Fork NackModule and RtpVideoStreamReceiver
Bug: webrtc:11595 Change-Id: I4d14c0bf9c32e09d1624099a256f2778afebd4df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175901 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31337}
This commit is contained in:
Tommi
386
video/rtp_video_stream_receiver2.h
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386
video/rtp_video_stream_receiver2.h
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/*
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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_RECEIVER2_H_
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#define VIDEO_RTP_VIDEO_STREAM_RECEIVER2_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/experiments/field_trial_parser.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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#include "video/rtp_video_stream_receiver_frame_transformer_delegate.h"
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namespace webrtc {
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class NackModule2;
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class PacketRouter;
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class ProcessThread;
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class ReceiveStatistics;
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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 RtpVideoStreamReceiver2 : 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 RtpVideoFrameReceiver {
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public:
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RtpVideoStreamReceiver2(
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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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RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
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RtcpCnameCallback* rtcp_cname_callback,
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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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~RtpVideoStreamReceiver2() 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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// Sets a frame transformer after a stream has started, if no transformer
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// has previously been set. Does not reset the decoder state.
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void SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
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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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// Implements RtpVideoFrameReceiver.
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void ManageFrame(
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std::unique_ptr<video_coding::RtpFrameObject> frame) override;
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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() RTC_LOCKS_EXCLUDED(cs_) 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) RTC_LOCKS_EXCLUDED(cs_) 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)
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RTC_LOCKS_EXCLUDED(cs_) override;
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// Send all RTCP feedback messages buffered thus far.
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void SendBufferedRtcpFeedback() RTC_LOCKS_EXCLUDED(cs_);
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private:
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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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struct ConsumedRtcpFeedback {
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bool request_key_frame = false;
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std::vector<uint16_t> nack_sequence_numbers;
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absl::optional<LossNotificationState> lntf_state;
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};
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ConsumedRtcpFeedback ConsumeRtcpFeedback() RTC_LOCKS_EXCLUDED(cs_);
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ConsumedRtcpFeedback ConsumeRtcpFeedbackLocked()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(cs_);
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// This method is called both with and without cs_ held.
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void SendRtcpFeedback(ConsumedRtcpFeedback feedback);
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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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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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// Set by the field trial WebRTC-ForcePlayoutDelay to override any playout
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// delay that is specified in the received packets.
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FieldTrialOptional<int> forced_playout_delay_max_ms_;
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FieldTrialOptional<int> forced_playout_delay_min_ms_;
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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<NackModule2> 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<RtpVideoStreamReceiverFrameTransformerDelegate>
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frame_transformer_delegate_;
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};
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} // namespace webrtc
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#endif // VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_
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