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| 1 /* | |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #include "webrtc/video/rtp_stream_receiver.h" | |
| 12 | |
| 13 #include <vector> | |
| 14 #include <utility> | |
| 15 | |
| 16 #include "webrtc/base/checks.h" | |
| 17 #include "webrtc/base/location.h" | |
| 18 #include "webrtc/base/logging.h" | |
| 19 #include "webrtc/common_types.h" | |
| 20 #include "webrtc/config.h" | |
| 21 #include "webrtc/media/base/mediaconstants.h" | |
| 22 #include "webrtc/modules/pacing/packet_router.h" | |
| 23 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat
or.h" | |
| 24 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | |
| 25 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" | |
| 26 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | |
| 27 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | |
| 28 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | |
| 29 #include "webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h" | |
| 30 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" | |
| 31 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | |
| 32 #include "webrtc/modules/video_coding/frame_object.h" | |
| 33 #include "webrtc/modules/video_coding/h264_sprop_parameter_sets.h" | |
| 34 #include "webrtc/modules/video_coding/h264_sps_pps_tracker.h" | |
| 35 #include "webrtc/modules/video_coding/packet_buffer.h" | |
| 36 #include "webrtc/modules/video_coding/video_coding_impl.h" | |
| 37 #include "webrtc/system_wrappers/include/field_trial.h" | |
| 38 #include "webrtc/system_wrappers/include/metrics.h" | |
| 39 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h" | |
| 40 #include "webrtc/video/receive_statistics_proxy.h" | |
| 41 | |
| 42 namespace webrtc { | |
| 43 | |
| 44 namespace { | |
| 45 constexpr int kPacketBufferStartSize = 32; | |
| 46 constexpr int kPacketBufferMaxSixe = 2048; | |
| 47 } | |
| 48 | |
| 49 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule( | |
| 50 ReceiveStatistics* receive_statistics, | |
| 51 Transport* outgoing_transport, | |
| 52 RtcpRttStats* rtt_stats, | |
| 53 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, | |
| 54 TransportSequenceNumberAllocator* transport_sequence_number_allocator) { | |
| 55 RtpRtcp::Configuration configuration; | |
| 56 configuration.audio = false; | |
| 57 configuration.receiver_only = true; | |
| 58 configuration.receive_statistics = receive_statistics; | |
| 59 configuration.outgoing_transport = outgoing_transport; | |
| 60 configuration.intra_frame_callback = nullptr; | |
| 61 configuration.rtt_stats = rtt_stats; | |
| 62 configuration.rtcp_packet_type_counter_observer = | |
| 63 rtcp_packet_type_counter_observer; | |
| 64 configuration.transport_sequence_number_allocator = | |
| 65 transport_sequence_number_allocator; | |
| 66 configuration.send_bitrate_observer = nullptr; | |
| 67 configuration.send_frame_count_observer = nullptr; | |
| 68 configuration.send_side_delay_observer = nullptr; | |
| 69 configuration.send_packet_observer = nullptr; | |
| 70 configuration.bandwidth_callback = nullptr; | |
| 71 configuration.transport_feedback_callback = nullptr; | |
| 72 | |
| 73 std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration)); | |
| 74 rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); | |
| 75 | |
| 76 return rtp_rtcp; | |
| 77 } | |
| 78 | |
| 79 static const int kPacketLogIntervalMs = 10000; | |
| 80 | |
| 81 RtpStreamReceiver::RtpStreamReceiver( | |
| 82 Transport* transport, | |
| 83 RtcpRttStats* rtt_stats, | |
| 84 PacketRouter* packet_router, | |
| 85 const VideoReceiveStream::Config* config, | |
| 86 ReceiveStatisticsProxy* receive_stats_proxy, | |
| 87 ProcessThread* process_thread, | |
| 88 NackSender* nack_sender, | |
| 89 KeyFrameRequestSender* keyframe_request_sender, | |
| 90 video_coding::OnCompleteFrameCallback* complete_frame_callback, | |
| 91 VCMTiming* timing) | |
| 92 : clock_(Clock::GetRealTimeClock()), | |
| 93 config_(*config), | |
| 94 packet_router_(packet_router), | |
| 95 process_thread_(process_thread), | |
| 96 ntp_estimator_(clock_), | |
| 97 rtp_header_parser_(RtpHeaderParser::Create()), | |
| 98 rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_, | |
| 99 this, | |
| 100 this, | |
| 101 &rtp_payload_registry_)), | |
| 102 rtp_receive_statistics_(ReceiveStatistics::Create(clock_)), | |
| 103 ulpfec_receiver_(UlpfecReceiver::Create(this)), | |
| 104 receiving_(false), | |
| 105 restored_packet_in_use_(false), | |
| 106 last_packet_log_ms_(-1), | |
| 107 rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(), | |
| 108 transport, | |
| 109 rtt_stats, | |
| 110 receive_stats_proxy, | |
| 111 packet_router)), | |
| 112 complete_frame_callback_(complete_frame_callback), | |
| 113 keyframe_request_sender_(keyframe_request_sender), | |
| 114 timing_(timing), | |
| 115 has_received_frame_(false) { | |
| 116 packet_router_->AddReceiveRtpModule(rtp_rtcp_.get()); | |
| 117 rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy); | |
| 118 rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy); | |
| 119 | |
| 120 RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff) | |
| 121 << "A stream should not be configured with RTCP disabled. This value is " | |
| 122 "reserved for internal usage."; | |
| 123 RTC_DCHECK(config_.rtp.remote_ssrc != 0); | |
| 124 // TODO(pbos): What's an appropriate local_ssrc for receive-only streams? | |
| 125 RTC_DCHECK(config_.rtp.local_ssrc != 0); | |
| 126 RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc); | |
| 127 | |
| 128 rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode); | |
| 129 rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc); | |
| 130 rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp); | |
| 131 | |
| 132 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { | |
| 133 EnableReceiveRtpHeaderExtension(config_.rtp.extensions[i].uri, | |
| 134 config_.rtp.extensions[i].id); | |
| 135 } | |
| 136 | |
| 137 static const int kMaxPacketAgeToNack = 450; | |
| 138 const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0) | |
| 139 ? kMaxPacketAgeToNack | |
| 140 : kDefaultMaxReorderingThreshold; | |
| 141 rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold); | |
| 142 | |
| 143 if (config_.rtp.rtx_ssrc) { | |
| 144 rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc); | |
| 145 | |
| 146 for (const auto& kv : config_.rtp.rtx_payload_types) { | |
| 147 RTC_DCHECK(kv.second != 0); | |
| 148 rtp_payload_registry_.SetRtxPayloadType(kv.second, kv.first); | |
| 149 } | |
| 150 } | |
| 151 | |
| 152 if (IsUlpfecEnabled()) { | |
| 153 VideoCodec ulpfec_codec = {}; | |
| 154 ulpfec_codec.codecType = kVideoCodecULPFEC; | |
| 155 strncpy(ulpfec_codec.plName, "ulpfec", sizeof(ulpfec_codec.plName)); | |
| 156 ulpfec_codec.plType = config_.rtp.ulpfec.ulpfec_payload_type; | |
| 157 RTC_CHECK(AddReceiveCodec(ulpfec_codec)); | |
| 158 } | |
| 159 | |
| 160 if (IsRedEnabled()) { | |
| 161 VideoCodec red_codec = {}; | |
| 162 red_codec.codecType = kVideoCodecRED; | |
| 163 strncpy(red_codec.plName, "red", sizeof(red_codec.plName)); | |
| 164 red_codec.plType = config_.rtp.ulpfec.red_payload_type; | |
| 165 RTC_CHECK(AddReceiveCodec(red_codec)); | |
| 166 if (config_.rtp.ulpfec.red_rtx_payload_type != -1) { | |
| 167 rtp_payload_registry_.SetRtxPayloadType( | |
| 168 config_.rtp.ulpfec.red_rtx_payload_type, | |
| 169 config_.rtp.ulpfec.red_payload_type); | |
| 170 } | |
| 171 } | |
| 172 | |
| 173 if (config_.rtp.rtcp_xr.receiver_reference_time_report) | |
| 174 rtp_rtcp_->SetRtcpXrRrtrStatus(true); | |
| 175 | |
| 176 // Stats callback for CNAME changes. | |
| 177 rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy); | |
| 178 | |
| 179 process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); | |
| 180 | |
| 181 if (config_.rtp.nack.rtp_history_ms != 0) { | |
| 182 nack_module_.reset( | |
| 183 new NackModule(clock_, nack_sender, keyframe_request_sender)); | |
| 184 process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE); | |
| 185 } | |
| 186 | |
| 187 packet_buffer_ = video_coding::PacketBuffer::Create( | |
| 188 clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this); | |
| 189 reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this)); | |
| 190 } | |
| 191 | |
| 192 RtpStreamReceiver::~RtpStreamReceiver() { | |
| 193 if (nack_module_) { | |
| 194 process_thread_->DeRegisterModule(nack_module_.get()); | |
| 195 } | |
| 196 | |
| 197 process_thread_->DeRegisterModule(rtp_rtcp_.get()); | |
| 198 | |
| 199 packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get()); | |
| 200 UpdateHistograms(); | |
| 201 } | |
| 202 | |
| 203 bool RtpStreamReceiver::AddReceiveCodec( | |
| 204 const VideoCodec& video_codec, | |
| 205 const std::map<std::string, std::string>& codec_params) { | |
| 206 pt_codec_params_.insert(make_pair(video_codec.plType, codec_params)); | |
| 207 return AddReceiveCodec(video_codec); | |
| 208 } | |
| 209 | |
| 210 bool RtpStreamReceiver::AddReceiveCodec(const VideoCodec& video_codec) { | |
| 211 int8_t old_pltype = -1; | |
| 212 if (rtp_payload_registry_.ReceivePayloadType(video_codec, &old_pltype) != | |
| 213 -1) { | |
| 214 rtp_payload_registry_.DeRegisterReceivePayload(old_pltype); | |
| 215 } | |
| 216 return rtp_payload_registry_.RegisterReceivePayload(video_codec) == 0; | |
| 217 } | |
| 218 | |
| 219 uint32_t RtpStreamReceiver::GetRemoteSsrc() const { | |
| 220 return rtp_receiver_->SSRC(); | |
| 221 } | |
| 222 | |
| 223 int RtpStreamReceiver::GetCsrcs(uint32_t* csrcs) const { | |
| 224 return rtp_receiver_->CSRCs(csrcs); | |
| 225 } | |
| 226 | |
| 227 RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const { | |
| 228 return rtp_receiver_.get(); | |
| 229 } | |
| 230 | |
| 231 int32_t RtpStreamReceiver::OnReceivedPayloadData( | |
| 232 const uint8_t* payload_data, | |
| 233 size_t payload_size, | |
| 234 const WebRtcRTPHeader* rtp_header) { | |
| 235 WebRtcRTPHeader rtp_header_with_ntp = *rtp_header; | |
| 236 rtp_header_with_ntp.ntp_time_ms = | |
| 237 ntp_estimator_.Estimate(rtp_header->header.timestamp); | |
| 238 VCMPacket packet(payload_data, payload_size, rtp_header_with_ntp); | |
| 239 packet.timesNacked = | |
| 240 nack_module_ ? nack_module_->OnReceivedPacket(packet) : -1; | |
| 241 | |
| 242 // In the case of a video stream without picture ids and no rtx the | |
| 243 // RtpFrameReferenceFinder will need to know about padding to | |
| 244 // correctly calculate frame references. | |
| 245 if (packet.sizeBytes == 0) { | |
| 246 reference_finder_->PaddingReceived(packet.seqNum); | |
| 247 return 0; | |
| 248 } | |
| 249 | |
| 250 if (packet.codec == kVideoCodecH264) { | |
| 251 // Only when we start to receive packets will we know what payload type | |
| 252 // that will be used. When we know the payload type insert the correct | |
| 253 // sps/pps into the tracker. | |
| 254 if (packet.payloadType != last_payload_type_) { | |
| 255 last_payload_type_ = packet.payloadType; | |
| 256 InsertSpsPpsIntoTracker(packet.payloadType); | |
| 257 } | |
| 258 | |
| 259 switch (tracker_.CopyAndFixBitstream(&packet)) { | |
| 260 case video_coding::H264SpsPpsTracker::kRequestKeyframe: | |
| 261 keyframe_request_sender_->RequestKeyFrame(); | |
| 262 FALLTHROUGH(); | |
| 263 case video_coding::H264SpsPpsTracker::kDrop: | |
| 264 return 0; | |
| 265 case video_coding::H264SpsPpsTracker::kInsert: | |
| 266 break; | |
| 267 } | |
| 268 | |
| 269 } else { | |
| 270 uint8_t* data = new uint8_t[packet.sizeBytes]; | |
| 271 memcpy(data, packet.dataPtr, packet.sizeBytes); | |
| 272 packet.dataPtr = data; | |
| 273 } | |
| 274 | |
| 275 packet_buffer_->InsertPacket(&packet); | |
| 276 return 0; | |
| 277 } | |
| 278 | |
| 279 // TODO(nisse): Try to delete this method. Obstacles: It is used by | |
| 280 // ParseAndHandleEncapsulatingHeader, for handling Rtx packets, and | |
| 281 // for callbacks from |ulpfec_receiver_|. | |
| 282 void RtpStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet, | |
| 283 size_t rtp_packet_length) { | |
| 284 RTPHeader header; | |
| 285 if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { | |
| 286 return; | |
| 287 } | |
| 288 header.payload_type_frequency = kVideoPayloadTypeFrequency; | |
| 289 bool in_order = IsPacketInOrder(header); | |
| 290 ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); | |
| 291 } | |
| 292 | |
| 293 // TODO(pbos): Remove as soon as audio can handle a changing payload type | |
| 294 // without this callback. | |
| 295 int32_t RtpStreamReceiver::OnInitializeDecoder( | |
| 296 const int8_t payload_type, | |
| 297 const char payload_name[RTP_PAYLOAD_NAME_SIZE], | |
| 298 const int frequency, | |
| 299 const size_t channels, | |
| 300 const uint32_t rate) { | |
| 301 RTC_NOTREACHED(); | |
| 302 return 0; | |
| 303 } | |
| 304 | |
| 305 void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) { | |
| 306 rtp_rtcp_->SetRemoteSSRC(ssrc); | |
| 307 } | |
| 308 | |
| 309 // This method handles both regular RTP packets and packets recovered | |
| 310 // via FlexFEC. | |
| 311 void RtpStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) { | |
| 312 { | |
| 313 rtc::CritScope lock(&receive_cs_); | |
| 314 if (!receiving_) { | |
| 315 return; | |
| 316 } | |
| 317 | |
| 318 if (!packet.recovered()) { | |
| 319 int64_t now_ms = clock_->TimeInMilliseconds(); | |
| 320 | |
| 321 // Periodically log the RTP header of incoming packets. | |
| 322 if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) { | |
| 323 std::stringstream ss; | |
| 324 ss << "Packet received on SSRC: " << packet.Ssrc() | |
| 325 << " with payload type: " << static_cast<int>(packet.PayloadType()) | |
| 326 << ", timestamp: " << packet.Timestamp() | |
| 327 << ", sequence number: " << packet.SequenceNumber() | |
| 328 << ", arrival time: " << packet.arrival_time_ms(); | |
| 329 int32_t time_offset; | |
| 330 if (packet.GetExtension<TransmissionOffset>(&time_offset)) { | |
| 331 ss << ", toffset: " << time_offset; | |
| 332 } | |
| 333 uint32_t send_time; | |
| 334 if (packet.GetExtension<AbsoluteSendTime>(&send_time)) { | |
| 335 ss << ", abs send time: " << send_time; | |
| 336 } | |
| 337 LOG(LS_INFO) << ss.str(); | |
| 338 last_packet_log_ms_ = now_ms; | |
| 339 } | |
| 340 } | |
| 341 } | |
| 342 | |
| 343 // TODO(nisse): Delete use of GetHeader, but needs refactoring of | |
| 344 // ReceivePacket and IncomingPacket methods below. | |
| 345 RTPHeader header; | |
| 346 packet.GetHeader(&header); | |
| 347 | |
| 348 header.payload_type_frequency = kVideoPayloadTypeFrequency; | |
| 349 | |
| 350 bool in_order = IsPacketInOrder(header); | |
| 351 if (!packet.recovered()) { | |
| 352 // TODO(nisse): Why isn't this done for recovered packets? | |
| 353 rtp_payload_registry_.SetIncomingPayloadType(header); | |
| 354 } | |
| 355 ReceivePacket(packet.data(), packet.size(), header, in_order); | |
| 356 // Update receive statistics after ReceivePacket. | |
| 357 // Receive statistics will be reset if the payload type changes (make sure | |
| 358 // that the first packet is included in the stats). | |
| 359 if (!packet.recovered()) { | |
| 360 // TODO(nisse): We should pass a recovered flag to stats, to aid | |
| 361 // fixing bug bugs.webrtc.org/6339. | |
| 362 rtp_receive_statistics_->IncomingPacket( | |
| 363 header, packet.size(), IsPacketRetransmitted(header, in_order)); | |
| 364 } | |
| 365 } | |
| 366 | |
| 367 int32_t RtpStreamReceiver::RequestKeyFrame() { | |
| 368 return rtp_rtcp_->RequestKeyFrame(); | |
| 369 } | |
| 370 | |
| 371 bool RtpStreamReceiver::IsUlpfecEnabled() const { | |
| 372 return config_.rtp.ulpfec.ulpfec_payload_type != -1; | |
| 373 } | |
| 374 | |
| 375 bool RtpStreamReceiver::IsRedEnabled() const { | |
| 376 return config_.rtp.ulpfec.red_payload_type != -1; | |
| 377 } | |
| 378 | |
| 379 bool RtpStreamReceiver::IsRetransmissionsEnabled() const { | |
| 380 return config_.rtp.nack.rtp_history_ms > 0; | |
| 381 } | |
| 382 | |
| 383 void RtpStreamReceiver::RequestPacketRetransmit( | |
| 384 const std::vector<uint16_t>& sequence_numbers) { | |
| 385 rtp_rtcp_->SendNack(sequence_numbers); | |
| 386 } | |
| 387 | |
| 388 int32_t RtpStreamReceiver::ResendPackets(const uint16_t* sequence_numbers, | |
| 389 uint16_t length) { | |
| 390 return rtp_rtcp_->SendNACK(sequence_numbers, length); | |
| 391 } | |
| 392 | |
| 393 void RtpStreamReceiver::OnReceivedFrame( | |
| 394 std::unique_ptr<video_coding::RtpFrameObject> frame) { | |
| 395 | |
| 396 if (!has_received_frame_) { | |
| 397 has_received_frame_ = true; | |
| 398 if (frame->FrameType() != kVideoFrameKey) | |
| 399 keyframe_request_sender_->RequestKeyFrame(); | |
| 400 } | |
| 401 | |
| 402 if (!frame->delayed_by_retransmission()) | |
| 403 timing_->IncomingTimestamp(frame->timestamp, clock_->TimeInMilliseconds()); | |
| 404 reference_finder_->ManageFrame(std::move(frame)); | |
| 405 } | |
| 406 | |
| 407 void RtpStreamReceiver::OnCompleteFrame( | |
| 408 std::unique_ptr<video_coding::FrameObject> frame) { | |
| 409 { | |
| 410 rtc::CritScope lock(&last_seq_num_cs_); | |
| 411 video_coding::RtpFrameObject* rtp_frame = | |
| 412 static_cast<video_coding::RtpFrameObject*>(frame.get()); | |
| 413 last_seq_num_for_pic_id_[rtp_frame->picture_id] = rtp_frame->last_seq_num(); | |
| 414 } | |
| 415 complete_frame_callback_->OnCompleteFrame(std::move(frame)); | |
| 416 } | |
| 417 | |
| 418 void RtpStreamReceiver::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { | |
| 419 if (nack_module_) | |
| 420 nack_module_->UpdateRtt(max_rtt_ms); | |
| 421 } | |
| 422 | |
| 423 rtc::Optional<int64_t> RtpStreamReceiver::LastReceivedPacketMs() const { | |
| 424 return packet_buffer_->LastReceivedPacketMs(); | |
| 425 } | |
| 426 | |
| 427 rtc::Optional<int64_t> RtpStreamReceiver::LastReceivedKeyframePacketMs() const { | |
| 428 return packet_buffer_->LastReceivedKeyframePacketMs(); | |
| 429 } | |
| 430 | |
| 431 void RtpStreamReceiver::ReceivePacket(const uint8_t* packet, | |
| 432 size_t packet_length, | |
| 433 const RTPHeader& header, | |
| 434 bool in_order) { | |
| 435 if (rtp_payload_registry_.IsEncapsulated(header)) { | |
| 436 ParseAndHandleEncapsulatingHeader(packet, packet_length, header); | |
| 437 return; | |
| 438 } | |
| 439 const uint8_t* payload = packet + header.headerLength; | |
| 440 assert(packet_length >= header.headerLength); | |
| 441 size_t payload_length = packet_length - header.headerLength; | |
| 442 PayloadUnion payload_specific; | |
| 443 if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType, | |
| 444 &payload_specific)) { | |
| 445 return; | |
| 446 } | |
| 447 rtp_receiver_->IncomingRtpPacket(header, payload, payload_length, | |
| 448 payload_specific, in_order); | |
| 449 } | |
| 450 | |
| 451 void RtpStreamReceiver::ParseAndHandleEncapsulatingHeader( | |
| 452 const uint8_t* packet, size_t packet_length, const RTPHeader& header) { | |
| 453 if (rtp_payload_registry_.IsRed(header)) { | |
| 454 int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type(); | |
| 455 if (packet[header.headerLength] == ulpfec_pt) { | |
| 456 rtp_receive_statistics_->FecPacketReceived(header, packet_length); | |
| 457 // Notify video_receiver about received FEC packets to avoid NACKing these | |
| 458 // packets. | |
| 459 NotifyReceiverOfFecPacket(header); | |
| 460 } | |
| 461 if (ulpfec_receiver_->AddReceivedRedPacket(header, packet, packet_length, | |
| 462 ulpfec_pt) != 0) { | |
| 463 return; | |
| 464 } | |
| 465 ulpfec_receiver_->ProcessReceivedFec(); | |
| 466 } else if (rtp_payload_registry_.IsRtx(header)) { | |
| 467 if (header.headerLength + header.paddingLength == packet_length) { | |
| 468 // This is an empty packet and should be silently dropped before trying to | |
| 469 // parse the RTX header. | |
| 470 return; | |
| 471 } | |
| 472 // Remove the RTX header and parse the original RTP header. | |
| 473 if (packet_length < header.headerLength) | |
| 474 return; | |
| 475 if (packet_length > sizeof(restored_packet_)) | |
| 476 return; | |
| 477 rtc::CritScope lock(&receive_cs_); | |
| 478 if (restored_packet_in_use_) { | |
| 479 LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet."; | |
| 480 return; | |
| 481 } | |
| 482 if (!rtp_payload_registry_.RestoreOriginalPacket( | |
| 483 restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(), | |
| 484 header)) { | |
| 485 LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: " | |
| 486 << header.ssrc << " payload type: " | |
| 487 << static_cast<int>(header.payloadType); | |
| 488 return; | |
| 489 } | |
| 490 restored_packet_in_use_ = true; | |
| 491 OnRecoveredPacket(restored_packet_, packet_length); | |
| 492 restored_packet_in_use_ = false; | |
| 493 } | |
| 494 } | |
| 495 | |
| 496 void RtpStreamReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) { | |
| 497 int8_t last_media_payload_type = | |
| 498 rtp_payload_registry_.last_received_media_payload_type(); | |
| 499 if (last_media_payload_type < 0) { | |
| 500 LOG(LS_WARNING) << "Failed to get last media payload type."; | |
| 501 return; | |
| 502 } | |
| 503 // Fake an empty media packet. | |
| 504 WebRtcRTPHeader rtp_header = {}; | |
| 505 rtp_header.header = header; | |
| 506 rtp_header.header.payloadType = last_media_payload_type; | |
| 507 rtp_header.header.paddingLength = 0; | |
| 508 PayloadUnion payload_specific; | |
| 509 if (!rtp_payload_registry_.GetPayloadSpecifics(last_media_payload_type, | |
| 510 &payload_specific)) { | |
| 511 LOG(LS_WARNING) << "Failed to get payload specifics."; | |
| 512 return; | |
| 513 } | |
| 514 rtp_header.type.Video.codec = payload_specific.Video.videoCodecType; | |
| 515 rtp_header.type.Video.rotation = kVideoRotation_0; | |
| 516 if (header.extension.hasVideoRotation) { | |
| 517 rtp_header.type.Video.rotation = header.extension.videoRotation; | |
| 518 } | |
| 519 rtp_header.type.Video.content_type = VideoContentType::UNSPECIFIED; | |
| 520 if (header.extension.hasVideoContentType) { | |
| 521 rtp_header.type.Video.content_type = header.extension.videoContentType; | |
| 522 } | |
| 523 rtp_header.type.Video.playout_delay = header.extension.playout_delay; | |
| 524 | |
| 525 OnReceivedPayloadData(nullptr, 0, &rtp_header); | |
| 526 } | |
| 527 | |
| 528 bool RtpStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet, | |
| 529 size_t rtcp_packet_length) { | |
| 530 { | |
| 531 rtc::CritScope lock(&receive_cs_); | |
| 532 if (!receiving_) { | |
| 533 return false; | |
| 534 } | |
| 535 } | |
| 536 | |
| 537 rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); | |
| 538 | |
| 539 int64_t rtt = 0; | |
| 540 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr); | |
| 541 if (rtt == 0) { | |
| 542 // Waiting for valid rtt. | |
| 543 return true; | |
| 544 } | |
| 545 uint32_t ntp_secs = 0; | |
| 546 uint32_t ntp_frac = 0; | |
| 547 uint32_t rtp_timestamp = 0; | |
| 548 if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr, | |
| 549 &rtp_timestamp) != 0) { | |
| 550 // Waiting for RTCP. | |
| 551 return true; | |
| 552 } | |
| 553 ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); | |
| 554 | |
| 555 return true; | |
| 556 } | |
| 557 | |
| 558 void RtpStreamReceiver::FrameContinuous(uint16_t picture_id) { | |
| 559 if (!nack_module_) | |
| 560 return; | |
| 561 | |
| 562 int seq_num = -1; | |
| 563 { | |
| 564 rtc::CritScope lock(&last_seq_num_cs_); | |
| 565 auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id); | |
| 566 if (seq_num_it != last_seq_num_for_pic_id_.end()) | |
| 567 seq_num = seq_num_it->second; | |
| 568 } | |
| 569 if (seq_num != -1) | |
| 570 nack_module_->ClearUpTo(seq_num); | |
| 571 } | |
| 572 | |
| 573 void RtpStreamReceiver::FrameDecoded(uint16_t picture_id) { | |
| 574 int seq_num = -1; | |
| 575 { | |
| 576 rtc::CritScope lock(&last_seq_num_cs_); | |
| 577 auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id); | |
| 578 if (seq_num_it != last_seq_num_for_pic_id_.end()) { | |
| 579 seq_num = seq_num_it->second; | |
| 580 last_seq_num_for_pic_id_.erase(last_seq_num_for_pic_id_.begin(), | |
| 581 ++seq_num_it); | |
| 582 } | |
| 583 } | |
| 584 if (seq_num != -1) { | |
| 585 packet_buffer_->ClearTo(seq_num); | |
| 586 reference_finder_->ClearTo(seq_num); | |
| 587 } | |
| 588 } | |
| 589 | |
| 590 void RtpStreamReceiver::SignalNetworkState(NetworkState state) { | |
| 591 rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode | |
| 592 : RtcpMode::kOff); | |
| 593 } | |
| 594 | |
| 595 void RtpStreamReceiver::StartReceive() { | |
| 596 rtc::CritScope lock(&receive_cs_); | |
| 597 receiving_ = true; | |
| 598 } | |
| 599 | |
| 600 void RtpStreamReceiver::StopReceive() { | |
| 601 rtc::CritScope lock(&receive_cs_); | |
| 602 receiving_ = false; | |
| 603 } | |
| 604 | |
| 605 bool RtpStreamReceiver::IsPacketInOrder(const RTPHeader& header) const { | |
| 606 StreamStatistician* statistician = | |
| 607 rtp_receive_statistics_->GetStatistician(header.ssrc); | |
| 608 if (!statistician) | |
| 609 return false; | |
| 610 return statistician->IsPacketInOrder(header.sequenceNumber); | |
| 611 } | |
| 612 | |
| 613 bool RtpStreamReceiver::IsPacketRetransmitted(const RTPHeader& header, | |
| 614 bool in_order) const { | |
| 615 // Retransmissions are handled separately if RTX is enabled. | |
| 616 if (rtp_payload_registry_.RtxEnabled()) | |
| 617 return false; | |
| 618 StreamStatistician* statistician = | |
| 619 rtp_receive_statistics_->GetStatistician(header.ssrc); | |
| 620 if (!statistician) | |
| 621 return false; | |
| 622 // Check if this is a retransmission. | |
| 623 int64_t min_rtt = 0; | |
| 624 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr); | |
| 625 return !in_order && | |
| 626 statistician->IsRetransmitOfOldPacket(header, min_rtt); | |
| 627 } | |
| 628 | |
| 629 void RtpStreamReceiver::UpdateHistograms() { | |
| 630 FecPacketCounter counter = ulpfec_receiver_->GetPacketCounter(); | |
| 631 if (counter.first_packet_time_ms == -1) | |
| 632 return; | |
| 633 | |
| 634 int64_t elapsed_sec = | |
| 635 (clock_->TimeInMilliseconds() - counter.first_packet_time_ms) / 1000; | |
| 636 if (elapsed_sec < metrics::kMinRunTimeInSeconds) | |
| 637 return; | |
| 638 | |
| 639 if (counter.num_packets > 0) { | |
| 640 RTC_HISTOGRAM_PERCENTAGE( | |
| 641 "WebRTC.Video.ReceivedFecPacketsInPercent", | |
| 642 static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets)); | |
| 643 } | |
| 644 if (counter.num_fec_packets > 0) { | |
| 645 RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec", | |
| 646 static_cast<int>(counter.num_recovered_packets * | |
| 647 100 / counter.num_fec_packets)); | |
| 648 } | |
| 649 } | |
| 650 | |
| 651 void RtpStreamReceiver::EnableReceiveRtpHeaderExtension( | |
| 652 const std::string& extension, int id) { | |
| 653 // One-byte-extension local identifiers are in the range 1-14 inclusive. | |
| 654 RTC_DCHECK_GE(id, 1); | |
| 655 RTC_DCHECK_LE(id, 14); | |
| 656 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); | |
| 657 RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( | |
| 658 StringToRtpExtensionType(extension), id)); | |
| 659 } | |
| 660 | |
| 661 void RtpStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) { | |
| 662 auto codec_params_it = pt_codec_params_.find(payload_type); | |
| 663 if (codec_params_it == pt_codec_params_.end()) | |
| 664 return; | |
| 665 | |
| 666 LOG(LS_INFO) << "Found out of band supplied codec parameters for" | |
| 667 << " payload type: " << static_cast<int>(payload_type); | |
| 668 | |
| 669 H264SpropParameterSets sprop_decoder; | |
| 670 auto sprop_base64_it = | |
| 671 codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets); | |
| 672 | |
| 673 if (sprop_base64_it == codec_params_it->second.end()) | |
| 674 return; | |
| 675 | |
| 676 if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str())) | |
| 677 return; | |
| 678 | |
| 679 tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(), | |
| 680 sprop_decoder.pps_nalu()); | |
| 681 } | |
| 682 | |
| 683 } // namespace webrtc | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 2926253002:
Rename class RtpStreamReceiver --> RtpVideoStreamReceiver. (Closed)
