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 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 void SetSendPayloadType(int8_t payload_type);
105 int8_t SendPayloadType() const;
107 int SendPayloadFrequency() const;
109 void SetSendingStatus(bool enabled);
111 void SetSendingMediaStatus(const bool enabled);
112 bool SendingMedia() const;
114 void GetDataCounters(StreamDataCounters* rtp_stats,
115 StreamDataCounters* rtx_stats) const;
117 void ResetDataCounters();
119 uint32_t StartTimestamp() const;
120 void SetStartTimestamp(uint32_t timestamp, bool force);
122 uint32_t GenerateNewSSRC();
123 void SetSSRC(const uint32_t ssrc);
125 virtual uint16_t SequenceNumber() const OVERRIDE;
126 void SetSequenceNumber(uint16_t seq);
128 int32_t CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const;
130 void SetCSRCStatus(const bool include);
132 void SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize],
133 const uint8_t arr_length);
135 int32_t SetMaxPayloadLength(const uint16_t length,
136 const uint16_t packet_over_head);
138 int32_t SendOutgoingData(const FrameType frame_type,
139 const int8_t payload_type,
140 const uint32_t timestamp,
141 int64_t capture_time_ms,
142 const uint8_t* payload_data,
143 const uint32_t payload_size,
144 const RTPFragmentationHeader* fragmentation,
145 VideoCodecInformation* codec_info = NULL,
146 const RTPVideoTypeHeader* rtp_type_hdr = NULL);
148 // RTP header extension
149 int32_t SetTransmissionTimeOffset(
150 const int32_t transmission_time_offset);
151 int32_t SetAbsoluteSendTime(
152 const uint32_t absolute_send_time);
154 int32_t RegisterRtpHeaderExtension(const RTPExtensionType type,
157 int32_t DeregisterRtpHeaderExtension(const RTPExtensionType type);
159 uint16_t RtpHeaderExtensionTotalLength() const;
161 uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer) const;
163 uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const;
164 uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
165 uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const;
167 bool UpdateAudioLevel(uint8_t *rtp_packet,
168 const uint16_t rtp_packet_length,
169 const RTPHeader &rtp_header,
170 const bool is_voiced,
171 const uint8_t dBov) const;
173 bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
174 bool retransmission);
175 int TimeToSendPadding(int bytes);
178 int SelectiveRetransmissions() const;
179 int SetSelectiveRetransmissions(uint8_t settings);
180 void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
181 const uint16_t avg_rtt);
183 void SetStorePacketsStatus(const bool enable,
184 const uint16_t number_to_store);
186 bool StorePackets() const;
188 int32_t ReSendPacket(uint16_t packet_id, uint32_t min_resend_time = 0);
190 bool ProcessNACKBitRate(const uint32_t now);
193 void SetRTXStatus(int mode);
195 void RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const;
197 uint32_t RtxSsrc() const;
198 void SetRtxSsrc(uint32_t ssrc);
200 void SetRtxPayloadType(int payloadType);
202 // Functions wrapping RTPSenderInterface.
203 virtual int32_t BuildRTPheader(
204 uint8_t* data_buffer,
205 const int8_t payload_type,
206 const bool marker_bit,
207 const uint32_t capture_timestamp,
208 int64_t capture_time_ms,
209 const bool timestamp_provided = true,
210 const bool inc_sequence_number = true) OVERRIDE;
212 virtual uint16_t RTPHeaderLength() const OVERRIDE;
213 virtual uint16_t IncrementSequenceNumber() OVERRIDE;
214 virtual uint16_t MaxPayloadLength() const OVERRIDE;
215 virtual uint16_t PacketOverHead() const OVERRIDE;
217 // Current timestamp.
218 virtual uint32_t Timestamp() const OVERRIDE;
219 virtual uint32_t SSRC() const OVERRIDE;
221 virtual int32_t SendToNetwork(
222 uint8_t *data_buffer, int payload_length, int rtp_header_length,
223 int64_t capture_time_ms, StorageType storage,
224 PacedSender::Priority priority) OVERRIDE;
228 // Send a DTMF tone using RFC 2833 (4733).
229 int32_t SendTelephoneEvent(const uint8_t key,
230 const uint16_t time_ms,
231 const uint8_t level);
233 bool SendTelephoneEventActive(int8_t *telephone_event) const;
235 // Set audio packet size, used to determine when it's time to send a DTMF
236 // packet in silence (CNG).
237 int32_t SetAudioPacketSize(const uint16_t packet_size_samples);
239 // Store the audio level in d_bov for
240 // header-extension-for-audio-level-indication.
241 int32_t SetAudioLevel(const uint8_t level_d_bov);
243 // Set payload type for Redundant Audio Data RFC 2198.
244 int32_t SetRED(const int8_t payload_type);
246 // Get payload type for Redundant Audio Data RFC 2198.
247 int32_t RED(int8_t *payload_type) const;
250 VideoCodecInformation *CodecInformationVideo();
252 RtpVideoCodecTypes VideoCodecType() const;
254 uint32_t MaxConfiguredBitrateVideo() const;
256 int32_t SendRTPIntraRequest();
259 int32_t SetGenericFECStatus(const bool enable,
260 const uint8_t payload_type_red,
261 const uint8_t payload_type_fec);
263 int32_t GenericFECStatus(bool *enable, uint8_t *payload_type_red,
264 uint8_t *payload_type_fec) const;
266 int32_t SetFecParameters(const FecProtectionParams *delta_params,
267 const FecProtectionParams *key_params);
269 int SendPadData(uint32_t timestamp,
270 int64_t capture_time_ms,
273 // Called on update of RTP statistics.
274 void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
275 StreamDataCountersCallback* GetRtpStatisticsCallback() const;
277 uint32_t BitrateSent() const;
279 virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE;
281 void SetRtpState(const RtpState& rtp_state);
282 RtpState GetRtpState() const;
283 void SetRtxRtpState(const RtpState& rtp_state);
284 RtpState GetRtxRtpState() const;
287 int32_t CheckPayloadType(const int8_t payload_type,
288 RtpVideoCodecTypes *video_type);
291 // Maps capture time in milliseconds to send-side delay in milliseconds.
292 // Send-side delay is the difference between transmission time and capture
294 typedef std::map<int64_t, int> SendDelayMap;
296 int CreateRTPHeader(uint8_t* header, int8_t payload_type,
297 uint32_t ssrc, bool marker_bit,
298 uint32_t timestamp, uint16_t sequence_number,
299 const uint32_t* csrcs, uint8_t csrcs_length) const;
301 void UpdateNACKBitRate(const uint32_t bytes, const uint32_t now);
303 bool PrepareAndSendPacket(uint8_t* buffer,
305 int64_t capture_time_ms,
309 // Return the number of bytes sent.
310 int TrySendRedundantPayloads(int bytes);
311 int TrySendPadData(int bytes);
313 int BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes);
315 void BuildRtxPacket(uint8_t* buffer, uint16_t* length,
316 uint8_t* buffer_rtx);
318 bool SendPacketToNetwork(const uint8_t *packet, uint32_t size);
320 void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
322 void UpdateTransmissionTimeOffset(uint8_t *rtp_packet,
323 const uint16_t rtp_packet_length,
324 const RTPHeader &rtp_header,
325 const int64_t time_diff_ms) const;
326 void UpdateAbsoluteSendTime(uint8_t *rtp_packet,
327 const uint16_t rtp_packet_length,
328 const RTPHeader &rtp_header,
329 const int64_t now_ms) const;
331 void UpdateRtpStats(const uint8_t* buffer,
333 const RTPHeader& header,
336 bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
339 Bitrate bitrate_sent_;
342 const bool audio_configured_;
343 RTPSenderAudio *audio_;
344 RTPSenderVideo *video_;
346 PacedSender *paced_sender_;
347 CriticalSectionWrapper *send_critsect_;
349 Transport *transport_;
350 bool sending_media_ GUARDED_BY(send_critsect_);
352 uint16_t max_payload_length_;
353 uint16_t packet_over_head_;
355 int8_t payload_type_ GUARDED_BY(send_critsect_);
356 std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
358 RtpHeaderExtensionMap rtp_header_extension_map_;
359 int32_t transmission_time_offset_;
360 uint32_t absolute_send_time_;
363 uint32_t nack_byte_count_times_[NACK_BYTECOUNT_SIZE];
364 int32_t nack_byte_count_[NACK_BYTECOUNT_SIZE];
365 Bitrate nack_bitrate_;
367 RTPPacketHistory packet_history_;
370 scoped_ptr<CriticalSectionWrapper> statistics_crit_;
371 SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
372 std::map<FrameType, uint32_t> frame_counts_ GUARDED_BY(statistics_crit_);
373 StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
374 StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
375 StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
376 BitrateStatisticsObserver* const bitrate_callback_;
377 FrameCountObserver* const frame_count_observer_;
378 SendSideDelayObserver* const send_side_delay_observer_;
381 bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
382 uint32_t start_timestamp_ GUARDED_BY(send_critsect_);
383 SSRCDatabase& ssrc_db_ GUARDED_BY(send_critsect_);
384 uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
385 bool sequence_number_forced_ GUARDED_BY(send_critsect_);
386 uint16_t sequence_number_ GUARDED_BY(send_critsect_);
387 uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
388 bool ssrc_forced_ GUARDED_BY(send_critsect_);
389 uint32_t ssrc_ GUARDED_BY(send_critsect_);
390 uint32_t timestamp_ GUARDED_BY(send_critsect_);
391 int64_t capture_time_ms_ GUARDED_BY(send_critsect_);
392 int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);
393 bool media_has_been_sent_ GUARDED_BY(send_critsect_);
394 bool last_packet_marker_bit_ GUARDED_BY(send_critsect_);
395 uint8_t num_csrcs_ GUARDED_BY(send_critsect_);
396 uint32_t csrcs_[kRtpCsrcSize] GUARDED_BY(send_critsect_);
397 bool include_csrcs_ GUARDED_BY(send_critsect_);
398 int rtx_ GUARDED_BY(send_critsect_);
399 uint32_t ssrc_rtx_ GUARDED_BY(send_critsect_);
400 int payload_type_rtx_ GUARDED_BY(send_critsect_);
402 // Note: Don't access this variable directly, always go through
403 // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
404 // that by the time the function returns there is no guarantee
405 // that the target bitrate is still valid.
406 scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
407 uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
410 } // namespace webrtc
412 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_