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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_RECEIVER_H_
#define VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_
#include <list>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/crypto/framedecryptorinterface.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.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/contributing_sources.h"
#include "modules/video_coding/h264_sps_pps_tracker.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "modules/video_coding/packet_buffer.h"
#include "modules/video_coding/rtp_frame_reference_finder.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/sequenced_task_checker.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
#include "video/buffered_frame_decryptor.h"
namespace webrtc {
class NackModule;
class PacketRouter;
class ProcessThread;
class ReceiveStatistics;
class ReceiveStatisticsProxy;
class RtcpRttStats;
class RtpPacketReceived;
class Transport;
class UlpfecReceiver;
class RtpVideoStreamReceiver : public RecoveredPacketReceiver,
public RtpPacketSinkInterface,
public VCMFrameTypeCallback,
public VCMPacketRequestCallback,
public video_coding::OnReceivedFrameCallback,
public video_coding::OnCompleteFrameCallback,
public OnDecryptedFrameCallback {
public:
RtpVideoStreamReceiver(
Transport* transport,
RtcpRttStats* rtt_stats,
PacketRouter* packet_router,
const VideoReceiveStream::Config* config,
ReceiveStatistics* rtp_receive_statistics,
ReceiveStatisticsProxy* receive_stats_proxy,
ProcessThread* process_thread,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender,
video_coding::OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
~RtpVideoStreamReceiver() override;
void AddReceiveCodec(const VideoCodec& video_codec,
const std::map<std::string, std::string>& codec_params);
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 in the packet buffer.
int GetUniqueFramesSeen() const;
// Implements RtpPacketSinkInterface.
void OnRtpPacket(const RtpPacketReceived& packet) override;
// TODO(philipel): Stop using VCMPacket in the new jitter buffer and then
// remove this function.
int32_t OnReceivedPayloadData(const uint8_t* payload_data,
size_t payload_size,
const WebRtcRTPHeader* rtp_header);
int32_t OnReceivedPayloadData(
const uint8_t* payload_data,
size_t payload_size,
const WebRtcRTPHeader* rtp_header,
const absl::optional<RtpGenericFrameDescriptor>& generic_descriptor,
bool is_recovered);
// Implements RecoveredPacketReceiver.
void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
// Implements VCMFrameTypeCallback.
int32_t RequestKeyFrame() override;
bool IsUlpfecEnabled() const;
bool IsRetransmissionsEnabled() const;
// Don't use, still experimental.
void RequestPacketRetransmit(const std::vector<uint16_t>& sequence_numbers);
// Implements VCMPacketRequestCallback.
int32_t ResendPackets(const uint16_t* sequenceNumbers,
uint16_t length) override;
// Implements OnReceivedFrameCallback.
void OnReceivedFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) override;
// Implements OnCompleteFrameCallback.
void OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> frame) override;
// Implements OnDecryptedFrameCallback.
void OnDecryptedFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) override;
// 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);
std::vector<webrtc::RtpSource> GetSources() const;
private:
// 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);
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_;
ReceiveStatistics* const rtp_receive_statistics_;
std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
rtc::SequencedTaskChecker 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_;
// Members for the new jitter buffer experiment.
video_coding::OnCompleteFrameCallback* complete_frame_callback_;
KeyFrameRequestSender* keyframe_request_sender_;
std::unique_ptr<NackModule> nack_module_;
rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_;
std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_;
rtc::CriticalSection last_seq_num_cs_;
std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
RTC_GUARDED_BY(last_seq_num_cs_);
video_coding::H264SpsPpsTracker tracker_;
std::map<uint8_t, VideoCodecType> pt_codec_type_;
// 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 GetSources and GetSyncInfo is updated on network or worker thread,
// queried on the worker thread.
rtc::CriticalSection rtp_sources_lock_;
ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_);
absl::optional<uint32_t> last_received_rtp_timestamp_
RTC_GUARDED_BY(rtp_sources_lock_);
absl::optional<int64_t> last_received_rtp_system_time_ms_
RTC_GUARDED_BY(rtp_sources_lock_);
// Used to validate the buffered frame decryptor is always run on the correct
// thread.
rtc::ThreadChecker 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_);
absl::optional<ColorSpace> last_color_space_;
};
} // namespace webrtc
#endif // VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_