread. |
| void ParseAndHandleEncapsulatingHeader(const RtpPacketReceived& packet) |
| RTC_RUN_ON(packet_sequence_checker_); |
| void NotifyReceiverOfEmptyPacket(uint16_t seq_num) |
| RTC_RUN_ON(packet_sequence_checker_); |
| bool IsRedEnabled() const; |
| void InsertSpsPpsIntoTracker(uint8_t payload_type) |
| RTC_RUN_ON(packet_sequence_checker_); |
| void OnInsertedPacket(video_coding::PacketBuffer::InsertResult result) |
| RTC_RUN_ON(packet_sequence_checker_); |
| ParseGenericDependenciesResult ParseGenericDependenciesExtension( |
| const RtpPacketReceived& rtp_packet, |
| RTPVideoHeader* video_header) RTC_RUN_ON(packet_sequence_checker_); |
| void OnAssembledFrame(std::unique_ptr<RtpFrameObject> frame) |
| RTC_RUN_ON(packet_sequence_checker_); |
| void UpdatePacketReceiveTimestamps(const RtpPacketReceived& packet, |
| bool is_keyframe) |
| RTC_RUN_ON(packet_sequence_checker_); |
| |
| const FieldTrialsView& field_trials_; |
| TaskQueueBase* const worker_queue_; |
| Clock* const clock_; |
| // Ownership of this object lies with VideoReceiveStreamInterface, which owns |
| // `this`. |
| const VideoReceiveStreamInterface::Config& config_; |
| PacketRouter* const packet_router_; |
| |
| RemoteNtpTimeEstimator ntp_estimator_; |
| |
| // Set by the field trial WebRTC-ForcePlayoutDelay to override any playout |
| // delay that is specified in the received packets. |
| FieldTrialOptional<int> forced_playout_delay_max_ms_; |
| FieldTrialOptional<int> forced_playout_delay_min_ms_; |
| ReceiveStatistics* const rtp_receive_statistics_; |
| std::unique_ptr<UlpfecReceiver> ulpfec_receiver_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| int red_payload_type_ RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_task_checker_; |
| // TODO(bugs.webrtc.org/11993): This checker conceptually represents |
| // operations that belong to the network thread. The Call class is currently |
| // moving towards handling network packets on the network thread and while |
| // that work is ongoing, this checker may in practice represent the worker |
| // thread, but still serves as a mechanism of grouping together concepts |
| // that belong to the network thread. Once the packets are fully delivered |
| // on the network thread, this comment will be deleted. |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_; |
| RtpPacketSinkInterface* packet_sink_ RTC_GUARDED_BY(packet_sequence_checker_); |
| bool receiving_ RTC_GUARDED_BY(packet_sequence_checker_); |
| int64_t last_packet_log_ms_ RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| const std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; |
| |
| NackPeriodicProcessor* const nack_periodic_processor_; |
| OnCompleteFrameCallback* complete_frame_callback_; |
| const KeyFrameReqMethod keyframe_request_method_; |
| |
| RtcpFeedbackBuffer rtcp_feedback_buffer_; |
| // TODO(tommi): Consider absl::optional<NackRequester> instead of unique_ptr |
| // since nack is usually configured. |
| std::unique_ptr<NackRequester> nack_module_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| std::unique_ptr<LossNotificationController> loss_notification_controller_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| video_coding::PacketBuffer packet_buffer_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| UniqueTimestampCounter frame_counter_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| SeqNumUnwrapper<uint16_t> frame_id_unwrapper_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| // Video structure provided in the dependency descriptor in a first packet |
| // of a key frame. It is required to parse dependency descriptor in the |
| // following delta packets. |
| std::unique_ptr<FrameDependencyStructure> video_structure_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| // Frame id of the last frame with the attached video structure. |
| // absl::nullopt when `video_structure_ == nullptr`; |
| absl::optional<int64_t> video_structure_frame_id_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| Timestamp last_logged_failed_to_parse_dd_ |
| RTC_GUARDED_BY(packet_sequence_checker_) = Timestamp::MinusInfinity(); |
| |
| std::unique_ptr<RtpFrameReferenceFinder> reference_finder_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| absl::optional<VideoCodecType> current_codec_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| uint32_t last_assembled_frame_rtp_timestamp_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| std::map<int64_t, uint16_t> last_seq_num_for_pic_id_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| video_coding::H264SpsPpsTracker tracker_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| // Maps payload id to the depacketizer. |
| std::map<uint8_t, std::unique_ptr<VideoRtpDepacketizer>> payload_type_map_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| // TODO(johan): Remove pt_codec_params_ once |
| // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved. |
| // Maps a payload type to a map of out-of-band supplied codec parameters. |
| std::map<uint8_t, webrtc::CodecParameterMap> pt_codec_params_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| int16_t last_payload_type_ RTC_GUARDED_BY(packet_sequence_checker_) = -1; |
| |
| bool has_received_frame_ RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| absl::optional<uint32_t> last_received_rtp_timestamp_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| absl::optional<uint32_t> last_received_keyframe_rtp_timestamp_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| absl::optional<Timestamp> last_received_rtp_system_time_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| absl::optional<Timestamp> last_received_keyframe_rtp_system_time_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| // Handles incoming encrypted frames and forwards them to the |
| // rtp_reference_finder if they are decryptable. |
| std::unique_ptr<BufferedFrameDecryptor> buffered_frame_decryptor_ |
| RTC_PT_GUARDED_BY(packet_sequence_checker_); |
| bool frames_decryptable_ RTC_GUARDED_BY(worker_task_checker_); |
| absl::optional<ColorSpace> last_color_space_; |
| |
| AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| CaptureClockOffsetUpdater capture_clock_offset_updater_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| int64_t last_completed_picture_id_ = 0; |
| |
| rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate> |
| frame_transformer_delegate_; |
| |
| SeqNumUnwrapper<uint16_t> rtp_seq_num_unwrapper_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| std::map<int64_t, RtpPacketInfo> packet_infos_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| std::vector<RtpPacketReceived> stashed_packets_ |
| RTC_GUARDED_BY(packet_sequence_checker_); |
| |
| Timestamp next_keyframe_request_for_missing_video_structure_ = |
| Timestamp::MinusInfinity(); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_ |