| /* |
| * 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 WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_ |
| #define WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_ |
| |
| #include <list> |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/call/rtp_packet_sink_interface.h" |
| #include "webrtc/call/video_receive_stream.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/modules/video_coding/h264_sps_pps_tracker.h" |
| #include "webrtc/modules/video_coding/include/video_coding_defines.h" |
| #include "webrtc/modules/video_coding/packet_buffer.h" |
| #include "webrtc/modules/video_coding/rtp_frame_reference_finder.h" |
| #include "webrtc/modules/video_coding/sequence_number_util.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| #include "webrtc/rtc_base/criticalsection.h" |
| #include "webrtc/rtc_base/sequenced_task_checker.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class NackModule; |
| class PacedSender; |
| class PacketRouter; |
| class ProcessThread; |
| class ReceiveStatistics; |
| class ReceiveStatisticsProxy; |
| class RemoteNtpTimeEstimator; |
| class RtcpRttStats; |
| class RtpHeaderParser; |
| class RtpPacketReceived; |
| class RTPPayloadRegistry; |
| class RtpReceiver; |
| class Transport; |
| class UlpfecReceiver; |
| class VCMTiming; |
| |
| namespace vcm { |
| class VideoReceiver; |
| } // namespace vcm |
| |
| class RtpVideoStreamReceiver : public RtpData, |
| public RecoveredPacketReceiver, |
| public RtpFeedback, |
| public RtpPacketSinkInterface, |
| public VCMFrameTypeCallback, |
| public VCMPacketRequestCallback, |
| public video_coding::OnReceivedFrameCallback, |
| public video_coding::OnCompleteFrameCallback, |
| public CallStatsObserver { |
| public: |
| RtpVideoStreamReceiver( |
| Transport* transport, |
| RtcpRttStats* rtt_stats, |
| PacketRouter* packet_router, |
| const VideoReceiveStream::Config* config, |
| ReceiveStatistics* rtp_receive_statistics, |
| ReceiveStatisticsProxy* receive_stats_proxy, |
| ProcessThread* process_thread, |
| NackSender* nack_sender, |
| KeyFrameRequestSender* keyframe_request_sender, |
| video_coding::OnCompleteFrameCallback* complete_frame_callback, |
| VCMTiming* timing); |
| ~RtpVideoStreamReceiver(); |
| |
| bool AddReceiveCodec(const VideoCodec& video_codec, |
| const std::map<std::string, std::string>& codec_params); |
| uint32_t GetRemoteSsrc() const; |
| int GetCsrcs(uint32_t* csrcs) const; |
| |
| RtpReceiver* GetRtpReceiver() const; |
| RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); } |
| |
| void StartReceive(); |
| void StopReceive(); |
| |
| 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); |
| |
| // Implements RtpPacketSinkInterface. |
| void OnRtpPacket(const RtpPacketReceived& packet) override; |
| |
| // Implements RtpData. |
| int32_t OnReceivedPayloadData(const uint8_t* payload_data, |
| size_t payload_size, |
| const WebRtcRTPHeader* rtp_header) override; |
| // Implements RecoveredPacketReceiver. |
| void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override; |
| |
| // Implements RtpFeedback. |
| int32_t OnInitializeDecoder(int8_t payload_type, |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| int frequency, |
| size_t channels, |
| uint32_t rate) override; |
| void OnIncomingSSRCChanged(uint32_t ssrc) override {} |
| void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override {} |
| |
| // Implements VCMFrameTypeCallback. |
| int32_t RequestKeyFrame() override; |
| |
| bool IsUlpfecEnabled() const; |
| bool IsRetransmissionsEnabled() const; |
| // Don't use, still experimental. |
| void RequestPacketRetransmit(const std::vector<uint16_t>& sequence_numbers); |
| |
| // Implements VCMPacketRequestCallback. |
| int32_t ResendPackets(const uint16_t* sequenceNumbers, |
| uint16_t length) override; |
| |
| // Implements OnReceivedFrameCallback. |
| void OnReceivedFrame( |
| std::unique_ptr<video_coding::RtpFrameObject> frame) override; |
| |
| // Implements OnCompleteFrameCallback. |
| void OnCompleteFrame( |
| std::unique_ptr<video_coding::FrameObject> frame) override; |
| |
| void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override; |
| |
| rtc::Optional<int64_t> LastReceivedPacketMs() const; |
| rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const; |
| |
| // RtpDemuxer only forwards a given RTP packet to one sink. However, some |
| // sinks, such as FlexFEC, might wish to be informed of all of the packets |
| // a given sink receives (or any set of sinks). They may do so by registering |
| // themselves as secondary sinks. |
| void AddSecondarySink(RtpPacketSinkInterface* sink); |
| void RemoveSecondarySink(const RtpPacketSinkInterface* sink); |
| |
| private: |
| bool AddReceiveCodec(const VideoCodec& video_codec); |
| void ReceivePacket(const uint8_t* packet, |
| size_t packet_length, |
| const RTPHeader& header, |
| bool in_order); |
| // Parses and handles for instance RTX and RED headers. |
| // This function assumes that it's being called from only one thread. |
| void ParseAndHandleEncapsulatingHeader(const uint8_t* packet, |
| size_t packet_length, |
| const RTPHeader& header); |
| void NotifyReceiverOfFecPacket(const RTPHeader& header); |
| bool IsPacketInOrder(const RTPHeader& header) const; |
| bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; |
| void UpdateHistograms(); |
| void EnableReceiveRtpHeaderExtension(const std::string& extension, int id); |
| bool IsRedEnabled() const; |
| void InsertSpsPpsIntoTracker(uint8_t payload_type); |
| |
| Clock* const clock_; |
| // Ownership of this object lies with VideoReceiveStream, which owns |this|. |
| const VideoReceiveStream::Config& config_; |
| PacketRouter* const packet_router_; |
| ProcessThread* const process_thread_; |
| |
| RemoteNtpTimeEstimator ntp_estimator_; |
| RTPPayloadRegistry rtp_payload_registry_; |
| |
| const std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
| const std::unique_ptr<RtpReceiver> rtp_receiver_; |
| ReceiveStatistics* const rtp_receive_statistics_; |
| std::unique_ptr<UlpfecReceiver> ulpfec_receiver_; |
| |
| rtc::SequencedTaskChecker worker_task_checker_; |
| bool receiving_ GUARDED_BY(worker_task_checker_); |
| int64_t last_packet_log_ms_ GUARDED_BY(worker_task_checker_); |
| |
| const std::unique_ptr<RtpRtcp> rtp_rtcp_; |
| |
| // Members for the new jitter buffer experiment. |
| video_coding::OnCompleteFrameCallback* complete_frame_callback_; |
| KeyFrameRequestSender* keyframe_request_sender_; |
| VCMTiming* timing_; |
| std::unique_ptr<NackModule> nack_module_; |
| rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_; |
| std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_; |
| rtc::CriticalSection last_seq_num_cs_; |
| std::map<int64_t, uint16_t> last_seq_num_for_pic_id_ |
| GUARDED_BY(last_seq_num_cs_); |
| video_coding::H264SpsPpsTracker tracker_; |
| // 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, std::map<std::string, std::string>> pt_codec_params_; |
| int16_t last_payload_type_ = -1; |
| |
| bool has_received_frame_; |
| |
| std::vector<RtpPacketSinkInterface*> secondary_sinks_ |
| GUARDED_BY(worker_task_checker_); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_ |