| Index: webrtc/video/rtp_stream_receiver.cc
|
| diff --git a/webrtc/video/rtp_stream_receiver.cc b/webrtc/video/rtp_stream_receiver.cc
|
| deleted file mode 100644
|
| index 78057771d6e5705b65e37eebdbba59ef24fa8c88..0000000000000000000000000000000000000000
|
| --- a/webrtc/video/rtp_stream_receiver.cc
|
| +++ /dev/null
|
| @@ -1,683 +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.
|
| - */
|
| -
|
| -#include "webrtc/video/rtp_stream_receiver.h"
|
| -
|
| -#include <vector>
|
| -#include <utility>
|
| -
|
| -#include "webrtc/base/checks.h"
|
| -#include "webrtc/base/location.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/common_types.h"
|
| -#include "webrtc/config.h"
|
| -#include "webrtc/media/base/mediaconstants.h"
|
| -#include "webrtc/modules/pacing/packet_router.h"
|
| -#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h"
|
| -#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
|
| -#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
|
| -#include "webrtc/modules/video_coding/frame_object.h"
|
| -#include "webrtc/modules/video_coding/h264_sprop_parameter_sets.h"
|
| -#include "webrtc/modules/video_coding/h264_sps_pps_tracker.h"
|
| -#include "webrtc/modules/video_coding/packet_buffer.h"
|
| -#include "webrtc/modules/video_coding/video_coding_impl.h"
|
| -#include "webrtc/system_wrappers/include/field_trial.h"
|
| -#include "webrtc/system_wrappers/include/metrics.h"
|
| -#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
|
| -#include "webrtc/video/receive_statistics_proxy.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -namespace {
|
| -constexpr int kPacketBufferStartSize = 32;
|
| -constexpr int kPacketBufferMaxSixe = 2048;
|
| -}
|
| -
|
| -std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
|
| - ReceiveStatistics* receive_statistics,
|
| - Transport* outgoing_transport,
|
| - RtcpRttStats* rtt_stats,
|
| - RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
|
| - TransportSequenceNumberAllocator* transport_sequence_number_allocator) {
|
| - RtpRtcp::Configuration configuration;
|
| - configuration.audio = false;
|
| - configuration.receiver_only = true;
|
| - configuration.receive_statistics = receive_statistics;
|
| - configuration.outgoing_transport = outgoing_transport;
|
| - configuration.intra_frame_callback = nullptr;
|
| - configuration.rtt_stats = rtt_stats;
|
| - configuration.rtcp_packet_type_counter_observer =
|
| - rtcp_packet_type_counter_observer;
|
| - configuration.transport_sequence_number_allocator =
|
| - transport_sequence_number_allocator;
|
| - configuration.send_bitrate_observer = nullptr;
|
| - configuration.send_frame_count_observer = nullptr;
|
| - configuration.send_side_delay_observer = nullptr;
|
| - configuration.send_packet_observer = nullptr;
|
| - configuration.bandwidth_callback = nullptr;
|
| - configuration.transport_feedback_callback = nullptr;
|
| -
|
| - std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
|
| - rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
|
| -
|
| - return rtp_rtcp;
|
| -}
|
| -
|
| -static const int kPacketLogIntervalMs = 10000;
|
| -
|
| -RtpStreamReceiver::RtpStreamReceiver(
|
| - Transport* transport,
|
| - RtcpRttStats* rtt_stats,
|
| - PacketRouter* packet_router,
|
| - const VideoReceiveStream::Config* config,
|
| - ReceiveStatisticsProxy* receive_stats_proxy,
|
| - ProcessThread* process_thread,
|
| - NackSender* nack_sender,
|
| - KeyFrameRequestSender* keyframe_request_sender,
|
| - video_coding::OnCompleteFrameCallback* complete_frame_callback,
|
| - VCMTiming* timing)
|
| - : clock_(Clock::GetRealTimeClock()),
|
| - config_(*config),
|
| - packet_router_(packet_router),
|
| - process_thread_(process_thread),
|
| - ntp_estimator_(clock_),
|
| - rtp_header_parser_(RtpHeaderParser::Create()),
|
| - rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
|
| - this,
|
| - this,
|
| - &rtp_payload_registry_)),
|
| - rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
|
| - ulpfec_receiver_(UlpfecReceiver::Create(this)),
|
| - receiving_(false),
|
| - restored_packet_in_use_(false),
|
| - last_packet_log_ms_(-1),
|
| - rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
|
| - transport,
|
| - rtt_stats,
|
| - receive_stats_proxy,
|
| - packet_router)),
|
| - complete_frame_callback_(complete_frame_callback),
|
| - keyframe_request_sender_(keyframe_request_sender),
|
| - timing_(timing),
|
| - has_received_frame_(false) {
|
| - packet_router_->AddReceiveRtpModule(rtp_rtcp_.get());
|
| - rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy);
|
| - rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
|
| -
|
| - RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
|
| - << "A stream should not be configured with RTCP disabled. This value is "
|
| - "reserved for internal usage.";
|
| - RTC_DCHECK(config_.rtp.remote_ssrc != 0);
|
| - // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
|
| - RTC_DCHECK(config_.rtp.local_ssrc != 0);
|
| - RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
|
| -
|
| - rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
|
| - rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
|
| - rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
|
| -
|
| - for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
|
| - EnableReceiveRtpHeaderExtension(config_.rtp.extensions[i].uri,
|
| - config_.rtp.extensions[i].id);
|
| - }
|
| -
|
| - static const int kMaxPacketAgeToNack = 450;
|
| - const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
|
| - ? kMaxPacketAgeToNack
|
| - : kDefaultMaxReorderingThreshold;
|
| - rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
|
| -
|
| - if (config_.rtp.rtx_ssrc) {
|
| - rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc);
|
| -
|
| - for (const auto& kv : config_.rtp.rtx_payload_types) {
|
| - RTC_DCHECK(kv.second != 0);
|
| - rtp_payload_registry_.SetRtxPayloadType(kv.second, kv.first);
|
| - }
|
| - }
|
| -
|
| - if (IsUlpfecEnabled()) {
|
| - VideoCodec ulpfec_codec = {};
|
| - ulpfec_codec.codecType = kVideoCodecULPFEC;
|
| - strncpy(ulpfec_codec.plName, "ulpfec", sizeof(ulpfec_codec.plName));
|
| - ulpfec_codec.plType = config_.rtp.ulpfec.ulpfec_payload_type;
|
| - RTC_CHECK(AddReceiveCodec(ulpfec_codec));
|
| - }
|
| -
|
| - if (IsRedEnabled()) {
|
| - VideoCodec red_codec = {};
|
| - red_codec.codecType = kVideoCodecRED;
|
| - strncpy(red_codec.plName, "red", sizeof(red_codec.plName));
|
| - red_codec.plType = config_.rtp.ulpfec.red_payload_type;
|
| - RTC_CHECK(AddReceiveCodec(red_codec));
|
| - if (config_.rtp.ulpfec.red_rtx_payload_type != -1) {
|
| - rtp_payload_registry_.SetRtxPayloadType(
|
| - config_.rtp.ulpfec.red_rtx_payload_type,
|
| - config_.rtp.ulpfec.red_payload_type);
|
| - }
|
| - }
|
| -
|
| - if (config_.rtp.rtcp_xr.receiver_reference_time_report)
|
| - rtp_rtcp_->SetRtcpXrRrtrStatus(true);
|
| -
|
| - // Stats callback for CNAME changes.
|
| - rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
|
| -
|
| - process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
|
| -
|
| - if (config_.rtp.nack.rtp_history_ms != 0) {
|
| - nack_module_.reset(
|
| - new NackModule(clock_, nack_sender, keyframe_request_sender));
|
| - process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE);
|
| - }
|
| -
|
| - packet_buffer_ = video_coding::PacketBuffer::Create(
|
| - clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this);
|
| - reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this));
|
| -}
|
| -
|
| -RtpStreamReceiver::~RtpStreamReceiver() {
|
| - if (nack_module_) {
|
| - process_thread_->DeRegisterModule(nack_module_.get());
|
| - }
|
| -
|
| - process_thread_->DeRegisterModule(rtp_rtcp_.get());
|
| -
|
| - packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
|
| - UpdateHistograms();
|
| -}
|
| -
|
| -bool RtpStreamReceiver::AddReceiveCodec(
|
| - const VideoCodec& video_codec,
|
| - const std::map<std::string, std::string>& codec_params) {
|
| - pt_codec_params_.insert(make_pair(video_codec.plType, codec_params));
|
| - return AddReceiveCodec(video_codec);
|
| -}
|
| -
|
| -bool RtpStreamReceiver::AddReceiveCodec(const VideoCodec& video_codec) {
|
| - int8_t old_pltype = -1;
|
| - if (rtp_payload_registry_.ReceivePayloadType(video_codec, &old_pltype) !=
|
| - -1) {
|
| - rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
|
| - }
|
| - return rtp_payload_registry_.RegisterReceivePayload(video_codec) == 0;
|
| -}
|
| -
|
| -uint32_t RtpStreamReceiver::GetRemoteSsrc() const {
|
| - return rtp_receiver_->SSRC();
|
| -}
|
| -
|
| -int RtpStreamReceiver::GetCsrcs(uint32_t* csrcs) const {
|
| - return rtp_receiver_->CSRCs(csrcs);
|
| -}
|
| -
|
| -RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const {
|
| - return rtp_receiver_.get();
|
| -}
|
| -
|
| -int32_t RtpStreamReceiver::OnReceivedPayloadData(
|
| - const uint8_t* payload_data,
|
| - size_t payload_size,
|
| - const WebRtcRTPHeader* rtp_header) {
|
| - WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
|
| - rtp_header_with_ntp.ntp_time_ms =
|
| - ntp_estimator_.Estimate(rtp_header->header.timestamp);
|
| - VCMPacket packet(payload_data, payload_size, rtp_header_with_ntp);
|
| - packet.timesNacked =
|
| - nack_module_ ? nack_module_->OnReceivedPacket(packet) : -1;
|
| -
|
| - // In the case of a video stream without picture ids and no rtx the
|
| - // RtpFrameReferenceFinder will need to know about padding to
|
| - // correctly calculate frame references.
|
| - if (packet.sizeBytes == 0) {
|
| - reference_finder_->PaddingReceived(packet.seqNum);
|
| - return 0;
|
| - }
|
| -
|
| - if (packet.codec == kVideoCodecH264) {
|
| - // Only when we start to receive packets will we know what payload type
|
| - // that will be used. When we know the payload type insert the correct
|
| - // sps/pps into the tracker.
|
| - if (packet.payloadType != last_payload_type_) {
|
| - last_payload_type_ = packet.payloadType;
|
| - InsertSpsPpsIntoTracker(packet.payloadType);
|
| - }
|
| -
|
| - switch (tracker_.CopyAndFixBitstream(&packet)) {
|
| - case video_coding::H264SpsPpsTracker::kRequestKeyframe:
|
| - keyframe_request_sender_->RequestKeyFrame();
|
| - FALLTHROUGH();
|
| - case video_coding::H264SpsPpsTracker::kDrop:
|
| - return 0;
|
| - case video_coding::H264SpsPpsTracker::kInsert:
|
| - break;
|
| - }
|
| -
|
| - } else {
|
| - uint8_t* data = new uint8_t[packet.sizeBytes];
|
| - memcpy(data, packet.dataPtr, packet.sizeBytes);
|
| - packet.dataPtr = data;
|
| - }
|
| -
|
| - packet_buffer_->InsertPacket(&packet);
|
| - return 0;
|
| -}
|
| -
|
| -// TODO(nisse): Try to delete this method. Obstacles: It is used by
|
| -// ParseAndHandleEncapsulatingHeader, for handling Rtx packets, and
|
| -// for callbacks from |ulpfec_receiver_|.
|
| -void RtpStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
|
| - size_t rtp_packet_length) {
|
| - RTPHeader header;
|
| - if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
|
| - return;
|
| - }
|
| - header.payload_type_frequency = kVideoPayloadTypeFrequency;
|
| - bool in_order = IsPacketInOrder(header);
|
| - ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
|
| -}
|
| -
|
| -// TODO(pbos): Remove as soon as audio can handle a changing payload type
|
| -// without this callback.
|
| -int32_t RtpStreamReceiver::OnInitializeDecoder(
|
| - const int8_t payload_type,
|
| - const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
| - const int frequency,
|
| - const size_t channels,
|
| - const uint32_t rate) {
|
| - RTC_NOTREACHED();
|
| - return 0;
|
| -}
|
| -
|
| -void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) {
|
| - rtp_rtcp_->SetRemoteSSRC(ssrc);
|
| -}
|
| -
|
| -// This method handles both regular RTP packets and packets recovered
|
| -// via FlexFEC.
|
| -void RtpStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
|
| - {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - if (!receiving_) {
|
| - return;
|
| - }
|
| -
|
| - if (!packet.recovered()) {
|
| - int64_t now_ms = clock_->TimeInMilliseconds();
|
| -
|
| - // Periodically log the RTP header of incoming packets.
|
| - if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
|
| - std::stringstream ss;
|
| - ss << "Packet received on SSRC: " << packet.Ssrc()
|
| - << " with payload type: " << static_cast<int>(packet.PayloadType())
|
| - << ", timestamp: " << packet.Timestamp()
|
| - << ", sequence number: " << packet.SequenceNumber()
|
| - << ", arrival time: " << packet.arrival_time_ms();
|
| - int32_t time_offset;
|
| - if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
|
| - ss << ", toffset: " << time_offset;
|
| - }
|
| - uint32_t send_time;
|
| - if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
|
| - ss << ", abs send time: " << send_time;
|
| - }
|
| - LOG(LS_INFO) << ss.str();
|
| - last_packet_log_ms_ = now_ms;
|
| - }
|
| - }
|
| - }
|
| -
|
| - // TODO(nisse): Delete use of GetHeader, but needs refactoring of
|
| - // ReceivePacket and IncomingPacket methods below.
|
| - RTPHeader header;
|
| - packet.GetHeader(&header);
|
| -
|
| - header.payload_type_frequency = kVideoPayloadTypeFrequency;
|
| -
|
| - bool in_order = IsPacketInOrder(header);
|
| - if (!packet.recovered()) {
|
| - // TODO(nisse): Why isn't this done for recovered packets?
|
| - rtp_payload_registry_.SetIncomingPayloadType(header);
|
| - }
|
| - ReceivePacket(packet.data(), packet.size(), header, in_order);
|
| - // Update receive statistics after ReceivePacket.
|
| - // Receive statistics will be reset if the payload type changes (make sure
|
| - // that the first packet is included in the stats).
|
| - if (!packet.recovered()) {
|
| - // TODO(nisse): We should pass a recovered flag to stats, to aid
|
| - // fixing bug bugs.webrtc.org/6339.
|
| - rtp_receive_statistics_->IncomingPacket(
|
| - header, packet.size(), IsPacketRetransmitted(header, in_order));
|
| - }
|
| -}
|
| -
|
| -int32_t RtpStreamReceiver::RequestKeyFrame() {
|
| - return rtp_rtcp_->RequestKeyFrame();
|
| -}
|
| -
|
| -bool RtpStreamReceiver::IsUlpfecEnabled() const {
|
| - return config_.rtp.ulpfec.ulpfec_payload_type != -1;
|
| -}
|
| -
|
| -bool RtpStreamReceiver::IsRedEnabled() const {
|
| - return config_.rtp.ulpfec.red_payload_type != -1;
|
| -}
|
| -
|
| -bool RtpStreamReceiver::IsRetransmissionsEnabled() const {
|
| - return config_.rtp.nack.rtp_history_ms > 0;
|
| -}
|
| -
|
| -void RtpStreamReceiver::RequestPacketRetransmit(
|
| - const std::vector<uint16_t>& sequence_numbers) {
|
| - rtp_rtcp_->SendNack(sequence_numbers);
|
| -}
|
| -
|
| -int32_t RtpStreamReceiver::ResendPackets(const uint16_t* sequence_numbers,
|
| - uint16_t length) {
|
| - return rtp_rtcp_->SendNACK(sequence_numbers, length);
|
| -}
|
| -
|
| -void RtpStreamReceiver::OnReceivedFrame(
|
| - std::unique_ptr<video_coding::RtpFrameObject> frame) {
|
| -
|
| - if (!has_received_frame_) {
|
| - has_received_frame_ = true;
|
| - if (frame->FrameType() != kVideoFrameKey)
|
| - keyframe_request_sender_->RequestKeyFrame();
|
| - }
|
| -
|
| - if (!frame->delayed_by_retransmission())
|
| - timing_->IncomingTimestamp(frame->timestamp, clock_->TimeInMilliseconds());
|
| - reference_finder_->ManageFrame(std::move(frame));
|
| -}
|
| -
|
| -void RtpStreamReceiver::OnCompleteFrame(
|
| - std::unique_ptr<video_coding::FrameObject> frame) {
|
| - {
|
| - rtc::CritScope lock(&last_seq_num_cs_);
|
| - video_coding::RtpFrameObject* rtp_frame =
|
| - static_cast<video_coding::RtpFrameObject*>(frame.get());
|
| - last_seq_num_for_pic_id_[rtp_frame->picture_id] = rtp_frame->last_seq_num();
|
| - }
|
| - complete_frame_callback_->OnCompleteFrame(std::move(frame));
|
| -}
|
| -
|
| -void RtpStreamReceiver::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
|
| - if (nack_module_)
|
| - nack_module_->UpdateRtt(max_rtt_ms);
|
| -}
|
| -
|
| -rtc::Optional<int64_t> RtpStreamReceiver::LastReceivedPacketMs() const {
|
| - return packet_buffer_->LastReceivedPacketMs();
|
| -}
|
| -
|
| -rtc::Optional<int64_t> RtpStreamReceiver::LastReceivedKeyframePacketMs() const {
|
| - return packet_buffer_->LastReceivedKeyframePacketMs();
|
| -}
|
| -
|
| -void RtpStreamReceiver::ReceivePacket(const uint8_t* packet,
|
| - size_t packet_length,
|
| - const RTPHeader& header,
|
| - bool in_order) {
|
| - if (rtp_payload_registry_.IsEncapsulated(header)) {
|
| - ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
|
| - return;
|
| - }
|
| - const uint8_t* payload = packet + header.headerLength;
|
| - assert(packet_length >= header.headerLength);
|
| - size_t payload_length = packet_length - header.headerLength;
|
| - PayloadUnion payload_specific;
|
| - if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
|
| - &payload_specific)) {
|
| - return;
|
| - }
|
| - rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
|
| - payload_specific, in_order);
|
| -}
|
| -
|
| -void RtpStreamReceiver::ParseAndHandleEncapsulatingHeader(
|
| - const uint8_t* packet, size_t packet_length, const RTPHeader& header) {
|
| - if (rtp_payload_registry_.IsRed(header)) {
|
| - int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
|
| - if (packet[header.headerLength] == ulpfec_pt) {
|
| - rtp_receive_statistics_->FecPacketReceived(header, packet_length);
|
| - // Notify video_receiver about received FEC packets to avoid NACKing these
|
| - // packets.
|
| - NotifyReceiverOfFecPacket(header);
|
| - }
|
| - if (ulpfec_receiver_->AddReceivedRedPacket(header, packet, packet_length,
|
| - ulpfec_pt) != 0) {
|
| - return;
|
| - }
|
| - ulpfec_receiver_->ProcessReceivedFec();
|
| - } else if (rtp_payload_registry_.IsRtx(header)) {
|
| - if (header.headerLength + header.paddingLength == packet_length) {
|
| - // This is an empty packet and should be silently dropped before trying to
|
| - // parse the RTX header.
|
| - return;
|
| - }
|
| - // Remove the RTX header and parse the original RTP header.
|
| - if (packet_length < header.headerLength)
|
| - return;
|
| - if (packet_length > sizeof(restored_packet_))
|
| - return;
|
| - rtc::CritScope lock(&receive_cs_);
|
| - if (restored_packet_in_use_) {
|
| - LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
|
| - return;
|
| - }
|
| - if (!rtp_payload_registry_.RestoreOriginalPacket(
|
| - restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
|
| - header)) {
|
| - LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
|
| - << header.ssrc << " payload type: "
|
| - << static_cast<int>(header.payloadType);
|
| - return;
|
| - }
|
| - restored_packet_in_use_ = true;
|
| - OnRecoveredPacket(restored_packet_, packet_length);
|
| - restored_packet_in_use_ = false;
|
| - }
|
| -}
|
| -
|
| -void RtpStreamReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
|
| - int8_t last_media_payload_type =
|
| - rtp_payload_registry_.last_received_media_payload_type();
|
| - if (last_media_payload_type < 0) {
|
| - LOG(LS_WARNING) << "Failed to get last media payload type.";
|
| - return;
|
| - }
|
| - // Fake an empty media packet.
|
| - WebRtcRTPHeader rtp_header = {};
|
| - rtp_header.header = header;
|
| - rtp_header.header.payloadType = last_media_payload_type;
|
| - rtp_header.header.paddingLength = 0;
|
| - PayloadUnion payload_specific;
|
| - if (!rtp_payload_registry_.GetPayloadSpecifics(last_media_payload_type,
|
| - &payload_specific)) {
|
| - LOG(LS_WARNING) << "Failed to get payload specifics.";
|
| - return;
|
| - }
|
| - rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
|
| - rtp_header.type.Video.rotation = kVideoRotation_0;
|
| - if (header.extension.hasVideoRotation) {
|
| - rtp_header.type.Video.rotation = header.extension.videoRotation;
|
| - }
|
| - rtp_header.type.Video.content_type = VideoContentType::UNSPECIFIED;
|
| - if (header.extension.hasVideoContentType) {
|
| - rtp_header.type.Video.content_type = header.extension.videoContentType;
|
| - }
|
| - rtp_header.type.Video.playout_delay = header.extension.playout_delay;
|
| -
|
| - OnReceivedPayloadData(nullptr, 0, &rtp_header);
|
| -}
|
| -
|
| -bool RtpStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
|
| - size_t rtcp_packet_length) {
|
| - {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - if (!receiving_) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
|
| -
|
| - int64_t rtt = 0;
|
| - rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
|
| - if (rtt == 0) {
|
| - // Waiting for valid rtt.
|
| - return true;
|
| - }
|
| - uint32_t ntp_secs = 0;
|
| - uint32_t ntp_frac = 0;
|
| - uint32_t rtp_timestamp = 0;
|
| - if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
|
| - &rtp_timestamp) != 0) {
|
| - // Waiting for RTCP.
|
| - return true;
|
| - }
|
| - ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
|
| -
|
| - return true;
|
| -}
|
| -
|
| -void RtpStreamReceiver::FrameContinuous(uint16_t picture_id) {
|
| - if (!nack_module_)
|
| - return;
|
| -
|
| - int seq_num = -1;
|
| - {
|
| - rtc::CritScope lock(&last_seq_num_cs_);
|
| - auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
|
| - if (seq_num_it != last_seq_num_for_pic_id_.end())
|
| - seq_num = seq_num_it->second;
|
| - }
|
| - if (seq_num != -1)
|
| - nack_module_->ClearUpTo(seq_num);
|
| -}
|
| -
|
| -void RtpStreamReceiver::FrameDecoded(uint16_t picture_id) {
|
| - int seq_num = -1;
|
| - {
|
| - rtc::CritScope lock(&last_seq_num_cs_);
|
| - auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
|
| - if (seq_num_it != last_seq_num_for_pic_id_.end()) {
|
| - seq_num = seq_num_it->second;
|
| - last_seq_num_for_pic_id_.erase(last_seq_num_for_pic_id_.begin(),
|
| - ++seq_num_it);
|
| - }
|
| - }
|
| - if (seq_num != -1) {
|
| - packet_buffer_->ClearTo(seq_num);
|
| - reference_finder_->ClearTo(seq_num);
|
| - }
|
| -}
|
| -
|
| -void RtpStreamReceiver::SignalNetworkState(NetworkState state) {
|
| - rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
|
| - : RtcpMode::kOff);
|
| -}
|
| -
|
| -void RtpStreamReceiver::StartReceive() {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - receiving_ = true;
|
| -}
|
| -
|
| -void RtpStreamReceiver::StopReceive() {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - receiving_ = false;
|
| -}
|
| -
|
| -bool RtpStreamReceiver::IsPacketInOrder(const RTPHeader& header) const {
|
| - StreamStatistician* statistician =
|
| - rtp_receive_statistics_->GetStatistician(header.ssrc);
|
| - if (!statistician)
|
| - return false;
|
| - return statistician->IsPacketInOrder(header.sequenceNumber);
|
| -}
|
| -
|
| -bool RtpStreamReceiver::IsPacketRetransmitted(const RTPHeader& header,
|
| - bool in_order) const {
|
| - // Retransmissions are handled separately if RTX is enabled.
|
| - if (rtp_payload_registry_.RtxEnabled())
|
| - return false;
|
| - StreamStatistician* statistician =
|
| - rtp_receive_statistics_->GetStatistician(header.ssrc);
|
| - if (!statistician)
|
| - return false;
|
| - // Check if this is a retransmission.
|
| - int64_t min_rtt = 0;
|
| - rtp_rtcp_->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr);
|
| - return !in_order &&
|
| - statistician->IsRetransmitOfOldPacket(header, min_rtt);
|
| -}
|
| -
|
| -void RtpStreamReceiver::UpdateHistograms() {
|
| - FecPacketCounter counter = ulpfec_receiver_->GetPacketCounter();
|
| - if (counter.first_packet_time_ms == -1)
|
| - return;
|
| -
|
| - int64_t elapsed_sec =
|
| - (clock_->TimeInMilliseconds() - counter.first_packet_time_ms) / 1000;
|
| - if (elapsed_sec < metrics::kMinRunTimeInSeconds)
|
| - return;
|
| -
|
| - if (counter.num_packets > 0) {
|
| - RTC_HISTOGRAM_PERCENTAGE(
|
| - "WebRTC.Video.ReceivedFecPacketsInPercent",
|
| - static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
|
| - }
|
| - if (counter.num_fec_packets > 0) {
|
| - RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
|
| - static_cast<int>(counter.num_recovered_packets *
|
| - 100 / counter.num_fec_packets));
|
| - }
|
| -}
|
| -
|
| -void RtpStreamReceiver::EnableReceiveRtpHeaderExtension(
|
| - const std::string& extension, int id) {
|
| - // One-byte-extension local identifiers are in the range 1-14 inclusive.
|
| - RTC_DCHECK_GE(id, 1);
|
| - RTC_DCHECK_LE(id, 14);
|
| - RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
|
| - RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
|
| - StringToRtpExtensionType(extension), id));
|
| -}
|
| -
|
| -void RtpStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) {
|
| - auto codec_params_it = pt_codec_params_.find(payload_type);
|
| - if (codec_params_it == pt_codec_params_.end())
|
| - return;
|
| -
|
| - LOG(LS_INFO) << "Found out of band supplied codec parameters for"
|
| - << " payload type: " << static_cast<int>(payload_type);
|
| -
|
| - H264SpropParameterSets sprop_decoder;
|
| - auto sprop_base64_it =
|
| - codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets);
|
| -
|
| - if (sprop_base64_it == codec_params_it->second.end())
|
| - return;
|
| -
|
| - if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
|
| - return;
|
| -
|
| - tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
|
| - sprop_decoder.pps_nalu());
|
| -}
|
| -
|
| -} // namespace webrtc
|
|
|