2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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.
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
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"
30 #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1.
34 class CriticalSectionWrapper;
38 class RTPSenderInterface {
40 RTPSenderInterface() {}
41 virtual ~RTPSenderInterface() {}
43 virtual uint32_t SSRC() const = 0;
44 virtual uint32_t Timestamp() const = 0;
46 virtual int32_t BuildRTPheader(uint8_t* data_buffer,
47 const int8_t payload_type,
48 const bool marker_bit,
49 const uint32_t capture_timestamp,
50 int64_t capture_time_ms,
51 const bool timestamp_provided = true,
52 const bool inc_sequence_number = true) = 0;
54 virtual uint16_t RTPHeaderLength() const = 0;
55 virtual uint16_t IncrementSequenceNumber() = 0;
56 virtual uint16_t SequenceNumber() const = 0;
57 virtual uint16_t MaxPayloadLength() const = 0;
58 virtual uint16_t MaxDataPayloadLength() const = 0;
59 virtual uint16_t PacketOverHead() const = 0;
60 virtual uint16_t ActualSendBitrateKbit() const = 0;
62 virtual int32_t SendToNetwork(
63 uint8_t *data_buffer, int payload_length, int rtp_header_length,
64 int64_t capture_time_ms, StorageType storage,
65 PacedSender::Priority priority) = 0;
68 class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
70 RTPSender(const int32_t id, const bool audio, Clock *clock,
71 Transport *transport, RtpAudioFeedback *audio_feedback,
72 PacedSender *paced_sender,
73 BitrateStatisticsObserver* bitrate_callback,
74 FrameCountObserver* frame_count_observer,
75 SendSideDelayObserver* send_side_delay_observer);
78 void ProcessBitrate();
80 virtual uint16_t ActualSendBitrateKbit() const OVERRIDE;
82 uint32_t VideoBitrateSent() const;
83 uint32_t FecOverheadRate() const;
84 uint32_t NackOverheadRate() const;
86 // Returns true if the statistics have been calculated, and false if no frame
87 // was sent within the statistics window.
88 bool GetSendSideDelay(int* avg_send_delay_ms, int* max_send_delay_ms) const;
90 void SetTargetBitrate(uint32_t bitrate);
91 uint32_t GetTargetBitrate();
93 virtual uint16_t MaxDataPayloadLength() const
94 OVERRIDE; // with RTP and FEC headers.
96 int32_t RegisterPayload(
97 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
98 const int8_t payload_type, const uint32_t frequency,
99 const uint8_t channels, const uint32_t rate);
101 int32_t DeRegisterSendPayload(const int8_t payload_type);
103 int8_t SendPayloadType() const;
105 int SendPayloadFrequency() const;
107 void SetSendingStatus(bool enabled);
109 void SetSendingMediaStatus(const bool enabled);
110 bool SendingMedia() const;
112 void GetDataCounters(StreamDataCounters* rtp_stats,
113 StreamDataCounters* rtx_stats) const;
115 void ResetDataCounters();
117 uint32_t StartTimestamp() const;
118 void SetStartTimestamp(uint32_t timestamp, bool force);
120 uint32_t GenerateNewSSRC();
121 void SetSSRC(const uint32_t ssrc);
123 virtual uint16_t SequenceNumber() const OVERRIDE;
124 void SetSequenceNumber(uint16_t seq);
126 int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const;
128 void SetCSRCStatus(const bool include);
130 void SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize],
131 const uint8_t arr_length);
133 int32_t SetMaxPayloadLength(const uint16_t length,
134 const uint16_t packet_over_head);
136 int32_t SendOutgoingData(const FrameType frame_type,
137 const int8_t payload_type,
138 const uint32_t timestamp,
139 int64_t capture_time_ms,
140 const uint8_t* payload_data,
141 const uint32_t payload_size,
142 const RTPFragmentationHeader* fragmentation,
143 VideoCodecInformation* codec_info = NULL,
144 const RTPVideoTypeHeader* rtp_type_hdr = NULL);
146 // RTP header extension
147 int32_t SetTransmissionTimeOffset(
148 const int32_t transmission_time_offset);
149 int32_t SetAbsoluteSendTime(
150 const uint32_t absolute_send_time);
152 int32_t RegisterRtpHeaderExtension(const RTPExtensionType type,
155 int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type);
157 uint16_t RtpHeaderExtensionTotalLength() const;
159 uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer) const;
161 uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const;
162 uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
163 uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const;
165 bool UpdateAudioLevel(uint8_t *rtp_packet,
166 const uint16_t rtp_packet_length,
167 const RTPHeader &rtp_header,
168 const bool is_voiced,
169 const uint8_t dBov) const;
171 bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
172 bool retransmission);
173 int TimeToSendPadding(int bytes);
176 int SelectiveRetransmissions() const;
177 int SetSelectiveRetransmissions(uint8_t settings);
178 void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
179 const uint16_t avg_rtt);
181 void SetStorePacketsStatus(const bool enable,
182 const uint16_t number_to_store);
184 bool StorePackets() const;
186 int32_t ReSendPacket(uint16_t packet_id, uint32_t min_resend_time = 0);
188 bool ProcessNACKBitRate(const uint32_t now);
191 void SetRTXStatus(int mode);
193 void RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const;
195 uint32_t RtxSsrc() const;
196 void SetRtxSsrc(uint32_t ssrc);
198 void SetRtxPayloadType(int payloadType);
200 // Functions wrapping RTPSenderInterface.
201 virtual int32_t BuildRTPheader(
202 uint8_t* data_buffer,
203 const int8_t payload_type,
204 const bool marker_bit,
205 const uint32_t capture_timestamp,
206 int64_t capture_time_ms,
207 const bool timestamp_provided = true,
208 const bool inc_sequence_number = true) OVERRIDE;
210 virtual uint16_t RTPHeaderLength() const OVERRIDE;
211 virtual uint16_t IncrementSequenceNumber() OVERRIDE;
212 virtual uint16_t MaxPayloadLength() const OVERRIDE;
213 virtual uint16_t PacketOverHead() const OVERRIDE;
215 // Current timestamp.
216 virtual uint32_t Timestamp() const OVERRIDE;
217 virtual uint32_t SSRC() const OVERRIDE;
219 virtual int32_t SendToNetwork(
220 uint8_t *data_buffer, int payload_length, int rtp_header_length,
221 int64_t capture_time_ms, StorageType storage,
222 PacedSender::Priority priority) OVERRIDE;
226 // Send a DTMF tone using RFC 2833 (4733).
227 int32_t SendTelephoneEvent(const uint8_t key,
228 const uint16_t time_ms,
229 const uint8_t level);
231 bool SendTelephoneEventActive(int8_t *telephone_event) const;
233 // Set audio packet size, used to determine when it's time to send a DTMF
234 // packet in silence (CNG).
235 int32_t SetAudioPacketSize(const uint16_t packet_size_samples);
237 // Store the audio level in d_bov for
238 // header-extension-for-audio-level-indication.
239 int32_t SetAudioLevel(const uint8_t level_d_bov);
241 // Set payload type for Redundant Audio Data RFC 2198.
242 int32_t SetRED(const int8_t payload_type);
244 // Get payload type for Redundant Audio Data RFC 2198.
245 int32_t RED(int8_t *payload_type) const;
248 VideoCodecInformation *CodecInformationVideo();
250 RtpVideoCodecTypes VideoCodecType() const;
252 uint32_t MaxConfiguredBitrateVideo() const;
254 int32_t SendRTPIntraRequest();
257 int32_t SetGenericFECStatus(const bool enable,
258 const uint8_t payload_type_red,
259 const uint8_t payload_type_fec);
261 int32_t GenericFECStatus(bool *enable, uint8_t *payload_type_red,
262 uint8_t *payload_type_fec) const;
264 int32_t SetFecParameters(const FecProtectionParams *delta_params,
265 const FecProtectionParams *key_params);
267 int SendPadData(uint32_t timestamp,
268 int64_t capture_time_ms,
271 // Called on update of RTP statistics.
272 void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
273 StreamDataCountersCallback* GetRtpStatisticsCallback() const;
275 uint32_t BitrateSent() const;
277 virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE;
279 void SetRtpState(const RtpState& rtp_state);
280 RtpState GetRtpState() const;
281 void SetRtxRtpState(const RtpState& rtp_state);
282 RtpState GetRtxRtpState() const;
285 int32_t CheckPayloadType(const int8_t payload_type,
286 RtpVideoCodecTypes *video_type);
289 // Maps capture time in milliseconds to send-side delay in milliseconds.
290 // Send-side delay is the difference between transmission time and capture
292 typedef std::map<int64_t, int> SendDelayMap;
294 int CreateRTPHeader(uint8_t* header, int8_t payload_type,
295 uint32_t ssrc, bool marker_bit,
296 uint32_t timestamp, uint16_t sequence_number,
297 const uint32_t* csrcs, uint8_t csrcs_length) const;
299 void UpdateNACKBitRate(const uint32_t bytes, const uint32_t now);
301 bool PrepareAndSendPacket(uint8_t* buffer,
303 int64_t capture_time_ms,
307 // Return the number of bytes sent.
308 int TrySendRedundantPayloads(int bytes);
309 int TrySendPadData(int bytes);
311 int BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes);
313 void BuildRtxPacket(uint8_t* buffer, uint16_t* length,
314 uint8_t* buffer_rtx);
316 bool SendPacketToNetwork(const uint8_t *packet, uint32_t size);
318 void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
320 void UpdateTransmissionTimeOffset(uint8_t *rtp_packet,
321 const uint16_t rtp_packet_length,
322 const RTPHeader &rtp_header,
323 const int64_t time_diff_ms) const;
324 void UpdateAbsoluteSendTime(uint8_t *rtp_packet,
325 const uint16_t rtp_packet_length,
326 const RTPHeader &rtp_header,
327 const int64_t now_ms) const;
329 void UpdateRtpStats(const uint8_t* buffer,
331 const RTPHeader& header,
334 bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
337 Bitrate bitrate_sent_;
340 const bool audio_configured_;
341 RTPSenderAudio *audio_;
342 RTPSenderVideo *video_;
344 PacedSender *paced_sender_;
345 CriticalSectionWrapper *send_critsect_;
347 Transport *transport_;
348 bool sending_media_ GUARDED_BY(send_critsect_);
350 uint16_t max_payload_length_;
351 uint16_t packet_over_head_;
353 int8_t payload_type_ GUARDED_BY(send_critsect_);
354 std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
356 RtpHeaderExtensionMap rtp_header_extension_map_;
357 int32_t transmission_time_offset_;
358 uint32_t absolute_send_time_;
361 uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
362 int32_t nack_byte_count_[NACK_BYTECOUNT_SIZE];
363 Bitrate nack_bitrate_;
365 RTPPacketHistory packet_history_;
368 scoped_ptr<CriticalSectionWrapper> statistics_crit_;
369 SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
370 std::map<FrameType, uint32_t> frame_counts_ GUARDED_BY(statistics_crit_);
371 StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
372 StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
373 StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
374 BitrateStatisticsObserver* const bitrate_callback_;
375 FrameCountObserver* const frame_count_observer_;
376 SendSideDelayObserver* const send_side_delay_observer_;
379 bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
380 uint32_t start_timestamp_ GUARDED_BY(send_critsect_);
381 SSRCDatabase& ssrc_db_ GUARDED_BY(send_critsect_);
382 uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
383 bool sequence_number_forced_ GUARDED_BY(send_critsect_);
384 uint16_t sequence_number_ GUARDED_BY(send_critsect_);
385 uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
386 bool ssrc_forced_ GUARDED_BY(send_critsect_);
387 uint32_t ssrc_ GUARDED_BY(send_critsect_);
388 uint32_t timestamp_ GUARDED_BY(send_critsect_);
389 int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
390 int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
391 bool media_has_been_sent_ GUARDED_BY(send_critsect_);
392 bool last_packet_marker_bit_ GUARDED_BY(send_critsect_);
393 uint8_t num_csrcs_ GUARDED_BY(send_critsect_);
394 uint32_t csrcs_[kRtpCsrcSize] GUARDED_BY(send_critsect_);
395 bool include_csrcs_ GUARDED_BY(send_critsect_);
396 int rtx_ GUARDED_BY(send_critsect_);
397 uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
398 int payload_type_rtx_ GUARDED_BY(send_critsect_);
400 // Note: Don't access this variable directly, always go through
401 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
402 // that by the time the function returns there is no guarantee
403 // that the target bitrate is still valid.
404 scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
405 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
408 } // namespace webrtc
410 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_