Add a new API in RTReceiverInterface, to be called from the browser side to insert a frame transformer between the Depacketizer and the Decoder. The frame transformer is passed from RTReceiverInterface through the library to be eventually set in RtpVideoStreamReceiver, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169130. This change is part of the implementation of the Insertable Streams Web API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30654}
379 lines
15 KiB
C++
379 lines
15 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_
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#define VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_
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#include <atomic>
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#include <list>
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/video/color_space.h"
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#include "api/video_codecs/video_codec.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "call/syncable.h"
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#include "call/video_receive_stream.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/absolute_capture_time_receiver.h"
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#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
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#include "modules/video_coding/h264_sps_pps_tracker.h"
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#include "modules/video_coding/loss_notification_controller.h"
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#include "modules/video_coding/packet_buffer.h"
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#include "modules/video_coding/rtp_frame_reference_finder.h"
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#include "modules/video_coding/unique_timestamp_counter.h"
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#include "rtc_base/constructor_magic.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/numerics/sequence_number_util.h"
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#include "rtc_base/synchronization/sequence_checker.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/thread_checker.h"
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#include "video/buffered_frame_decryptor.h"
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namespace webrtc {
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class NackModule;
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class PacketRouter;
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class ProcessThread;
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class ReceiveStatistics;
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class ReceiveStatisticsProxy;
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class RtcpRttStats;
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class RtpPacketReceived;
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class Transport;
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class UlpfecReceiver;
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class RtpVideoStreamReceiver : public LossNotificationSender,
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public RecoveredPacketReceiver,
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public RtpPacketSinkInterface,
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public KeyFrameRequestSender,
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public video_coding::OnCompleteFrameCallback,
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public OnDecryptedFrameCallback,
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public OnDecryptionStatusChangeCallback {
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public:
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RtpVideoStreamReceiver(
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Clock* clock,
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Transport* transport,
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RtcpRttStats* rtt_stats,
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// The packet router is optional; if provided, the RtpRtcp module for this
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// stream is registered as a candidate for sending REMB and transport
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// feedback.
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PacketRouter* packet_router,
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const VideoReceiveStream::Config* config,
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ReceiveStatistics* rtp_receive_statistics,
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ReceiveStatisticsProxy* receive_stats_proxy,
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ProcessThread* process_thread,
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NackSender* nack_sender,
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// The KeyFrameRequestSender is optional; if not provided, key frame
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// requests are sent via the internal RtpRtcp module.
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KeyFrameRequestSender* keyframe_request_sender,
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video_coding::OnCompleteFrameCallback* complete_frame_callback,
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
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// TODO(bugs.webrtc.org/11380) remove after updating downstream dependencies
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// to use the new constructor.
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RTC_DEPRECATED
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RtpVideoStreamReceiver(
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Clock* clock,
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Transport* transport,
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RtcpRttStats* rtt_stats,
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// The packet router is optional; if provided, the RtpRtcp module for this
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// stream is registered as a candidate for sending REMB and transport
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// feedback.
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PacketRouter* packet_router,
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const VideoReceiveStream::Config* config,
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ReceiveStatistics* rtp_receive_statistics,
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ReceiveStatisticsProxy* receive_stats_proxy,
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ProcessThread* process_thread,
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NackSender* nack_sender,
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// The KeyFrameRequestSender is optional; if not provided, key frame
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// requests are sent via the internal RtpRtcp module.
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KeyFrameRequestSender* keyframe_request_sender,
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video_coding::OnCompleteFrameCallback* complete_frame_callback,
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
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~RtpVideoStreamReceiver() override;
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void AddReceiveCodec(const VideoCodec& video_codec,
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const std::map<std::string, std::string>& codec_params,
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bool raw_payload);
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void StartReceive();
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void StopReceive();
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// Produces the transport-related timestamps; current_delay_ms is left unset.
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absl::optional<Syncable::Info> GetSyncInfo() const;
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bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
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void FrameContinuous(int64_t seq_num);
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void FrameDecoded(int64_t seq_num);
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void SignalNetworkState(NetworkState state);
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// Returns number of different frames seen.
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int GetUniqueFramesSeen() const {
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RTC_DCHECK_RUN_ON(&worker_task_checker_);
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return frame_counter_.GetUniqueSeen();
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}
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// Implements RtpPacketSinkInterface.
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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// TODO(philipel): Stop using VCMPacket in the new jitter buffer and then
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// remove this function. Public only for tests.
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void OnReceivedPayloadData(rtc::CopyOnWriteBuffer codec_payload,
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const RtpPacketReceived& rtp_packet,
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const RTPVideoHeader& video);
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// Implements RecoveredPacketReceiver.
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void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
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// Send an RTCP keyframe request.
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void RequestKeyFrame() override;
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// Implements LossNotificationSender.
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void SendLossNotification(uint16_t last_decoded_seq_num,
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uint16_t last_received_seq_num,
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bool decodability_flag,
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bool buffering_allowed) override;
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bool IsUlpfecEnabled() const;
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bool IsRetransmissionsEnabled() const;
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// Returns true if a decryptor is attached and frames can be decrypted.
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// Updated by OnDecryptionStatusChangeCallback. Note this refers to Frame
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// Decryption not SRTP.
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bool IsDecryptable() const;
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// Don't use, still experimental.
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void RequestPacketRetransmit(const std::vector<uint16_t>& sequence_numbers);
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// Implements OnCompleteFrameCallback.
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void OnCompleteFrame(
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std::unique_ptr<video_coding::EncodedFrame> frame) override;
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// Implements OnDecryptedFrameCallback.
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void OnDecryptedFrame(
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std::unique_ptr<video_coding::RtpFrameObject> frame) override;
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// Implements OnDecryptionStatusChangeCallback.
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void OnDecryptionStatusChange(
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FrameDecryptorInterface::Status status) override;
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// Optionally set a frame decryptor after a stream has started. This will not
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// reset the decoder state.
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void SetFrameDecryptor(
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
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// Called by VideoReceiveStream when stats are updated.
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void UpdateRtt(int64_t max_rtt_ms);
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absl::optional<int64_t> LastReceivedPacketMs() const;
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absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
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// RtpDemuxer only forwards a given RTP packet to one sink. However, some
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// sinks, such as FlexFEC, might wish to be informed of all of the packets
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// a given sink receives (or any set of sinks). They may do so by registering
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// themselves as secondary sinks.
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void AddSecondarySink(RtpPacketSinkInterface* sink);
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void RemoveSecondarySink(const RtpPacketSinkInterface* sink);
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private:
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// Used for buffering RTCP feedback messages and sending them all together.
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// Note:
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// 1. Key frame requests and NACKs are mutually exclusive, with the
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// former taking precedence over the latter.
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// 2. Loss notifications are orthogonal to either. (That is, may be sent
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// alongside either.)
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class RtcpFeedbackBuffer : public KeyFrameRequestSender,
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public NackSender,
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public LossNotificationSender {
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public:
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RtcpFeedbackBuffer(KeyFrameRequestSender* key_frame_request_sender,
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NackSender* nack_sender,
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LossNotificationSender* loss_notification_sender);
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~RtcpFeedbackBuffer() override = default;
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// KeyFrameRequestSender implementation.
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void RequestKeyFrame() override;
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// NackSender implementation.
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void SendNack(const std::vector<uint16_t>& sequence_numbers,
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bool buffering_allowed) override;
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// LossNotificationSender implementation.
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void SendLossNotification(uint16_t last_decoded_seq_num,
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uint16_t last_received_seq_num,
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bool decodability_flag,
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bool buffering_allowed) override;
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// Send all RTCP feedback messages buffered thus far.
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void SendBufferedRtcpFeedback();
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private:
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KeyFrameRequestSender* const key_frame_request_sender_;
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NackSender* const nack_sender_;
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LossNotificationSender* const loss_notification_sender_;
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// NACKs are accessible from two threads due to nack_module_ being a module.
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rtc::CriticalSection cs_;
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// Key-frame-request-related state.
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bool request_key_frame_ RTC_GUARDED_BY(cs_);
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// NACK-related state.
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std::vector<uint16_t> nack_sequence_numbers_ RTC_GUARDED_BY(cs_);
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// LNTF-related state.
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struct LossNotificationState {
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LossNotificationState(uint16_t last_decoded_seq_num,
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uint16_t last_received_seq_num,
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bool decodability_flag)
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: last_decoded_seq_num(last_decoded_seq_num),
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last_received_seq_num(last_received_seq_num),
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decodability_flag(decodability_flag) {}
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uint16_t last_decoded_seq_num;
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uint16_t last_received_seq_num;
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bool decodability_flag;
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};
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absl::optional<LossNotificationState> lntf_state_ RTC_GUARDED_BY(cs_);
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};
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enum ParseGenericDependenciesResult {
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kDropPacket,
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kHasGenericDescriptor,
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kNoGenericDescriptor
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};
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// Entry point doing non-stats work for a received packet. Called
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// for the same packet both before and after RED decapsulation.
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void ReceivePacket(const RtpPacketReceived& packet);
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// Parses and handles RED headers.
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// This function assumes that it's being called from only one thread.
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void ParseAndHandleEncapsulatingHeader(const RtpPacketReceived& packet);
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void NotifyReceiverOfEmptyPacket(uint16_t seq_num);
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void UpdateHistograms();
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bool IsRedEnabled() const;
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void InsertSpsPpsIntoTracker(uint8_t payload_type);
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void OnInsertedPacket(video_coding::PacketBuffer::InsertResult result);
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ParseGenericDependenciesResult ParseGenericDependenciesExtension(
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const RtpPacketReceived& rtp_packet,
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RTPVideoHeader* video_header) RTC_RUN_ON(worker_task_checker_);
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void OnAssembledFrame(std::unique_ptr<video_coding::RtpFrameObject> frame);
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Clock* const clock_;
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// Ownership of this object lies with VideoReceiveStream, which owns |this|.
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const VideoReceiveStream::Config& config_;
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PacketRouter* const packet_router_;
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ProcessThread* const process_thread_;
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RemoteNtpTimeEstimator ntp_estimator_;
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RtpHeaderExtensionMap rtp_header_extensions_;
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ReceiveStatistics* const rtp_receive_statistics_;
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std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
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SequenceChecker worker_task_checker_;
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bool receiving_ RTC_GUARDED_BY(worker_task_checker_);
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int64_t last_packet_log_ms_ RTC_GUARDED_BY(worker_task_checker_);
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const std::unique_ptr<RtpRtcp> rtp_rtcp_;
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video_coding::OnCompleteFrameCallback* complete_frame_callback_;
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KeyFrameRequestSender* const keyframe_request_sender_;
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RtcpFeedbackBuffer rtcp_feedback_buffer_;
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std::unique_ptr<NackModule> nack_module_;
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std::unique_ptr<LossNotificationController> loss_notification_controller_;
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video_coding::PacketBuffer packet_buffer_;
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UniqueTimestampCounter frame_counter_ RTC_GUARDED_BY(worker_task_checker_);
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SeqNumUnwrapper<uint16_t> frame_id_unwrapper_
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RTC_GUARDED_BY(worker_task_checker_);
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// Video structure provided in the dependency descriptor in a first packet
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// of a key frame. It is required to parse dependency descriptor in the
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// following delta packets.
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std::unique_ptr<FrameDependencyStructure> video_structure_
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RTC_GUARDED_BY(worker_task_checker_);
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// Frame id of the last frame with the attached video structure.
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// absl::nullopt when `video_structure_ == nullptr`;
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absl::optional<int64_t> video_structure_frame_id_
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RTC_GUARDED_BY(worker_task_checker_);
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rtc::CriticalSection reference_finder_lock_;
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std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_
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RTC_GUARDED_BY(reference_finder_lock_);
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absl::optional<VideoCodecType> current_codec_;
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uint32_t last_assembled_frame_rtp_timestamp_;
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rtc::CriticalSection last_seq_num_cs_;
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std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
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RTC_GUARDED_BY(last_seq_num_cs_);
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video_coding::H264SpsPpsTracker tracker_;
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// Maps payload id to the depacketizer.
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std::map<uint8_t, std::unique_ptr<VideoRtpDepacketizer>> payload_type_map_;
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// TODO(johan): Remove pt_codec_params_ once
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
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// Maps a payload type to a map of out-of-band supplied codec parameters.
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std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_;
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int16_t last_payload_type_ = -1;
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bool has_received_frame_;
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std::vector<RtpPacketSinkInterface*> secondary_sinks_
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RTC_GUARDED_BY(worker_task_checker_);
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// Info for GetSyncInfo is updated on network or worker thread, and queried on
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// the worker thread.
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rtc::CriticalSection sync_info_lock_;
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absl::optional<uint32_t> last_received_rtp_timestamp_
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RTC_GUARDED_BY(sync_info_lock_);
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absl::optional<int64_t> last_received_rtp_system_time_ms_
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RTC_GUARDED_BY(sync_info_lock_);
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// Used to validate the buffered frame decryptor is always run on the correct
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// thread.
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rtc::ThreadChecker network_tc_;
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// Handles incoming encrypted frames and forwards them to the
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// rtp_reference_finder if they are decryptable.
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std::unique_ptr<BufferedFrameDecryptor> buffered_frame_decryptor_
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RTC_PT_GUARDED_BY(network_tc_);
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std::atomic<bool> frames_decryptable_;
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absl::optional<ColorSpace> last_color_space_;
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AbsoluteCaptureTimeReceiver absolute_capture_time_receiver_
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RTC_GUARDED_BY(worker_task_checker_);
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int64_t last_completed_picture_id_ = 0;
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
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};
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} // namespace webrtc
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#endif // VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_
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