webrtc_m130/video/rtp_video_stream_receiver2.h
Tomas Gunnarsson f25761d798 Remove dependency from RtpRtcp on the Module interface.
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.

Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.

The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.

Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
2020-06-04 08:11:21 +00:00

368 lines
15 KiB
C++

/*
* 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_