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 <assert.h> 15 #include <math.h> 16 17 #include <map> 18 19 #include "webrtc/common_types.h" 20 #include "webrtc/modules/pacing/include/paced_sender.h" 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 22 #include "webrtc/modules/rtp_rtcp/source/bitrate.h" 23 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" 26 #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" 27 #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h" 28 #include "webrtc/system_wrappers/interface/thread_annotations.h" 29 30 #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1. 31 32 namespace webrtc { 33 34 class CriticalSectionWrapper; 35 class RTPSenderAudio; 36 class RTPSenderVideo; 37 38 class RTPSenderInterface { 39 public: 40 RTPSenderInterface() {} 41 virtual ~RTPSenderInterface() {} 42 43 virtual uint32_t SSRC() const = 0; 44 virtual uint32_t Timestamp() const = 0; 45 46 virtual int32_t BuildRTPheader( 47 uint8_t *data_buffer, const int8_t payload_type, 48 const bool marker_bit, const uint32_t capture_time_stamp, 49 int64_t capture_time_ms, 50 const bool time_stamp_provided = true, 51 const bool inc_sequence_number = true) = 0; 52 53 virtual uint16_t RTPHeaderLength() const = 0; 54 virtual uint16_t IncrementSequenceNumber() = 0; 55 virtual uint16_t SequenceNumber() const = 0; 56 virtual uint16_t MaxPayloadLength() const = 0; 57 virtual uint16_t MaxDataPayloadLength() const = 0; 58 virtual uint16_t PacketOverHead() const = 0; 59 virtual uint16_t ActualSendBitrateKbit() const = 0; 60 61 virtual int32_t SendToNetwork( 62 uint8_t *data_buffer, int payload_length, int rtp_header_length, 63 int64_t capture_time_ms, StorageType storage, 64 PacedSender::Priority priority) = 0; 65 }; 66 67 class RTPSender : public RTPSenderInterface, public Bitrate::Observer { 68 public: 69 RTPSender(const int32_t id, const bool audio, Clock *clock, 70 Transport *transport, RtpAudioFeedback *audio_feedback, 71 PacedSender *paced_sender); 72 virtual ~RTPSender(); 73 74 void ProcessBitrate(); 75 76 virtual uint16_t ActualSendBitrateKbit() const OVERRIDE; 77 78 uint32_t VideoBitrateSent() const; 79 uint32_t FecOverheadRate() const; 80 uint32_t NackOverheadRate() const; 81 82 // Returns true if the statistics have been calculated, and false if no frame 83 // was sent within the statistics window. 84 bool GetSendSideDelay(int* avg_send_delay_ms, int* max_send_delay_ms) const; 85 86 void SetTargetBitrate(uint32_t bitrate); 87 uint32_t GetTargetBitrate(); 88 89 virtual uint16_t MaxDataPayloadLength() const 90 OVERRIDE; // with RTP and FEC headers. 91 92 int32_t RegisterPayload( 93 const char payload_name[RTP_PAYLOAD_NAME_SIZE], 94 const int8_t payload_type, const uint32_t frequency, 95 const uint8_t channels, const uint32_t rate); 96 97 int32_t DeRegisterSendPayload(const int8_t payload_type); 98 99 int8_t SendPayloadType() const; 100 101 int SendPayloadFrequency() const; 102 103 void SetSendingStatus(bool enabled); 104 105 void SetSendingMediaStatus(const bool enabled); 106 bool SendingMedia() const; 107 108 // Number of sent RTP packets. 109 uint32_t Packets() const; 110 111 // Number of sent RTP bytes. 112 uint32_t Bytes() const; 113 114 void ResetDataCounters(); 115 116 uint32_t StartTimestamp() const; 117 void SetStartTimestamp(uint32_t timestamp, bool force); 118 119 uint32_t GenerateNewSSRC(); 120 void SetSSRC(const uint32_t ssrc); 121 122 virtual uint16_t SequenceNumber() const OVERRIDE; 123 void SetSequenceNumber(uint16_t seq); 124 125 int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const; 126 127 void SetCSRCStatus(const bool include); 128 129 void SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize], 130 const uint8_t arr_length); 131 132 int32_t SetMaxPayloadLength(const uint16_t length, 133 const uint16_t packet_over_head); 134 135 int32_t SendOutgoingData( 136 const FrameType frame_type, const int8_t payload_type, 137 const uint32_t time_stamp, int64_t capture_time_ms, 138 const uint8_t *payload_data, const uint32_t payload_size, 139 const RTPFragmentationHeader *fragmentation, 140 VideoCodecInformation *codec_info = NULL, 141 const RTPVideoTypeHeader * rtp_type_hdr = NULL); 142 143 // RTP header extension 144 int32_t SetTransmissionTimeOffset( 145 const int32_t transmission_time_offset); 146 int32_t SetAbsoluteSendTime( 147 const uint32_t absolute_send_time); 148 149 int32_t RegisterRtpHeaderExtension(const RTPExtensionType type, 150 const uint8_t id); 151 152 int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type); 153 154 uint16_t RtpHeaderExtensionTotalLength() const; 155 156 uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer) const; 157 158 uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const; 159 uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const; 160 uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const; 161 162 bool UpdateAudioLevel(uint8_t *rtp_packet, 163 const uint16_t rtp_packet_length, 164 const RTPHeader &rtp_header, 165 const bool is_voiced, 166 const uint8_t dBov) const; 167 168 bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms, 169 bool retransmission); 170 int TimeToSendPadding(int bytes); 171 172 // NACK. 173 int SelectiveRetransmissions() const; 174 int SetSelectiveRetransmissions(uint8_t settings); 175 void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers, 176 const uint16_t avg_rtt); 177 178 void SetStorePacketsStatus(const bool enable, 179 const uint16_t number_to_store); 180 181 bool StorePackets() const; 182 183 int32_t ReSendPacket(uint16_t packet_id, uint32_t min_resend_time = 0); 184 185 bool ProcessNACKBitRate(const uint32_t now); 186 187 // RTX. 188 void SetRTXStatus(int mode); 189 190 void RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const; 191 192 void SetRtxSsrc(uint32_t ssrc); 193 194 void SetRtxPayloadType(int payloadType); 195 196 // Functions wrapping RTPSenderInterface. 197 virtual int32_t BuildRTPheader( 198 uint8_t *data_buffer, const int8_t payload_type, 199 const bool marker_bit, const uint32_t capture_time_stamp, 200 int64_t capture_time_ms, 201 const bool time_stamp_provided = true, 202 const bool inc_sequence_number = true) OVERRIDE; 203 204 virtual uint16_t RTPHeaderLength() const OVERRIDE; 205 virtual uint16_t IncrementSequenceNumber() OVERRIDE; 206 virtual uint16_t MaxPayloadLength() const OVERRIDE; 207 virtual uint16_t PacketOverHead() const OVERRIDE; 208 209 // Current timestamp. 210 virtual uint32_t Timestamp() const OVERRIDE; 211 virtual uint32_t SSRC() const OVERRIDE; 212 213 virtual int32_t SendToNetwork( 214 uint8_t *data_buffer, int payload_length, int rtp_header_length, 215 int64_t capture_time_ms, StorageType storage, 216 PacedSender::Priority priority) OVERRIDE; 217 218 // Audio. 219 220 // Send a DTMF tone using RFC 2833 (4733). 221 int32_t SendTelephoneEvent(const uint8_t key, 222 const uint16_t time_ms, 223 const uint8_t level); 224 225 bool SendTelephoneEventActive(int8_t *telephone_event) const; 226 227 // Set audio packet size, used to determine when it's time to send a DTMF 228 // packet in silence (CNG). 229 int32_t SetAudioPacketSize(const uint16_t packet_size_samples); 230 231 // Store the audio level in d_bov for 232 // header-extension-for-audio-level-indication. 233 int32_t SetAudioLevel(const uint8_t level_d_bov); 234 235 // Set payload type for Redundant Audio Data RFC 2198. 236 int32_t SetRED(const int8_t payload_type); 237 238 // Get payload type for Redundant Audio Data RFC 2198. 239 int32_t RED(int8_t *payload_type) const; 240 241 // Video. 242 VideoCodecInformation *CodecInformationVideo(); 243 244 RtpVideoCodecTypes VideoCodecType() const; 245 246 uint32_t MaxConfiguredBitrateVideo() const; 247 248 int32_t SendRTPIntraRequest(); 249 250 // FEC. 251 int32_t SetGenericFECStatus(const bool enable, 252 const uint8_t payload_type_red, 253 const uint8_t payload_type_fec); 254 255 int32_t GenericFECStatus(bool *enable, uint8_t *payload_type_red, 256 uint8_t *payload_type_fec) const; 257 258 int32_t SetFecParameters(const FecProtectionParams *delta_params, 259 const FecProtectionParams *key_params); 260 261 virtual void RegisterFrameCountObserver(FrameCountObserver* observer); 262 virtual FrameCountObserver* GetFrameCountObserver() const; 263 264 int SendPadData(int payload_type, uint32_t timestamp, int64_t capture_time_ms, 265 int32_t bytes, StorageType store, 266 bool force_full_size_packets, bool only_pad_after_markerbit); 267 268 // Called on update of RTP statistics. 269 void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback); 270 StreamDataCountersCallback* GetRtpStatisticsCallback() const; 271 272 // Called on new send bitrate estimate. 273 void RegisterBitrateObserver(BitrateStatisticsObserver* observer); 274 BitrateStatisticsObserver* GetBitrateObserver() const; 275 276 uint32_t BitrateSent() const; 277 278 virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE; 279 280 protected: 281 int32_t CheckPayloadType(const int8_t payload_type, 282 RtpVideoCodecTypes *video_type); 283 284 private: 285 // Maps capture time in milliseconds to send-side delay in milliseconds. 286 // Send-side delay is the difference between transmission time and capture 287 // time. 288 typedef std::map<int64_t, int> SendDelayMap; 289 290 int CreateRTPHeader(uint8_t* header, int8_t payload_type, 291 uint32_t ssrc, bool marker_bit, 292 uint32_t timestamp, uint16_t sequence_number, 293 const uint32_t* csrcs, uint8_t csrcs_length) const; 294 295 void UpdateNACKBitRate(const uint32_t bytes, const uint32_t now); 296 297 bool PrepareAndSendPacket(uint8_t* buffer, 298 uint16_t length, 299 int64_t capture_time_ms, 300 bool send_over_rtx, 301 bool is_retransmit); 302 303 int SendRedundantPayloads(int payload_type, int bytes); 304 305 bool SendPaddingAccordingToBitrate(int8_t payload_type, 306 uint32_t capture_timestamp, 307 int64_t capture_time_ms); 308 int BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes); 309 310 void BuildRtxPacket(uint8_t* buffer, uint16_t* length, 311 uint8_t* buffer_rtx); 312 313 bool SendPacketToNetwork(const uint8_t *packet, uint32_t size); 314 315 void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms); 316 317 void UpdateTransmissionTimeOffset(uint8_t *rtp_packet, 318 const uint16_t rtp_packet_length, 319 const RTPHeader &rtp_header, 320 const int64_t time_diff_ms) const; 321 void UpdateAbsoluteSendTime(uint8_t *rtp_packet, 322 const uint16_t rtp_packet_length, 323 const RTPHeader &rtp_header, 324 const int64_t now_ms) const; 325 326 void UpdateRtpStats(const uint8_t* buffer, 327 uint32_t size, 328 const RTPHeader& header, 329 bool is_rtx, 330 bool is_retransmit); 331 bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const; 332 333 Clock* clock_; 334 Bitrate bitrate_sent_; 335 336 int32_t id_; 337 const bool audio_configured_; 338 RTPSenderAudio *audio_; 339 RTPSenderVideo *video_; 340 341 PacedSender *paced_sender_; 342 CriticalSectionWrapper *send_critsect_; 343 344 Transport *transport_; 345 bool sending_media_ GUARDED_BY(send_critsect_); 346 347 uint16_t max_payload_length_; 348 uint16_t packet_over_head_; 349 350 int8_t payload_type_ GUARDED_BY(send_critsect_); 351 std::map<int8_t, ModuleRTPUtility::Payload *> payload_type_map_; 352 353 RtpHeaderExtensionMap rtp_header_extension_map_; 354 int32_t transmission_time_offset_; 355 uint32_t absolute_send_time_; 356 357 // NACK 358 uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE]; 359 int32_t nack_byte_count_[NACK_BYTECOUNT_SIZE]; 360 Bitrate nack_bitrate_; 361 362 RTPPacketHistory packet_history_; 363 364 // Statistics 365 scoped_ptr<CriticalSectionWrapper> statistics_crit_; 366 SendDelayMap send_delays_; 367 std::map<FrameType, uint32_t> frame_counts_; 368 FrameCountObserver* frame_count_observer_; 369 StreamDataCounters rtp_stats_; 370 StreamDataCounters rtx_rtp_stats_; 371 StreamDataCountersCallback* rtp_stats_callback_; 372 BitrateStatisticsObserver* bitrate_callback_; 373 374 // RTP variables 375 bool start_time_stamp_forced_; 376 uint32_t start_time_stamp_; 377 SSRCDatabase &ssrc_db_; 378 uint32_t remote_ssrc_; 379 bool sequence_number_forced_; 380 uint16_t sequence_number_; 381 uint16_t sequence_number_rtx_; 382 bool ssrc_forced_; 383 uint32_t ssrc_; 384 uint32_t timestamp_; 385 int64_t capture_time_ms_; 386 int64_t last_timestamp_time_ms_; 387 bool last_packet_marker_bit_; 388 uint8_t num_csrcs_; 389 uint32_t csrcs_[kRtpCsrcSize]; 390 bool include_csrcs_; 391 int rtx_; 392 uint32_t ssrc_rtx_; 393 int payload_type_rtx_; 394 395 // Note: Don't access this variable directly, always go through 396 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember 397 // that by the time the function returns there is no guarantee 398 // that the target bitrate is still valid. 399 scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_; 400 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_); 401 }; 402 403 } // namespace webrtc 404 405 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_ 406
