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/base/thread_annotations.h"
20 #include "webrtc/common_types.h"
21 #include "webrtc/modules/pacing/include/paced_sender.h"
22 #include "webrtc/modules/rtp_rtcp/interface/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/ssrc_database.h"
28 #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
30 #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1.
34 class BitrateAggregator;
35 class CriticalSectionWrapper;
39 class RTPSenderInterface {
41 RTPSenderInterface() {}
42 virtual ~RTPSenderInterface() {}
44 virtual uint32_t SSRC() const = 0;
45 virtual uint32_t Timestamp() const = 0;
47 virtual int32_t BuildRTPheader(uint8_t* data_buffer,
48 const int8_t payload_type,
49 const bool marker_bit,
50 const uint32_t capture_timestamp,
51 int64_t capture_time_ms,
52 const bool timestamp_provided = true,
53 const bool inc_sequence_number = true) = 0;
55 virtual uint16_t RTPHeaderLength() const = 0;
56 virtual uint16_t IncrementSequenceNumber() = 0;
57 virtual uint16_t SequenceNumber() const = 0;
58 virtual uint16_t MaxPayloadLength() const = 0;
59 virtual uint16_t MaxDataPayloadLength() const = 0;
60 virtual uint16_t PacketOverHead() const = 0;
61 virtual uint16_t ActualSendBitrateKbit() const = 0;
63 virtual int32_t SendToNetwork(
64 uint8_t *data_buffer, int payload_length, int rtp_header_length,
65 int64_t capture_time_ms, StorageType storage,
66 PacedSender::Priority priority) = 0;
69 class RTPSender : public RTPSenderInterface {
71 RTPSender(const int32_t id, const bool audio, Clock *clock,
72 Transport *transport, RtpAudioFeedback *audio_feedback,
73 PacedSender *paced_sender,
74 BitrateStatisticsObserver* bitrate_callback,
75 FrameCountObserver* frame_count_observer,
76 SendSideDelayObserver* send_side_delay_observer);
79 void ProcessBitrate();
81 virtual uint16_t ActualSendBitrateKbit() const OVERRIDE;
83 uint32_t VideoBitrateSent() const;
84 uint32_t FecOverheadRate() const;
85 uint32_t NackOverheadRate() const;
87 // Returns true if the statistics have been calculated, and false if no frame
88 // was sent within the statistics window.
89 bool GetSendSideDelay(int* avg_send_delay_ms, int* max_send_delay_ms) const;
91 void SetTargetBitrate(uint32_t bitrate);
92 uint32_t GetTargetBitrate();
94 virtual uint16_t MaxDataPayloadLength() const
95 OVERRIDE; // with RTP and FEC headers.
97 int32_t RegisterPayload(
98 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
99 const int8_t payload_type, const uint32_t frequency,
100 const uint8_t channels, const uint32_t rate);
102 int32_t DeRegisterSendPayload(const int8_t payload_type);
104 void SetSendPayloadType(int8_t payload_type);
106 int8_t SendPayloadType() const;
108 int SendPayloadFrequency() const;
110 void SetSendingStatus(bool enabled);
112 void SetSendingMediaStatus(const bool enabled);
113 bool SendingMedia() const;
115 void GetDataCounters(StreamDataCounters* rtp_stats,
116 StreamDataCounters* rtx_stats) const;
118 void ResetDataCounters();
120 uint32_t StartTimestamp() const;
121 void SetStartTimestamp(uint32_t timestamp, bool force);
123 uint32_t GenerateNewSSRC();
124 void SetSSRC(const uint32_t ssrc);
126 virtual uint16_t SequenceNumber() const OVERRIDE;
127 void SetSequenceNumber(uint16_t seq);
129 int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const;
131 void SetCSRCStatus(const bool include);
133 void SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize],
134 const uint8_t arr_length);
136 int32_t SetMaxPayloadLength(const uint16_t length,
137 const uint16_t packet_over_head);
139 int32_t SendOutgoingData(const FrameType frame_type,
140 const int8_t payload_type,
141 const uint32_t timestamp,
142 int64_t capture_time_ms,
143 const uint8_t* payload_data,
144 const uint32_t payload_size,
145 const RTPFragmentationHeader* fragmentation,
146 VideoCodecInformation* codec_info = NULL,
147 const RTPVideoTypeHeader* rtp_type_hdr = NULL);
149 // RTP header extension
150 int32_t SetTransmissionTimeOffset(
151 const int32_t transmission_time_offset);
152 int32_t SetAbsoluteSendTime(
153 const uint32_t absolute_send_time);
155 int32_t RegisterRtpHeaderExtension(const RTPExtensionType type,
158 int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type);
160 uint16_t RtpHeaderExtensionTotalLength() const;
162 uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer) const;
164 uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const;
165 uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
166 uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const;
168 bool UpdateAudioLevel(uint8_t *rtp_packet,
169 const uint16_t rtp_packet_length,
170 const RTPHeader &rtp_header,
171 const bool is_voiced,
172 const uint8_t dBov) const;
174 bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
175 bool retransmission);
176 int TimeToSendPadding(int bytes);
179 int SelectiveRetransmissions() const;
180 int SetSelectiveRetransmissions(uint8_t settings);
181 void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
182 const uint16_t avg_rtt);
184 void SetStorePacketsStatus(const bool enable,
185 const uint16_t number_to_store);
187 bool StorePackets() const;
189 int32_t ReSendPacket(uint16_t packet_id, uint32_t min_resend_time = 0);
191 bool ProcessNACKBitRate(const uint32_t now);
194 void SetRTXStatus(int mode);
196 void RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const;
198 uint32_t RtxSsrc() const;
199 void SetRtxSsrc(uint32_t ssrc);
201 void SetRtxPayloadType(int payloadType);
203 // Functions wrapping RTPSenderInterface.
204 virtual int32_t BuildRTPheader(
205 uint8_t* data_buffer,
206 const int8_t payload_type,
207 const bool marker_bit,
208 const uint32_t capture_timestamp,
209 int64_t capture_time_ms,
210 const bool timestamp_provided = true,
211 const bool inc_sequence_number = true) OVERRIDE;
213 virtual uint16_t RTPHeaderLength() const OVERRIDE;
214 virtual uint16_t IncrementSequenceNumber() OVERRIDE;
215 virtual uint16_t MaxPayloadLength() const OVERRIDE;
216 virtual uint16_t PacketOverHead() const OVERRIDE;
218 // Current timestamp.
219 virtual uint32_t Timestamp() const OVERRIDE;
220 virtual uint32_t SSRC() const OVERRIDE;
222 virtual int32_t SendToNetwork(
223 uint8_t *data_buffer, int payload_length, int rtp_header_length,
224 int64_t capture_time_ms, StorageType storage,
225 PacedSender::Priority priority) OVERRIDE;
229 // Send a DTMF tone using RFC 2833 (4733).
230 int32_t SendTelephoneEvent(const uint8_t key,
231 const uint16_t time_ms,
232 const uint8_t level);
234 bool SendTelephoneEventActive(int8_t *telephone_event) const;
236 // Set audio packet size, used to determine when it's time to send a DTMF
237 // packet in silence (CNG).
238 int32_t SetAudioPacketSize(const uint16_t packet_size_samples);
240 // Store the audio level in d_bov for
241 // header-extension-for-audio-level-indication.
242 int32_t SetAudioLevel(const uint8_t level_d_bov);
244 // Set payload type for Redundant Audio Data RFC 2198.
245 int32_t SetRED(const int8_t payload_type);
247 // Get payload type for Redundant Audio Data RFC 2198.
248 int32_t RED(int8_t *payload_type) const;
251 VideoCodecInformation *CodecInformationVideo();
253 RtpVideoCodecTypes VideoCodecType() const;
255 uint32_t MaxConfiguredBitrateVideo() const;
257 int32_t SendRTPIntraRequest();
260 int32_t SetGenericFECStatus(const bool enable,
261 const uint8_t payload_type_red,
262 const uint8_t payload_type_fec);
264 int32_t GenericFECStatus(bool *enable, uint8_t *payload_type_red,
265 uint8_t *payload_type_fec) const;
267 int32_t SetFecParameters(const FecProtectionParams *delta_params,
268 const FecProtectionParams *key_params);
270 int SendPadData(uint32_t timestamp,
271 int64_t capture_time_ms,
274 // Called on update of RTP statistics.
275 void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
276 StreamDataCountersCallback* GetRtpStatisticsCallback() const;
278 uint32_t BitrateSent() const;
280 void SetRtpState(const RtpState& rtp_state);
281 RtpState GetRtpState() const;
282 void SetRtxRtpState(const RtpState& rtp_state);
283 RtpState GetRtxRtpState() const;
286 int32_t CheckPayloadType(const int8_t payload_type,
287 RtpVideoCodecTypes *video_type);
290 // Maps capture time in milliseconds to send-side delay in milliseconds.
291 // Send-side delay is the difference between transmission time and capture
293 typedef std::map<int64_t, int> SendDelayMap;
295 int CreateRTPHeader(uint8_t* header, int8_t payload_type,
296 uint32_t ssrc, bool marker_bit,
297 uint32_t timestamp, uint16_t sequence_number,
298 const uint32_t* csrcs, uint8_t csrcs_length) const;
300 void UpdateNACKBitRate(const uint32_t bytes, const uint32_t now);
302 bool PrepareAndSendPacket(uint8_t* buffer,
304 int64_t capture_time_ms,
308 // Return the number of bytes sent.
309 int TrySendRedundantPayloads(int bytes);
310 int TrySendPadData(int bytes);
312 int BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes);
314 void BuildRtxPacket(uint8_t* buffer, uint16_t* length,
315 uint8_t* buffer_rtx);
317 bool SendPacketToNetwork(const uint8_t *packet, uint32_t size);
319 void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
321 void UpdateTransmissionTimeOffset(uint8_t *rtp_packet,
322 const uint16_t rtp_packet_length,
323 const RTPHeader &rtp_header,
324 const int64_t time_diff_ms) const;
325 void UpdateAbsoluteSendTime(uint8_t *rtp_packet,
326 const uint16_t rtp_packet_length,
327 const RTPHeader &rtp_header,
328 const int64_t now_ms) const;
330 void UpdateRtpStats(const uint8_t* buffer,
332 const RTPHeader& header,
335 bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
338 int64_t clock_delta_ms_;
340 scoped_ptr<BitrateAggregator> bitrates_;
341 Bitrate total_bitrate_sent_;
344 const bool audio_configured_;
345 RTPSenderAudio *audio_;
346 RTPSenderVideo *video_;
348 PacedSender *paced_sender_;
349 int64_t last_capture_time_ms_sent_;
350 CriticalSectionWrapper *send_critsect_;
352 Transport *transport_;
353 bool sending_media_ GUARDED_BY(send_critsect_);
355 uint16_t max_payload_length_;
356 uint16_t packet_over_head_;
358 int8_t payload_type_ GUARDED_BY(send_critsect_);
359 std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
361 RtpHeaderExtensionMap rtp_header_extension_map_;
362 int32_t transmission_time_offset_;
363 uint32_t absolute_send_time_;
366 uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
367 int32_t nack_byte_count_[NACK_BYTECOUNT_SIZE];
368 Bitrate nack_bitrate_;
370 RTPPacketHistory packet_history_;
373 scoped_ptr<CriticalSectionWrapper> statistics_crit_;
374 SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
375 std::map<FrameType, uint32_t> frame_counts_ GUARDED_BY(statistics_crit_);
376 StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
377 StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
378 StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
379 FrameCountObserver* const frame_count_observer_;
380 SendSideDelayObserver* const send_side_delay_observer_;
383 bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
384 uint32_t start_timestamp_ GUARDED_BY(send_critsect_);
385 SSRCDatabase& ssrc_db_ GUARDED_BY(send_critsect_);
386 uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
387 bool sequence_number_forced_ GUARDED_BY(send_critsect_);
388 uint16_t sequence_number_ GUARDED_BY(send_critsect_);
389 uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
390 bool ssrc_forced_ GUARDED_BY(send_critsect_);
391 uint32_t ssrc_ GUARDED_BY(send_critsect_);
392 uint32_t timestamp_ GUARDED_BY(send_critsect_);
393 int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
394 int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
395 bool media_has_been_sent_ GUARDED_BY(send_critsect_);
396 bool last_packet_marker_bit_ GUARDED_BY(send_critsect_);
397 uint8_t num_csrcs_ GUARDED_BY(send_critsect_);
398 uint32_t csrcs_[kRtpCsrcSize] GUARDED_BY(send_critsect_);
399 bool include_csrcs_ GUARDED_BY(send_critsect_);
400 int rtx_ GUARDED_BY(send_critsect_);
401 uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
402 int payload_type_rtx_ GUARDED_BY(send_critsect_);
404 // Note: Don't access this variable directly, always go through
405 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
406 // that by the time the function returns there is no guarantee
407 // that the target bitrate is still valid.
408 scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
409 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
412 } // namespace webrtc
414 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_