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aosp12/external/webrtc/video/rtp_video_stream_receiver2.h

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/*
* 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/video/color_space.h"
#include "api/video_codecs/video_codec.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/remote_ntp_time_estimator.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_receiver.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/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/packet_buffer.h"
#include "modules/video_coding/rtp_frame_reference_finder.h"
#include "modules/video_coding/unique_timestamp_counter.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/synchronization/sequence_checker.h"
#include "rtc_base/thread_annotations.h"
#include "video/buffered_frame_decryptor.h"
#include "video/rtp_video_stream_receiver_frame_transformer_delegate.h"
namespace webrtc {
class NackModule2;
class PacketRouter;
class ProcessThread;
class ReceiveStatistics;
class RtcpRttStats;
class RtpPacketReceived;
class Transport;
class UlpfecReceiver;
class RtpVideoStreamReceiver2 : public LossNotificationSender,
public RecoveredPacketReceiver,
public RtpPacketSinkInterface,
public KeyFrameRequestSender,
public video_coding::OnCompleteFrameCallback,
public OnDecryptedFrameCallback,
public OnDecryptionStatusChangeCallback,
public RtpVideoFrameReceiver {
public:
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 VideoReceiveStream::Config* config,
ReceiveStatistics* rtp_receive_statistics,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
ProcessThread* process_thread,
NackSender* nack_sender,
// The KeyFrameRequestSender is optional; if not provided, key frame
// requests are sent via the internal RtpRtcp module.
KeyFrameRequestSender* keyframe_request_sender,
video_coding::OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
~RtpVideoStreamReceiver2() override;
void AddReceiveCodec(const VideoCodec& video_codec,
const std::map<std::string, std::string>& codec_params,
bool raw_payload);
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(&worker_task_checker_);
return frame_counter_.GetUniqueSeen();
}
// Implements RtpPacketSinkInterface.
void OnRtpPacket(const RtpPacketReceived& packet) override;
// TODO(philipel): Stop using VCMPacket in the new jitter buffer and then
// remove this function. Public only for tests.
void OnReceivedPayloadData(rtc::CopyOnWriteBuffer codec_payload,
const RtpPacketReceived& rtp_packet,
const RTPVideoHeader& video);
// Implements RecoveredPacketReceiver.
void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
// Send an RTCP keyframe request.
void RequestKeyFrame() override;
// Implements LossNotificationSender.
void SendLossNotification(uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed) override;
bool IsUlpfecEnabled() const;
bool IsRetransmissionsEnabled() const;
// 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;
// Don't use, still experimental.
void RequestPacketRetransmit(const std::vector<uint16_t>& sequence_numbers);
// Implements OnCompleteFrameCallback.
void OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> frame) override;
// Implements OnDecryptedFrameCallback.
void OnDecryptedFrame(
std::unique_ptr<video_coding::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 VideoReceiveStream when stats are updated.
void UpdateRtt(int64_t max_rtt_ms);
absl::optional<int64_t> LastReceivedPacketMs() const;
absl::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:
// Implements RtpVideoFrameReceiver.
void ManageFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) override;
// 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();
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;
};
SequenceChecker worker_task_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(worker_task_checker_);
// NACK-related state.
std::vector<uint16_t> nack_sequence_numbers_
RTC_GUARDED_BY(worker_task_checker_);
absl::optional<LossNotificationState> lntf_state_
RTC_GUARDED_BY(worker_task_checker_);
};
enum ParseGenericDependenciesResult {
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);
// Parses and handles RED headers.
// This function assumes that it's being called from only one thread.
void ParseAndHandleEncapsulatingHeader(const RtpPacketReceived& packet);
void NotifyReceiverOfEmptyPacket(uint16_t seq_num);
void UpdateHistograms();
bool IsRedEnabled() const;
void InsertSpsPpsIntoTracker(uint8_t payload_type);
void OnInsertedPacket(video_coding::PacketBuffer::InsertResult result);
ParseGenericDependenciesResult ParseGenericDependenciesExtension(
const RtpPacketReceived& rtp_packet,
RTPVideoHeader* video_header) RTC_RUN_ON(worker_task_checker_);
void OnAssembledFrame(std::unique_ptr<video_coding::RtpFrameObject> frame);
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_;
RtpHeaderExtensionMap rtp_header_extensions_;
// 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_;
SequenceChecker worker_task_checker_;
bool receiving_ RTC_GUARDED_BY(worker_task_checker_);
int64_t last_packet_log_ms_ RTC_GUARDED_BY(worker_task_checker_);
const std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
video_coding::OnCompleteFrameCallback* complete_frame_callback_;
KeyFrameRequestSender* const keyframe_request_sender_;
RtcpFeedbackBuffer rtcp_feedback_buffer_;
const std::unique_ptr<NackModule2> nack_module_;
std::unique_ptr<LossNotificationController> loss_notification_controller_;
video_coding::PacketBuffer packet_buffer_;
UniqueTimestampCounter frame_counter_ RTC_GUARDED_BY(worker_task_checker_);
SeqNumUnwrapper<uint16_t> frame_id_unwrapper_
RTC_GUARDED_BY(worker_task_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(worker_task_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(worker_task_checker_);
std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_
RTC_GUARDED_BY(worker_task_checker_);
absl::optional<VideoCodecType> current_codec_
RTC_GUARDED_BY(worker_task_checker_);
uint32_t last_assembled_frame_rtp_timestamp_
RTC_GUARDED_BY(worker_task_checker_);
std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
RTC_GUARDED_BY(worker_task_checker_);
video_coding::H264SpsPpsTracker tracker_ RTC_GUARDED_BY(worker_task_checker_);
// Maps payload id to the depacketizer.
std::map<uint8_t, std::unique_ptr<VideoRtpDepacketizer>> payload_type_map_
RTC_GUARDED_BY(worker_task_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, std::map<std::string, std::string>> pt_codec_params_
RTC_GUARDED_BY(worker_task_checker_);
int16_t last_payload_type_ RTC_GUARDED_BY(worker_task_checker_) = -1;
bool has_received_frame_ RTC_GUARDED_BY(worker_task_checker_);
std::vector<RtpPacketSinkInterface*> secondary_sinks_
RTC_GUARDED_BY(worker_task_checker_);
absl::optional<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(worker_task_checker_);
absl::optional<int64_t> last_received_rtp_system_time_ms_
RTC_GUARDED_BY(worker_task_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(worker_task_checker_);
bool frames_decryptable_ RTC_GUARDED_BY(worker_task_checker_);
absl::optional<ColorSpace> last_color_space_;
AbsoluteCaptureTimeReceiver absolute_capture_time_receiver_
RTC_GUARDED_BY(worker_task_checker_);
int64_t last_completed_picture_id_ = 0;
rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate>
frame_transformer_delegate_;
};
} // namespace webrtc
#endif // VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_
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