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Reland "Remove legacy RtpVideoStreamReceiver."

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This reverts commit 768de78d10506ae192e5f12af48fce0c9b9e31ef.

Reason for revert: downstream cleaned up

Original change's description:
> Revert "Remove legacy RtpVideoStreamReceiver."
>
> This reverts commit da12e10aba4d12e7a6fb3882dc667901c9e17aa2.
>
> Reason for revert: Speculative revert due to failing downstream tests
>
> Original change's description:
> > Remove legacy RtpVideoStreamReceiver.
> >
> > Bug: none
> > Change-Id: I434a56980f4d6c68381abae973cd846c71441b08
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236846
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37177}
>
> Bug: none
> Change-Id: I960c7693955c061ad95f921905d35c87733a3283
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265397
> Commit-Queue: Björn Terelius <terelius@google.com>
> Owners-Override: Björn Terelius <terelius@google.com>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Björn Terelius <terelius@google.com>
> Cr-Commit-Position: refs/heads/main@{#37183}

Bug: none
Change-Id: Ia4a84c6287d5a6a24bb98174617bec2e469e194b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265840
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Björn Terelius <terelius@google.com>
Cr-Commit-Position: refs/heads/main@{#37221}
This commit is contained in:
Philip Eliasson
2022-06-15 08:42:35 +00:00
committed by WebRTC LUCI CQ
parent 9c55fa5a0e
commit eb0b93608e
4 changed files with 0 additions and 2955 deletions
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3
video/BUILD.gn
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@ -172,8 +172,6 @@ rtc_source_set("video_legacy") {
"call_stats.h",
"receive_statistics_proxy.cc",
"receive_statistics_proxy.h",
"rtp_video_stream_receiver.cc",
"rtp_video_stream_receiver.h",
"video_quality_observer.cc",
"video_quality_observer.h",
]
@ -1007,7 +1005,6 @@ if (rtc_include_tests) {
sources = [
"call_stats_unittest.cc",
"receive_statistics_proxy_unittest.cc",
"rtp_video_stream_receiver_unittest.cc",
]
deps = [
":video_legacy",

1278
video/rtp_video_stream_receiver.cc
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436
video/rtp_video_stream_receiver.h
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@ -1,436 +0,0 @@
/*
* 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_RECEIVER_H_
#define VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_
#include <atomic>
#include <list>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/sequence_checker.h"
#include "api/transport/field_trial_based_config.h"
#include "api/units/timestamp.h"
#include "api/video/color_space.h"
#include "api/video/video_codec_type.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.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
#include "modules/rtp_rtcp/source/capture_clock_offset_updater.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_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 "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread_annotations.h"
#include "video/buffered_frame_decryptor.h"
#include "video/rtp_video_stream_receiver_frame_transformer_delegate.h"
#include "video/unique_timestamp_counter.h"
namespace webrtc {
class DEPRECATED_NackModule;
class PacketRouter;
class ProcessThread;
class ReceiveStatistics;
class ReceiveStatisticsProxy;
class RtcpRttStats;
class RtpPacketReceived;
class Transport;
class UlpfecReceiver;
class RtpVideoStreamReceiver : public LossNotificationSender,
public RecoveredPacketReceiver,
public RtpPacketSinkInterface,
public KeyFrameRequestSender,
public OnDecryptedFrameCallback,
public OnDecryptionStatusChangeCallback,
public RtpVideoFrameReceiver {
public:
// A complete frame is a frame which has received all its packets and all its
// references are known.
class OnCompleteFrameCallback {
public:
virtual ~OnCompleteFrameCallback() {}
virtual void OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) = 0;
};
// DEPRECATED due to dependency on ReceiveStatisticsProxy.
RtpVideoStreamReceiver(
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 VideoReceiveStreamInterface::Config* config,
ReceiveStatistics* rtp_receive_statistics,
ReceiveStatisticsProxy* receive_stats_proxy,
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,
OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const FieldTrialsView* field_trials = nullptr);
RtpVideoStreamReceiver(
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 VideoReceiveStreamInterface::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,
OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const FieldTrialsView* field_trials = nullptr);
~RtpVideoStreamReceiver() override;
void AddReceiveCodec(uint8_t payload_type,
VideoCodecType codec_type,
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;
// 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);
void OnCompleteFrames(RtpFrameReferenceFinder::ReturnVector frames);
// Implements OnDecryptedFrameCallback.
void OnDecryptedFrame(std::unique_ptr<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 VideoReceiveStreamInterface 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<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() RTC_LOCKS_EXCLUDED(mutex_) override;
// NackSender implementation.
void SendNack(const std::vector<uint16_t>& sequence_numbers,
bool buffering_allowed) RTC_LOCKS_EXCLUDED(mutex_) override;
// LossNotificationSender implementation.
void SendLossNotification(uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed)
RTC_LOCKS_EXCLUDED(mutex_) override;
// Send all RTCP feedback messages buffered thus far.
void SendBufferedRtcpFeedback() RTC_LOCKS_EXCLUDED(mutex_);
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;
};
struct ConsumedRtcpFeedback {
bool request_key_frame = false;
std::vector<uint16_t> nack_sequence_numbers;
absl::optional<LossNotificationState> lntf_state;
};
ConsumedRtcpFeedback ConsumeRtcpFeedback() RTC_LOCKS_EXCLUDED(mutex_);
ConsumedRtcpFeedback ConsumeRtcpFeedbackLocked()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// This method is called both with and without mutex_ held.
void SendRtcpFeedback(ConsumedRtcpFeedback feedback);
KeyFrameRequestSender* const key_frame_request_sender_;
NackSender* const nack_sender_;
LossNotificationSender* const loss_notification_sender_;
// NACKs are accessible from two threads due to nack_module_ being a module.
Mutex mutex_;
// Key-frame-request-related state.
bool request_key_frame_ RTC_GUARDED_BY(mutex_);
// NACK-related state.
std::vector<uint16_t> nack_sequence_numbers_ RTC_GUARDED_BY(mutex_);
absl::optional<LossNotificationState> lntf_state_ RTC_GUARDED_BY(mutex_);
};
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<RtpFrameObject> frame)
RTC_LOCKS_EXCLUDED(packet_buffer_lock_);
void UpdatePacketReceiveTimestamps(const RtpPacketReceived& packet,
bool is_keyframe)
RTC_RUN_ON(worker_task_checker_);
const FieldTrialsView& field_trials_;
FieldTrialBasedConfig owned_field_trials_;
Clock* const clock_;
// Ownership of this object lies with VideoReceiveStreamInterface, which owns
// `this`.
const VideoReceiveStreamInterface::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_;
RTC_NO_UNIQUE_ADDRESS 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<RtpRtcp> rtp_rtcp_;
OnCompleteFrameCallback* complete_frame_callback_;
KeyFrameRequestSender* const keyframe_request_sender_;
RtcpFeedbackBuffer rtcp_feedback_buffer_;
std::unique_ptr<DEPRECATED_NackModule> nack_module_;
std::unique_ptr<LossNotificationController> loss_notification_controller_;
mutable Mutex packet_buffer_lock_;
video_coding::PacketBuffer packet_buffer_ RTC_GUARDED_BY(packet_buffer_lock_);
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_);
Mutex reference_finder_lock_;
std::unique_ptr<RtpFrameReferenceFinder> reference_finder_
RTC_GUARDED_BY(reference_finder_lock_);
absl::optional<VideoCodecType> current_codec_;
uint32_t last_assembled_frame_rtp_timestamp_;
Mutex last_seq_num_mutex_;
std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
RTC_GUARDED_BY(last_seq_num_mutex_);
video_coding::H264SpsPpsTracker tracker_;
// Maps payload id to the depacketizer.
std::map<uint8_t, std::unique_ptr<VideoRtpDepacketizer>> payload_type_map_;
// 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_
RTC_GUARDED_BY(worker_task_checker_);
// Info for GetSyncInfo is updated on network or worker thread, and queried on
// the worker thread.
mutable Mutex sync_info_lock_;
absl::optional<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(sync_info_lock_);
absl::optional<uint32_t> last_received_keyframe_rtp_timestamp_
RTC_GUARDED_BY(sync_info_lock_);
absl::optional<Timestamp> last_received_rtp_system_time_
RTC_GUARDED_BY(sync_info_lock_);
absl::optional<Timestamp> last_received_keyframe_rtp_system_time_
RTC_GUARDED_BY(sync_info_lock_);
// Used to validate the buffered frame decryptor is always run on the correct
// thread.
SequenceChecker network_tc_;
// 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(network_tc_);
std::atomic<bool> frames_decryptable_;
absl::optional<ColorSpace> last_color_space_;
AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_
RTC_GUARDED_BY(worker_task_checker_);
CaptureClockOffsetUpdater capture_clock_offset_updater_
RTC_GUARDED_BY(worker_task_checker_);
int64_t last_completed_picture_id_ = 0;
rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate>
frame_transformer_delegate_;
SeqNumUnwrapper<uint16_t> rtp_seq_num_unwrapper_
RTC_GUARDED_BY(packet_buffer_lock_);
std::map<int64_t, RtpPacketInfo> packet_infos_
RTC_GUARDED_BY(packet_buffer_lock_);
};
} // namespace webrtc
#endif // VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_

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video/rtp_video_stream_receiver_unittest.cc
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