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 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 13 14 #include <list> 15 #include <map> 16 #include <utility> 17 #include <vector> 18 19 #include "webrtc/base/random.h" 20 #include "webrtc/base/thread_annotations.h" 21 #include "webrtc/common_types.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 23 #include "webrtc/modules/rtp_rtcp/source/bitrate.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" 26 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" 27 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 28 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" 29 #include "webrtc/transport.h" 30 31 namespace webrtc { 32 33 class BitrateAggregator; 34 class CriticalSectionWrapper; 35 class RTPSenderAudio; 36 class RTPSenderVideo; 37 38 class RTPSenderInterface { 39 public: 40 RTPSenderInterface() {} 41 virtual ~RTPSenderInterface() {} 42 43 enum CVOMode { 44 kCVONone, 45 kCVOInactive, // CVO rtp header extension is registered but haven't 46 // received any frame with rotation pending. 47 kCVOActivated, // CVO rtp header extension will be present in the rtp 48 // packets. 49 }; 50 51 virtual uint32_t SSRC() const = 0; 52 virtual uint32_t Timestamp() const = 0; 53 54 virtual int32_t BuildRTPheader(uint8_t* data_buffer, 55 int8_t payload_type, 56 bool marker_bit, 57 uint32_t capture_timestamp, 58 int64_t capture_time_ms, 59 bool timestamp_provided = true, 60 bool inc_sequence_number = true) = 0; 61 62 virtual size_t RTPHeaderLength() const = 0; 63 // Returns the next sequence number to use for a packet and allocates 64 // 'packets_to_send' number of sequence numbers. It's important all allocated 65 // sequence numbers are used in sequence to avoid perceived packet loss. 66 virtual uint16_t AllocateSequenceNumber(uint16_t packets_to_send) = 0; 67 virtual uint16_t SequenceNumber() const = 0; 68 virtual size_t MaxPayloadLength() const = 0; 69 virtual size_t MaxDataPayloadLength() const = 0; 70 virtual uint16_t PacketOverHead() const = 0; 71 virtual uint16_t ActualSendBitrateKbit() const = 0; 72 73 virtual int32_t SendToNetwork(uint8_t* data_buffer, 74 size_t payload_length, 75 size_t rtp_header_length, 76 int64_t capture_time_ms, 77 StorageType storage, 78 RtpPacketSender::Priority priority) = 0; 79 80 virtual bool UpdateVideoRotation(uint8_t* rtp_packet, 81 size_t rtp_packet_length, 82 const RTPHeader& rtp_header, 83 VideoRotation rotation) const = 0; 84 virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) = 0; 85 virtual CVOMode ActivateCVORtpHeaderExtension() = 0; 86 }; 87 88 class RTPSender : public RTPSenderInterface { 89 public: 90 RTPSender(bool audio, 91 Clock* clock, 92 Transport* transport, 93 RtpAudioFeedback* audio_feedback, 94 RtpPacketSender* paced_sender, 95 TransportSequenceNumberAllocator* sequence_number_allocator, 96 TransportFeedbackObserver* transport_feedback_callback, 97 BitrateStatisticsObserver* bitrate_callback, 98 FrameCountObserver* frame_count_observer, 99 SendSideDelayObserver* send_side_delay_observer); 100 virtual ~RTPSender(); 101 102 void ProcessBitrate(); 103 104 uint16_t ActualSendBitrateKbit() const override; 105 106 uint32_t VideoBitrateSent() const; 107 uint32_t FecOverheadRate() const; 108 uint32_t NackOverheadRate() const; 109 110 void SetTargetBitrate(uint32_t bitrate); 111 uint32_t GetTargetBitrate(); 112 113 // Includes size of RTP and FEC headers. 114 size_t MaxDataPayloadLength() const override; 115 116 int32_t RegisterPayload( 117 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 118 const int8_t payload_type, const uint32_t frequency, 119 const size_t channels, const uint32_t rate); 120 121 int32_t DeRegisterSendPayload(const int8_t payload_type); 122 123 void SetSendPayloadType(int8_t payload_type); 124 125 int8_t SendPayloadType() const; 126 127 int SendPayloadFrequency() const; 128 129 void SetSendingStatus(bool enabled); 130 131 void SetSendingMediaStatus(bool enabled); 132 bool SendingMedia() const; 133 134 void GetDataCounters(StreamDataCounters* rtp_stats, 135 StreamDataCounters* rtx_stats) const; 136 137 uint32_t StartTimestamp() const; 138 void SetStartTimestamp(uint32_t timestamp, bool force); 139 140 uint32_t GenerateNewSSRC(); 141 void SetSSRC(uint32_t ssrc); 142 143 uint16_t SequenceNumber() const override; 144 void SetSequenceNumber(uint16_t seq); 145 146 void SetCsrcs(const std::vector<uint32_t>& csrcs); 147 148 int32_t SetMaxPayloadLength(size_t length, uint16_t packet_over_head); 149 150 int32_t SendOutgoingData(FrameType frame_type, 151 int8_t payload_type, 152 uint32_t timestamp, 153 int64_t capture_time_ms, 154 const uint8_t* payload_data, 155 size_t payload_size, 156 const RTPFragmentationHeader* fragmentation, 157 const RTPVideoHeader* rtp_hdr = NULL); 158 159 // RTP header extension 160 int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset); 161 int32_t SetAbsoluteSendTime(uint32_t absolute_send_time); 162 void SetVideoRotation(VideoRotation rotation); 163 int32_t SetTransportSequenceNumber(uint16_t sequence_number); 164 165 int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id); 166 bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override; 167 int32_t DeregisterRtpHeaderExtension(RTPExtensionType type); 168 169 size_t RtpHeaderExtensionTotalLength() const; 170 171 uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const; 172 173 uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const; 174 uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const; 175 uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const; 176 uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const; 177 uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer, 178 uint16_t sequence_number) const; 179 180 // Verifies that the specified extension is registered, and that it is 181 // present in rtp packet. If extension is not registered kNotRegistered is 182 // returned. If extension cannot be found in the rtp header, or if it is 183 // malformed, kError is returned. Otherwise *extension_offset is set to the 184 // offset of the extension from the beginning of the rtp packet and kOk is 185 // returned. 186 enum class ExtensionStatus { 187 kNotRegistered, 188 kOk, 189 kError, 190 }; 191 ExtensionStatus VerifyExtension(RTPExtensionType extension_type, 192 uint8_t* rtp_packet, 193 size_t rtp_packet_length, 194 const RTPHeader& rtp_header, 195 size_t extension_length_bytes, 196 size_t* extension_offset) const 197 EXCLUSIVE_LOCKS_REQUIRED(send_critsect_.get()); 198 199 bool UpdateAudioLevel(uint8_t* rtp_packet, 200 size_t rtp_packet_length, 201 const RTPHeader& rtp_header, 202 bool is_voiced, 203 uint8_t dBov) const; 204 205 bool UpdateVideoRotation(uint8_t* rtp_packet, 206 size_t rtp_packet_length, 207 const RTPHeader& rtp_header, 208 VideoRotation rotation) const override; 209 210 bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms, 211 bool retransmission); 212 size_t TimeToSendPadding(size_t bytes); 213 214 // NACK. 215 int SelectiveRetransmissions() const; 216 int SetSelectiveRetransmissions(uint8_t settings); 217 void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers, 218 int64_t avg_rtt); 219 220 void SetStorePacketsStatus(bool enable, uint16_t number_to_store); 221 222 bool StorePackets() const; 223 224 int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0); 225 226 bool ProcessNACKBitRate(uint32_t now); 227 228 // RTX. 229 void SetRtxStatus(int mode); 230 int RtxStatus() const; 231 232 uint32_t RtxSsrc() const; 233 void SetRtxSsrc(uint32_t ssrc); 234 235 void SetRtxPayloadType(int payload_type, int associated_payload_type); 236 std::pair<int, int> RtxPayloadType() const; 237 238 // Functions wrapping RTPSenderInterface. 239 int32_t BuildRTPheader(uint8_t* data_buffer, 240 int8_t payload_type, 241 bool marker_bit, 242 uint32_t capture_timestamp, 243 int64_t capture_time_ms, 244 const bool timestamp_provided = true, 245 const bool inc_sequence_number = true) override; 246 247 size_t RTPHeaderLength() const override; 248 uint16_t AllocateSequenceNumber(uint16_t packets_to_send) override; 249 size_t MaxPayloadLength() const override; 250 uint16_t PacketOverHead() const override; 251 252 // Current timestamp. 253 uint32_t Timestamp() const override; 254 uint32_t SSRC() const override; 255 256 int32_t SendToNetwork(uint8_t* data_buffer, 257 size_t payload_length, 258 size_t rtp_header_length, 259 int64_t capture_time_ms, 260 StorageType storage, 261 RtpPacketSender::Priority priority) override; 262 263 // Audio. 264 265 // Send a DTMF tone using RFC 2833 (4733). 266 int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); 267 268 // Set audio packet size, used to determine when it's time to send a DTMF 269 // packet in silence (CNG). 270 int32_t SetAudioPacketSize(uint16_t packet_size_samples); 271 272 // Store the audio level in d_bov for 273 // header-extension-for-audio-level-indication. 274 int32_t SetAudioLevel(uint8_t level_d_bov); 275 276 // Set payload type for Redundant Audio Data RFC 2198. 277 int32_t SetRED(int8_t payload_type); 278 279 // Get payload type for Redundant Audio Data RFC 2198. 280 int32_t RED(int8_t *payload_type) const; 281 282 RtpVideoCodecTypes VideoCodecType() const; 283 284 uint32_t MaxConfiguredBitrateVideo() const; 285 286 // FEC. 287 void SetGenericFECStatus(bool enable, 288 uint8_t payload_type_red, 289 uint8_t payload_type_fec); 290 291 void GenericFECStatus(bool* enable, 292 uint8_t* payload_type_red, 293 uint8_t* payload_type_fec) const; 294 295 int32_t SetFecParameters(const FecProtectionParams *delta_params, 296 const FecProtectionParams *key_params); 297 298 size_t SendPadData(size_t bytes, 299 bool timestamp_provided, 300 uint32_t timestamp, 301 int64_t capture_time_ms); 302 303 // Called on update of RTP statistics. 304 void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback); 305 StreamDataCountersCallback* GetRtpStatisticsCallback() const; 306 307 uint32_t BitrateSent() const; 308 309 void SetRtpState(const RtpState& rtp_state); 310 RtpState GetRtpState() const; 311 void SetRtxRtpState(const RtpState& rtp_state); 312 RtpState GetRtxRtpState() const; 313 CVOMode ActivateCVORtpHeaderExtension() override; 314 315 protected: 316 int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type); 317 318 private: 319 // Maps capture time in milliseconds to send-side delay in milliseconds. 320 // Send-side delay is the difference between transmission time and capture 321 // time. 322 typedef std::map<int64_t, int> SendDelayMap; 323 324 size_t CreateRtpHeader(uint8_t* header, 325 int8_t payload_type, 326 uint32_t ssrc, 327 bool marker_bit, 328 uint32_t timestamp, 329 uint16_t sequence_number, 330 const std::vector<uint32_t>& csrcs) const; 331 332 void UpdateNACKBitRate(uint32_t bytes, int64_t now); 333 334 bool PrepareAndSendPacket(uint8_t* buffer, 335 size_t length, 336 int64_t capture_time_ms, 337 bool send_over_rtx, 338 bool is_retransmit); 339 340 // Return the number of bytes sent. Note that both of these functions may 341 // return a larger value that their argument. 342 size_t TrySendRedundantPayloads(size_t bytes); 343 344 void BuildPaddingPacket(uint8_t* packet, 345 size_t header_length, 346 size_t padding_length); 347 348 void BuildRtxPacket(uint8_t* buffer, size_t* length, 349 uint8_t* buffer_rtx); 350 351 bool SendPacketToNetwork(const uint8_t* packet, 352 size_t size, 353 const PacketOptions& options); 354 355 void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms); 356 357 // Find the byte position of the RTP extension as indicated by |type| in 358 // |rtp_packet|. Return false if such extension doesn't exist. 359 bool FindHeaderExtensionPosition(RTPExtensionType type, 360 const uint8_t* rtp_packet, 361 size_t rtp_packet_length, 362 const RTPHeader& rtp_header, 363 size_t* position) const; 364 365 void UpdateTransmissionTimeOffset(uint8_t* rtp_packet, 366 size_t rtp_packet_length, 367 const RTPHeader& rtp_header, 368 int64_t time_diff_ms) const; 369 void UpdateAbsoluteSendTime(uint8_t* rtp_packet, 370 size_t rtp_packet_length, 371 const RTPHeader& rtp_header, 372 int64_t now_ms) const; 373 // Update the transport sequence number of the packet using a new sequence 374 // number allocated by SequenceNumberAllocator. Returns the assigned sequence 375 // number, or 0 if extension could not be updated. 376 uint16_t UpdateTransportSequenceNumber(uint8_t* rtp_packet, 377 size_t rtp_packet_length, 378 const RTPHeader& rtp_header) const; 379 380 void UpdateRtpStats(const uint8_t* buffer, 381 size_t packet_length, 382 const RTPHeader& header, 383 bool is_rtx, 384 bool is_retransmit); 385 bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const; 386 387 Clock* clock_; 388 int64_t clock_delta_ms_; 389 Random random_ GUARDED_BY(send_critsect_); 390 391 rtc::scoped_ptr<BitrateAggregator> bitrates_; 392 Bitrate total_bitrate_sent_; 393 394 const bool audio_configured_; 395 rtc::scoped_ptr<RTPSenderAudio> audio_; 396 rtc::scoped_ptr<RTPSenderVideo> video_; 397 398 RtpPacketSender* const paced_sender_; 399 TransportSequenceNumberAllocator* const transport_sequence_number_allocator_; 400 TransportFeedbackObserver* const transport_feedback_observer_; 401 int64_t last_capture_time_ms_sent_; 402 rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_; 403 404 Transport *transport_; 405 bool sending_media_ GUARDED_BY(send_critsect_); 406 407 size_t max_payload_length_; 408 uint16_t packet_over_head_; 409 410 int8_t payload_type_ GUARDED_BY(send_critsect_); 411 std::map<int8_t, RtpUtility::Payload*> payload_type_map_; 412 413 RtpHeaderExtensionMap rtp_header_extension_map_; 414 int32_t transmission_time_offset_; 415 uint32_t absolute_send_time_; 416 VideoRotation rotation_; 417 CVOMode cvo_mode_; 418 uint16_t transport_sequence_number_; 419 420 // NACK 421 uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE]; 422 size_t nack_byte_count_[NACK_BYTECOUNT_SIZE]; 423 Bitrate nack_bitrate_; 424 425 RTPPacketHistory packet_history_; 426 427 // Statistics 428 rtc::scoped_ptr<CriticalSectionWrapper> statistics_crit_; 429 SendDelayMap send_delays_ GUARDED_BY(statistics_crit_); 430 FrameCounts frame_counts_ GUARDED_BY(statistics_crit_); 431 StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_); 432 StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_); 433 StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_); 434 FrameCountObserver* const frame_count_observer_; 435 SendSideDelayObserver* const send_side_delay_observer_; 436 437 // RTP variables 438 bool start_timestamp_forced_ GUARDED_BY(send_critsect_); 439 uint32_t start_timestamp_ GUARDED_BY(send_critsect_); 440 SSRCDatabase& ssrc_db_ GUARDED_BY(send_critsect_); 441 uint32_t remote_ssrc_ GUARDED_BY(send_critsect_); 442 bool sequence_number_forced_ GUARDED_BY(send_critsect_); 443 uint16_t sequence_number_ GUARDED_BY(send_critsect_); 444 uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_); 445 bool ssrc_forced_ GUARDED_BY(send_critsect_); 446 uint32_t ssrc_ GUARDED_BY(send_critsect_); 447 uint32_t timestamp_ GUARDED_BY(send_critsect_); 448 int64_t capture_time_ms_ GUARDED_BY(send_critsect_); 449 int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_); 450 bool media_has_been_sent_ GUARDED_BY(send_critsect_); 451 bool last_packet_marker_bit_ GUARDED_BY(send_critsect_); 452 std::vector<uint32_t> csrcs_ GUARDED_BY(send_critsect_); 453 int rtx_ GUARDED_BY(send_critsect_); 454 uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_); 455 // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that 456 // only understand one RTX PT is no longer needed. 457 int rtx_payload_type_ GUARDED_BY(send_critsect_); 458 // Mapping rtx_payload_type_map_[associated] = rtx. 459 std::map<int8_t, int8_t> rtx_payload_type_map_ GUARDED_BY(send_critsect_); 460 461 // Note: Don't access this variable directly, always go through 462 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember 463 // that by the time the function returns there is no guarantee 464 // that the target bitrate is still valid. 465 rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_; 466 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); 467 }; 468 469 } // namespace webrtc 470 471 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 472
