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 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
13 #include <stdlib.h> // srand
15 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
16 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
17 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
18 #include "webrtc/system_wrappers/interface/trace.h"
19 #include "webrtc/system_wrappers/interface/trace_event.h"
23 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
24 const int kMaxPaddingLength = 224;
25 const int kSendSideDelayWindowMs = 1000;
29 const char* FrameTypeToString(const FrameType frame_type) {
31 case kFrameEmpty: return "empty";
32 case kAudioFrameSpeech: return "audio_speech";
33 case kAudioFrameCN: return "audio_cn";
34 case kVideoFrameKey: return "video_key";
35 case kVideoFrameDelta: return "video_delta";
42 RTPSender::RTPSender(const int32_t id,
46 RtpAudioFeedback* audio_feedback,
47 PacedSender* paced_sender)
49 bitrate_sent_(clock, this),
51 audio_configured_(audio),
54 paced_sender_(paced_sender),
55 send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
56 transport_(transport),
57 sending_media_(true), // Default to sending media.
58 max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
59 target_send_bitrate_(0),
60 packet_over_head_(28),
63 rtp_header_extension_map_(),
64 transmission_time_offset_(0),
65 absolute_send_time_(0),
67 nack_byte_count_times_(),
69 nack_bitrate_(clock, NULL),
70 packet_history_(clock),
72 statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
73 frame_count_observer_(NULL),
74 rtp_stats_callback_(NULL),
75 bitrate_callback_(NULL),
77 start_time_stamp_forced_(false),
79 ssrc_db_(*SSRCDatabase::GetSSRCDatabase()),
81 sequence_number_forced_(false),
85 last_timestamp_time_ms_(0),
86 last_packet_marker_bit_(false),
91 payload_type_rtx_(-1) {
92 memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
93 memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
94 memset(csrcs_, 0, sizeof(csrcs_));
95 // We need to seed the random generator.
96 srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
97 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
98 ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
99 // Random start, 16 bits. Can't be 0.
100 sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
101 sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
104 audio_ = new RTPSenderAudio(id, clock_, this);
105 audio_->RegisterAudioCallback(audio_feedback);
107 video_ = new RTPSenderVideo(id, clock_, this);
109 WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
112 RTPSender::~RTPSender() {
113 if (remote_ssrc_ != 0) {
114 ssrc_db_.ReturnSSRC(remote_ssrc_);
116 ssrc_db_.ReturnSSRC(ssrc_);
118 SSRCDatabase::ReturnSSRCDatabase();
119 delete send_critsect_;
120 while (!payload_type_map_.empty()) {
121 std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it =
122 payload_type_map_.begin();
124 payload_type_map_.erase(it);
129 WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id_, "%s deleted", __FUNCTION__);
132 void RTPSender::SetTargetSendBitrate(const uint32_t bits) {
133 target_send_bitrate_ = static_cast<uint16_t>(bits / 1000);
136 uint16_t RTPSender::ActualSendBitrateKbit() const {
137 return (uint16_t)(bitrate_sent_.BitrateNow() / 1000);
140 uint32_t RTPSender::VideoBitrateSent() const {
142 return video_->VideoBitrateSent();
147 uint32_t RTPSender::FecOverheadRate() const {
149 return video_->FecOverheadRate();
154 uint32_t RTPSender::NackOverheadRate() const {
155 return nack_bitrate_.BitrateLast();
158 bool RTPSender::GetSendSideDelay(int* avg_send_delay_ms,
159 int* max_send_delay_ms) const {
160 CriticalSectionScoped cs(statistics_crit_.get());
161 SendDelayMap::const_iterator it = send_delays_.upper_bound(
162 clock_->TimeInMilliseconds() - kSendSideDelayWindowMs);
163 if (!sending_media_ || it == send_delays_.end())
166 for (; it != send_delays_.end(); ++it) {
167 *max_send_delay_ms = std::max(*max_send_delay_ms, it->second);
168 *avg_send_delay_ms += it->second;
171 *avg_send_delay_ms = (*avg_send_delay_ms + num_delays / 2) / num_delays;
175 int32_t RTPSender::SetTransmissionTimeOffset(
176 const int32_t transmission_time_offset) {
177 if (transmission_time_offset > (0x800000 - 1) ||
178 transmission_time_offset < -(0x800000 - 1)) { // Word24.
181 CriticalSectionScoped cs(send_critsect_);
182 transmission_time_offset_ = transmission_time_offset;
186 int32_t RTPSender::SetAbsoluteSendTime(
187 const uint32_t absolute_send_time) {
188 if (absolute_send_time > 0xffffff) { // UWord24.
191 CriticalSectionScoped cs(send_critsect_);
192 absolute_send_time_ = absolute_send_time;
196 int32_t RTPSender::RegisterRtpHeaderExtension(const RTPExtensionType type,
198 CriticalSectionScoped cs(send_critsect_);
199 return rtp_header_extension_map_.Register(type, id);
202 int32_t RTPSender::DeregisterRtpHeaderExtension(
203 const RTPExtensionType type) {
204 CriticalSectionScoped cs(send_critsect_);
205 return rtp_header_extension_map_.Deregister(type);
208 uint16_t RTPSender::RtpHeaderExtensionTotalLength() const {
209 CriticalSectionScoped cs(send_critsect_);
210 return rtp_header_extension_map_.GetTotalLengthInBytes();
213 int32_t RTPSender::RegisterPayload(
214 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
215 const int8_t payload_number, const uint32_t frequency,
216 const uint8_t channels, const uint32_t rate) {
217 assert(payload_name);
218 CriticalSectionScoped cs(send_critsect_);
220 std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it =
221 payload_type_map_.find(payload_number);
223 if (payload_type_map_.end() != it) {
224 // We already use this payload type.
225 ModuleRTPUtility::Payload *payload = it->second;
228 // Check if it's the same as we already have.
229 if (ModuleRTPUtility::StringCompare(payload->name, payload_name,
230 RTP_PAYLOAD_NAME_SIZE - 1)) {
231 if (audio_configured_ && payload->audio &&
232 payload->typeSpecific.Audio.frequency == frequency &&
233 (payload->typeSpecific.Audio.rate == rate ||
234 payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
235 payload->typeSpecific.Audio.rate = rate;
236 // Ensure that we update the rate if new or old is zero.
239 if (!audio_configured_ && !payload->audio) {
245 int32_t ret_val = -1;
246 ModuleRTPUtility::Payload *payload = NULL;
247 if (audio_configured_) {
248 ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
249 frequency, channels, rate, payload);
251 ret_val = video_->RegisterVideoPayload(payload_name, payload_number, rate,
255 payload_type_map_[payload_number] = payload;
260 int32_t RTPSender::DeRegisterSendPayload(
261 const int8_t payload_type) {
262 CriticalSectionScoped lock(send_critsect_);
264 std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it =
265 payload_type_map_.find(payload_type);
267 if (payload_type_map_.end() == it) {
270 ModuleRTPUtility::Payload *payload = it->second;
272 payload_type_map_.erase(it);
276 int8_t RTPSender::SendPayloadType() const { return payload_type_; }
278 int RTPSender::SendPayloadFrequency() const {
279 return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
282 int32_t RTPSender::SetMaxPayloadLength(
283 const uint16_t max_payload_length,
284 const uint16_t packet_over_head) {
286 if (max_payload_length < 100 || max_payload_length > IP_PACKET_SIZE) {
287 WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "%s invalid argument",
291 CriticalSectionScoped cs(send_critsect_);
292 max_payload_length_ = max_payload_length;
293 packet_over_head_ = packet_over_head;
295 WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, id_, "SetMaxPayloadLength to %d.",
300 uint16_t RTPSender::MaxDataPayloadLength() const {
301 if (audio_configured_) {
302 return max_payload_length_ - RTPHeaderLength();
304 return max_payload_length_ - RTPHeaderLength() -
305 video_->FECPacketOverhead() - ((rtx_) ? 2 : 0);
306 // Include the FEC/ULP/RED overhead.
310 uint16_t RTPSender::MaxPayloadLength() const {
311 return max_payload_length_;
314 uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; }
316 void RTPSender::SetRTXStatus(int mode, bool set_ssrc, uint32_t ssrc) {
317 CriticalSectionScoped cs(send_critsect_);
319 if (rtx_ != kRtxOff) {
323 ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
328 void RTPSender::RTXStatus(int* mode, uint32_t* ssrc,
329 int* payload_type) const {
330 CriticalSectionScoped cs(send_critsect_);
333 *payload_type = payload_type_rtx_;
337 void RTPSender::SetRtxPayloadType(int payload_type) {
338 CriticalSectionScoped cs(send_critsect_);
339 payload_type_rtx_ = payload_type;
342 int32_t RTPSender::CheckPayloadType(const int8_t payload_type,
343 RtpVideoCodecTypes *video_type) {
344 CriticalSectionScoped cs(send_critsect_);
346 if (payload_type < 0) {
347 WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_, "\tinvalid payload_type (%d)",
351 if (audio_configured_) {
352 int8_t red_pl_type = -1;
353 if (audio_->RED(red_pl_type) == 0) {
354 // We have configured RED.
355 if (red_pl_type == payload_type) {
356 // And it's a match...
361 if (payload_type_ == payload_type) {
362 if (!audio_configured_) {
363 *video_type = video_->VideoCodecType();
367 std::map<int8_t, ModuleRTPUtility::Payload *>::iterator it =
368 payload_type_map_.find(payload_type);
369 if (it == payload_type_map_.end()) {
370 WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
371 "\tpayloadType:%d not registered", payload_type);
374 payload_type_ = payload_type;
375 ModuleRTPUtility::Payload *payload = it->second;
377 if (!payload->audio && !audio_configured_) {
378 video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
379 *video_type = payload->typeSpecific.Video.videoCodecType;
380 video_->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
385 int32_t RTPSender::SendOutgoingData(
386 const FrameType frame_type, const int8_t payload_type,
387 const uint32_t capture_timestamp, int64_t capture_time_ms,
388 const uint8_t *payload_data, const uint32_t payload_size,
389 const RTPFragmentationHeader *fragmentation,
390 VideoCodecInformation *codec_info, const RTPVideoTypeHeader *rtp_type_hdr) {
392 // Drop this packet if we're not sending media packets.
393 CriticalSectionScoped cs(send_critsect_);
394 if (!sending_media_) {
398 RtpVideoCodecTypes video_type = kRtpVideoGeneric;
399 if (CheckPayloadType(payload_type, &video_type) != 0) {
400 WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
401 "%s invalid argument failed to find payload_type:%d",
402 __FUNCTION__, payload_type);
407 if (audio_configured_) {
408 TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
409 "Send", "type", FrameTypeToString(frame_type));
410 assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
411 frame_type == kFrameEmpty);
413 ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
414 payload_data, payload_size, fragmentation);
416 TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
417 "Send", "type", FrameTypeToString(frame_type));
418 assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
420 if (frame_type == kFrameEmpty) {
421 if (paced_sender_->Enabled()) {
422 // Padding is driven by the pacer and not by the encoder.
425 return SendPaddingAccordingToBitrate(payload_type, capture_timestamp,
426 capture_time_ms) ? 0 : -1;
428 ret_val = video_->SendVideo(video_type, frame_type, payload_type,
429 capture_timestamp, capture_time_ms,
430 payload_data, payload_size,
431 fragmentation, codec_info,
436 CriticalSectionScoped cs(statistics_crit_.get());
437 uint32_t frame_count = ++frame_counts_[frame_type];
438 if (frame_count_observer_) {
439 frame_count_observer_->FrameCountUpdated(frame_type,
447 int RTPSender::SendRedundantPayloads(int payload_type, int bytes_to_send) {
448 if (!(rtx_ & kRtxRedundantPayloads))
450 uint8_t buffer[IP_PACKET_SIZE];
451 int bytes_left = bytes_to_send;
452 while (bytes_left > 0) {
453 uint16_t length = bytes_left;
454 int64_t capture_time_ms;
455 if (!packet_history_.GetBestFittingPacket(buffer, &length,
459 if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true))
461 ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length);
462 RTPHeader rtp_header;
463 rtp_parser.Parse(rtp_header);
464 bytes_left -= length - rtp_header.headerLength;
466 return bytes_to_send - bytes_left;
469 bool RTPSender::SendPaddingAccordingToBitrate(
470 int8_t payload_type, uint32_t capture_timestamp,
471 int64_t capture_time_ms) {
472 // Current bitrate since last estimate(1 second) averaged with the
473 // estimate since then, to get the most up to date bitrate.
474 uint32_t current_bitrate = bitrate_sent_.BitrateNow();
475 int bitrate_diff = target_send_bitrate_ * 1000 - current_bitrate;
476 if (bitrate_diff <= 0) {
480 if (current_bitrate == 0) {
481 // Start up phase. Send one 33.3 ms batch to start with.
482 bytes = (bitrate_diff / 8) / 30;
484 bytes = (bitrate_diff / 8);
485 // Cap at 200 ms of target send data.
486 int bytes_cap = target_send_bitrate_ * 25; // 1000 / 8 / 5.
487 if (bytes > bytes_cap) {
493 CriticalSectionScoped cs(send_critsect_);
494 // Add the random RTP timestamp offset and store the capture time for
495 // later calculation of the send time offset.
496 timestamp = start_time_stamp_ + capture_timestamp;
497 timestamp_ = timestamp;
498 capture_time_ms_ = capture_time_ms;
499 last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
501 int bytes_sent = SendPadData(payload_type, timestamp, capture_time_ms,
502 bytes, kDontRetransmit, false, false);
503 // We did not manage to send all bytes. Comparing with 31 due to modulus 32.
504 return bytes - bytes_sent < 31;
507 int RTPSender::BuildPaddingPacket(uint8_t* packet, int header_length,
509 int padding_bytes_in_packet = kMaxPaddingLength;
510 if (bytes < kMaxPaddingLength) {
511 padding_bytes_in_packet = bytes;
513 packet[0] |= 0x20; // Set padding bit.
515 reinterpret_cast<int32_t *>(&(packet[header_length]));
517 // Fill data buffer with random data.
518 for (int j = 0; j < (padding_bytes_in_packet >> 2); ++j) {
519 data[j] = rand(); // NOLINT
521 // Set number of padding bytes in the last byte of the packet.
522 packet[header_length + padding_bytes_in_packet - 1] = padding_bytes_in_packet;
523 return padding_bytes_in_packet;
526 int RTPSender::SendPadData(int payload_type, uint32_t timestamp,
527 int64_t capture_time_ms, int32_t bytes,
528 StorageType store, bool force_full_size_packets,
529 bool only_pad_after_markerbit) {
530 // Drop this packet if we're not sending media packets.
531 if (!sending_media_) {
534 int padding_bytes_in_packet = 0;
536 for (; bytes > 0; bytes -= padding_bytes_in_packet) {
537 // Always send full padding packets.
538 if (force_full_size_packets && bytes < kMaxPaddingLength)
539 bytes = kMaxPaddingLength;
540 if (bytes < kMaxPaddingLength) {
541 if (force_full_size_packets) {
542 bytes = kMaxPaddingLength;
544 // Round to the nearest multiple of 32.
545 bytes = (bytes + 16) & 0xffe0;
549 // Sanity don't send empty packets.
553 uint16_t sequence_number;
555 CriticalSectionScoped cs(send_critsect_);
556 // Only send padding packets following the last packet of a frame,
557 // indicated by the marker bit.
558 if (only_pad_after_markerbit && !last_packet_marker_bit_)
560 if (rtx_ == kRtxOff) {
562 sequence_number = sequence_number_;
566 sequence_number = sequence_number_rtx_;
567 ++sequence_number_rtx_;
570 uint8_t padding_packet[IP_PACKET_SIZE];
571 int header_length = CreateRTPHeader(padding_packet, payload_type, ssrc,
572 false, timestamp, sequence_number, NULL,
574 padding_bytes_in_packet = BuildPaddingPacket(padding_packet, header_length,
576 if (0 > SendToNetwork(padding_packet, padding_bytes_in_packet,
577 header_length, capture_time_ms, store,
578 PacedSender::kLowPriority)) {
579 // Error sending the packet.
582 bytes_sent += padding_bytes_in_packet;
587 void RTPSender::SetStorePacketsStatus(const bool enable,
588 const uint16_t number_to_store) {
589 packet_history_.SetStorePacketsStatus(enable, number_to_store);
592 bool RTPSender::StorePackets() const {
593 return packet_history_.StorePackets();
596 int32_t RTPSender::ReSendPacket(uint16_t packet_id, uint32_t min_resend_time) {
597 uint16_t length = IP_PACKET_SIZE;
598 uint8_t data_buffer[IP_PACKET_SIZE];
599 uint8_t *buffer_to_send_ptr = data_buffer;
600 int64_t capture_time_ms;
601 if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
602 data_buffer, &length,
608 ModuleRTPUtility::RTPHeaderParser rtp_parser(data_buffer, length);
610 if (!rtp_parser.Parse(header)) {
612 WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id_,
613 "Failed to parse RTP header of packet to be retransmitted.");
616 TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::ReSendPacket",
617 "timestamp", header.timestamp,
618 "seqnum", header.sequenceNumber);
621 if (!paced_sender_->SendPacket(PacedSender::kHighPriority,
623 header.sequenceNumber,
625 length - header.headerLength,
627 // We can't send the packet right now.
628 // We will be called when it is time.
633 uint8_t data_buffer_rtx[IP_PACKET_SIZE];
634 if ((rtx_ & kRtxRetransmitted) > 0) {
635 BuildRtxPacket(data_buffer, &length, data_buffer_rtx);
636 buffer_to_send_ptr = data_buffer_rtx;
639 if (SendPacketToNetwork(buffer_to_send_ptr, length)) {
640 UpdateRtpStats(buffer_to_send_ptr, length, header, rtx_ != kRtxOff, true);
646 bool RTPSender::SendPacketToNetwork(const uint8_t *packet, uint32_t size) {
649 bytes_sent = transport_->SendPacket(id_, packet, size);
651 TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::SendPacketToNetwork",
652 "size", size, "sent", bytes_sent);
653 // TODO(pwesin): Add a separate bitrate for sent bitrate after pacer.
654 if (bytes_sent <= 0) {
655 WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
656 "Transport failed to send packet");
662 int RTPSender::SelectiveRetransmissions() const {
665 return video_->SelectiveRetransmissions();
668 int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
671 return video_->SetSelectiveRetransmissions(settings);
674 void RTPSender::OnReceivedNACK(
675 const std::list<uint16_t>& nack_sequence_numbers,
676 const uint16_t avg_rtt) {
677 TRACE_EVENT2("webrtc_rtp", "RTPSender::OnReceivedNACK",
678 "num_seqnum", nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
679 const int64_t now = clock_->TimeInMilliseconds();
680 uint32_t bytes_re_sent = 0;
682 // Enough bandwidth to send NACK?
683 if (!ProcessNACKBitRate(now)) {
684 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
685 "NACK bitrate reached. Skip sending NACK response. Target %d",
686 target_send_bitrate_);
690 for (std::list<uint16_t>::const_iterator it = nack_sequence_numbers.begin();
691 it != nack_sequence_numbers.end(); ++it) {
692 const int32_t bytes_sent = ReSendPacket(*it, 5 + avg_rtt);
693 if (bytes_sent > 0) {
694 bytes_re_sent += bytes_sent;
695 } else if (bytes_sent == 0) {
696 // The packet has previously been resent.
697 // Try resending next packet in the list.
699 } else if (bytes_sent < 0) {
700 // Failed to send one Sequence number. Give up the rest in this nack.
701 WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, id_,
702 "Failed resending RTP packet %d, Discard rest of packets",
706 // Delay bandwidth estimate (RTT * BW).
707 if (target_send_bitrate_ != 0 && avg_rtt) {
708 // kbits/s * ms = bits => bits/8 = bytes
709 uint32_t target_bytes =
710 (static_cast<uint32_t>(target_send_bitrate_) * avg_rtt) >> 3;
711 if (bytes_re_sent > target_bytes) {
712 break; // Ignore the rest of the packets in the list.
716 if (bytes_re_sent > 0) {
717 // TODO(pwestin) consolidate these two methods.
718 UpdateNACKBitRate(bytes_re_sent, now);
719 nack_bitrate_.Update(bytes_re_sent);
723 bool RTPSender::ProcessNACKBitRate(const uint32_t now) {
725 int32_t byte_count = 0;
726 const uint32_t avg_interval = 1000;
728 CriticalSectionScoped cs(send_critsect_);
730 if (target_send_bitrate_ == 0) {
733 for (num = 0; num < NACK_BYTECOUNT_SIZE; ++num) {
734 if ((now - nack_byte_count_times_[num]) > avg_interval) {
735 // Don't use data older than 1sec.
738 byte_count += nack_byte_count_[num];
741 int32_t time_interval = avg_interval;
742 if (num == NACK_BYTECOUNT_SIZE) {
743 // More than NACK_BYTECOUNT_SIZE nack messages has been received
744 // during the last msg_interval.
745 time_interval = now - nack_byte_count_times_[num - 1];
746 if (time_interval < 0) {
747 time_interval = avg_interval;
750 return (byte_count * 8) < (target_send_bitrate_ * time_interval);
753 void RTPSender::UpdateNACKBitRate(const uint32_t bytes,
754 const uint32_t now) {
755 CriticalSectionScoped cs(send_critsect_);
757 // Save bitrate statistics.
760 // Add padding length.
761 nack_byte_count_[0] += bytes;
763 if (nack_byte_count_times_[0] == 0) {
767 for (int i = (NACK_BYTECOUNT_SIZE - 2); i >= 0; i--) {
768 nack_byte_count_[i + 1] = nack_byte_count_[i];
769 nack_byte_count_times_[i + 1] = nack_byte_count_times_[i];
772 nack_byte_count_[0] = bytes;
773 nack_byte_count_times_[0] = now;
778 // Called from pacer when we can send the packet.
779 bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
780 int64_t capture_time_ms,
781 bool retransmission) {
782 uint16_t length = IP_PACKET_SIZE;
783 uint8_t data_buffer[IP_PACKET_SIZE];
784 int64_t stored_time_ms;
786 if (!packet_history_.GetPacketAndSetSendTime(sequence_number,
792 // Packet cannot be found. Allow sending to continue.
795 if (!retransmission && capture_time_ms > 0) {
796 UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds());
798 return PrepareAndSendPacket(data_buffer, length, capture_time_ms,
799 retransmission && (rtx_ & kRtxRetransmitted) > 0);
802 bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
804 int64_t capture_time_ms,
805 bool send_over_rtx) {
806 uint8_t *buffer_to_send_ptr = buffer;
808 ModuleRTPUtility::RTPHeaderParser rtp_parser(buffer, length);
809 RTPHeader rtp_header;
810 rtp_parser.Parse(rtp_header);
811 TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::TimeToSendPacket",
812 "timestamp", rtp_header.timestamp,
813 "seqnum", rtp_header.sequenceNumber);
815 uint8_t data_buffer_rtx[IP_PACKET_SIZE];
817 BuildRtxPacket(buffer, &length, data_buffer_rtx);
818 buffer_to_send_ptr = data_buffer_rtx;
821 int64_t now_ms = clock_->TimeInMilliseconds();
822 int64_t diff_ms = now_ms - capture_time_ms;
823 bool updated_transmission_time_offset =
824 UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
826 bool updated_abs_send_time =
827 UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
828 if (updated_transmission_time_offset || updated_abs_send_time) {
829 // Update stored packet in case of receiving a re-transmission request.
830 packet_history_.ReplaceRTPHeader(buffer_to_send_ptr,
831 rtp_header.sequenceNumber,
832 rtp_header.headerLength);
835 bool ret = SendPacketToNetwork(buffer_to_send_ptr, length);
836 UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, false, false);
840 void RTPSender::UpdateRtpStats(const uint8_t* buffer,
842 const RTPHeader& header,
844 bool is_retransmit) {
845 CriticalSectionScoped lock(statistics_crit_.get());
846 StreamDataCounters* counters;
849 counters = &rtx_rtp_stats_;
852 counters = &rtp_stats_;
856 bitrate_sent_.Update(size);
858 if (IsFecPacket(buffer, header)) {
859 ++counters->fec_packets;
863 ++counters->retransmitted_packets;
865 counters->bytes += size - (header.headerLength + header.paddingLength);
866 counters->header_bytes += header.headerLength;
867 counters->padding_bytes += header.paddingLength;
870 if (rtp_stats_callback_) {
871 rtp_stats_callback_->DataCountersUpdated(*counters, ssrc);
875 bool RTPSender::IsFecPacket(const uint8_t* buffer,
876 const RTPHeader& header) const {
883 video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
884 return fec_enabled &&
885 header.payloadType == pt_red &&
886 buffer[header.headerLength] == pt_fec;
889 int RTPSender::TimeToSendPadding(int bytes) {
890 if (!sending_media_) {
894 int64_t capture_time_ms;
897 CriticalSectionScoped cs(send_critsect_);
898 payload_type = ((rtx_ & kRtxRedundantPayloads) > 0) ? payload_type_rtx_ :
900 timestamp = timestamp_;
901 capture_time_ms = capture_time_ms_;
902 if (last_timestamp_time_ms_ > 0) {
904 (clock_->TimeInMilliseconds() - last_timestamp_time_ms_) * 90;
906 (clock_->TimeInMilliseconds() - last_timestamp_time_ms_);
909 int bytes_sent = SendRedundantPayloads(payload_type, bytes);
912 int padding_sent = SendPadData(payload_type, timestamp, capture_time_ms,
913 bytes, kDontStore, true, true);
914 bytes_sent += padding_sent;
919 // TODO(pwestin): send in the RTPHeaderParser to avoid parsing it again.
920 int32_t RTPSender::SendToNetwork(
921 uint8_t *buffer, int payload_length, int rtp_header_length,
922 int64_t capture_time_ms, StorageType storage,
923 PacedSender::Priority priority) {
924 ModuleRTPUtility::RTPHeaderParser rtp_parser(
925 buffer, payload_length + rtp_header_length);
926 RTPHeader rtp_header;
927 rtp_parser.Parse(rtp_header);
929 int64_t now_ms = clock_->TimeInMilliseconds();
931 // |capture_time_ms| <= 0 is considered invalid.
932 // TODO(holmer): This should be changed all over Video Engine so that negative
933 // time is consider invalid, while 0 is considered a valid time.
934 if (capture_time_ms > 0) {
935 UpdateTransmissionTimeOffset(buffer, payload_length + rtp_header_length,
936 rtp_header, now_ms - capture_time_ms);
939 UpdateAbsoluteSendTime(buffer, payload_length + rtp_header_length,
942 // Used for NACK and to spread out the transmission of packets.
943 if (packet_history_.PutRTPPacket(buffer, rtp_header_length + payload_length,
944 max_payload_length_, capture_time_ms,
949 if (paced_sender_ && storage != kDontStore) {
950 if (!paced_sender_->SendPacket(priority, rtp_header.ssrc,
951 rtp_header.sequenceNumber, capture_time_ms,
952 payload_length, false)) {
953 // We can't send the packet right now.
954 // We will be called when it is time.
958 if (capture_time_ms > 0) {
959 UpdateDelayStatistics(capture_time_ms, now_ms);
961 uint32_t length = payload_length + rtp_header_length;
962 if (!SendPacketToNetwork(buffer, length))
964 UpdateRtpStats(buffer, length, rtp_header, false, false);
968 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
969 CriticalSectionScoped cs(statistics_crit_.get());
970 send_delays_[now_ms] = now_ms - capture_time_ms;
971 send_delays_.erase(send_delays_.begin(),
972 send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs));
975 void RTPSender::ProcessBitrate() {
976 CriticalSectionScoped cs(send_critsect_);
977 bitrate_sent_.Process();
978 nack_bitrate_.Process();
979 if (audio_configured_) {
982 video_->ProcessBitrate();
985 uint16_t RTPSender::RTPHeaderLength() const {
986 uint16_t rtp_header_length = 12;
987 if (include_csrcs_) {
988 rtp_header_length += sizeof(uint32_t) * num_csrcs_;
990 rtp_header_length += RtpHeaderExtensionTotalLength();
991 return rtp_header_length;
994 uint16_t RTPSender::IncrementSequenceNumber() {
995 CriticalSectionScoped cs(send_critsect_);
996 return sequence_number_++;
999 void RTPSender::ResetDataCounters() {
1000 CriticalSectionScoped lock(statistics_crit_.get());
1001 rtp_stats_ = StreamDataCounters();
1002 rtx_rtp_stats_ = StreamDataCounters();
1003 if (rtp_stats_callback_) {
1004 rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc_);
1005 rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx_);
1009 uint32_t RTPSender::Packets() const {
1010 CriticalSectionScoped lock(statistics_crit_.get());
1011 return rtp_stats_.packets + rtx_rtp_stats_.packets;
1014 // Number of sent RTP bytes.
1015 uint32_t RTPSender::Bytes() const {
1016 CriticalSectionScoped lock(statistics_crit_.get());
1017 return rtp_stats_.bytes + rtx_rtp_stats_.bytes;
1020 int RTPSender::CreateRTPHeader(
1021 uint8_t* header, int8_t payload_type, uint32_t ssrc, bool marker_bit,
1022 uint32_t timestamp, uint16_t sequence_number, const uint32_t* csrcs,
1023 uint8_t num_csrcs) const {
1024 header[0] = 0x80; // version 2.
1025 header[1] = static_cast<uint8_t>(payload_type);
1027 header[1] |= kRtpMarkerBitMask; // Marker bit is set.
1029 ModuleRTPUtility::AssignUWord16ToBuffer(header + 2, sequence_number);
1030 ModuleRTPUtility::AssignUWord32ToBuffer(header + 4, timestamp);
1031 ModuleRTPUtility::AssignUWord32ToBuffer(header + 8, ssrc);
1032 int32_t rtp_header_length = 12;
1034 // Add the CSRCs if any.
1035 if (num_csrcs > 0) {
1036 if (num_csrcs > kRtpCsrcSize) {
1041 uint8_t *ptr = &header[rtp_header_length];
1042 for (int i = 0; i < num_csrcs; ++i) {
1043 ModuleRTPUtility::AssignUWord32ToBuffer(ptr, csrcs[i]);
1046 header[0] = (header[0] & 0xf0) | num_csrcs;
1048 // Update length of header.
1049 rtp_header_length += sizeof(uint32_t) * num_csrcs;
1052 uint16_t len = BuildRTPHeaderExtension(header + rtp_header_length);
1054 header[0] |= 0x10; // Set extension bit.
1055 rtp_header_length += len;
1057 return rtp_header_length;
1060 int32_t RTPSender::BuildRTPheader(
1061 uint8_t *data_buffer, const int8_t payload_type,
1062 const bool marker_bit, const uint32_t capture_timestamp,
1063 int64_t capture_time_ms, const bool time_stamp_provided,
1064 const bool inc_sequence_number) {
1065 assert(payload_type >= 0);
1066 CriticalSectionScoped cs(send_critsect_);
1068 if (time_stamp_provided) {
1069 timestamp_ = start_time_stamp_ + capture_timestamp;
1071 // Make a unique time stamp.
1072 // We can't inc by the actual time, since then we increase the risk of back
1076 last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
1077 uint32_t sequence_number = sequence_number_++;
1078 capture_time_ms_ = capture_time_ms;
1079 last_packet_marker_bit_ = marker_bit;
1080 int csrcs_length = 0;
1082 csrcs_length = num_csrcs_;
1083 return CreateRTPHeader(data_buffer, payload_type, ssrc_, marker_bit,
1084 timestamp_, sequence_number, csrcs_, csrcs_length);
1087 uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) const {
1088 if (rtp_header_extension_map_.Size() <= 0) {
1091 // RTP header extension, RFC 3550.
1093 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
1094 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
1095 // | defined by profile | length |
1096 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
1097 // | header extension |
1100 const uint32_t kPosLength = 2;
1101 const uint32_t kHeaderLength = kRtpOneByteHeaderLength;
1103 // Add extension ID (0xBEDE).
1104 ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer,
1105 kRtpOneByteHeaderExtensionId);
1108 uint16_t total_block_length = 0;
1110 RTPExtensionType type = rtp_header_extension_map_.First();
1111 while (type != kRtpExtensionNone) {
1112 uint8_t block_length = 0;
1114 case kRtpExtensionTransmissionTimeOffset:
1115 block_length = BuildTransmissionTimeOffsetExtension(
1116 data_buffer + kHeaderLength + total_block_length);
1118 case kRtpExtensionAudioLevel:
1119 // Because AudioLevel is handled specially by RTPSenderAudio, we pretend
1120 // we don't have to care about it here, which is true until we wan't to
1121 // use it together with any of the other extensions we support.
1123 case kRtpExtensionAbsoluteSendTime:
1124 block_length = BuildAbsoluteSendTimeExtension(
1125 data_buffer + kHeaderLength + total_block_length);
1130 total_block_length += block_length;
1131 type = rtp_header_extension_map_.Next(type);
1133 if (total_block_length == 0) {
1134 // No extension added.
1137 // Set header length (in number of Word32, header excluded).
1138 assert(total_block_length % 4 == 0);
1139 ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + kPosLength,
1140 total_block_length / 4);
1141 // Total added length.
1142 return kHeaderLength + total_block_length;
1145 uint8_t RTPSender::BuildTransmissionTimeOffsetExtension(
1146 uint8_t* data_buffer) const {
1147 // From RFC 5450: Transmission Time Offsets in RTP Streams.
1149 // The transmission time is signaled to the receiver in-band using the
1150 // general mechanism for RTP header extensions [RFC5285]. The payload
1151 // of this extension (the transmitted value) is a 24-bit signed integer.
1152 // When added to the RTP timestamp of the packet, it represents the
1153 // "effective" RTP transmission time of the packet, on the RTP
1156 // The form of the transmission offset extension block:
1159 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
1160 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
1161 // | ID | len=2 | transmission offset |
1162 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
1164 // Get id defined by user.
1166 if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
1172 const uint8_t len = 2;
1173 data_buffer[pos++] = (id << 4) + len;
1174 ModuleRTPUtility::AssignUWord24ToBuffer(data_buffer + pos,
1175 transmission_time_offset_);
1177 assert(pos == kTransmissionTimeOffsetLength);
1178 return kTransmissionTimeOffsetLength;
1181 uint8_t RTPSender::BuildAbsoluteSendTimeExtension(
1182 uint8_t* data_buffer) const {
1183 // Absolute send time in RTP streams.
1185 // The absolute send time is signaled to the receiver in-band using the
1186 // general mechanism for RTP header extensions [RFC5285]. The payload
1187 // of this extension (the transmitted value) is a 24-bit unsigned integer
1188 // containing the sender's current time in seconds as a fixed point number
1189 // with 18 bits fractional part.
1191 // The form of the absolute send time extension block:
1194 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
1195 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
1196 // | ID | len=2 | absolute send time |
1197 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
1199 // Get id defined by user.
1201 if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
1207 const uint8_t len = 2;
1208 data_buffer[pos++] = (id << 4) + len;
1209 ModuleRTPUtility::AssignUWord24ToBuffer(data_buffer + pos,
1210 absolute_send_time_);
1212 assert(pos == kAbsoluteSendTimeLength);
1213 return kAbsoluteSendTimeLength;
1216 bool RTPSender::UpdateTransmissionTimeOffset(
1217 uint8_t *rtp_packet, const uint16_t rtp_packet_length,
1218 const RTPHeader &rtp_header, const int64_t time_diff_ms) const {
1219 CriticalSectionScoped cs(send_critsect_);
1221 // Get length until start of header extension block.
1222 int extension_block_pos =
1223 rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
1224 kRtpExtensionTransmissionTimeOffset);
1225 if (extension_block_pos < 0) {
1226 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1227 "Failed to update transmission time offset, not registered.");
1230 int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
1231 if (rtp_packet_length < block_pos + kTransmissionTimeOffsetLength ||
1232 rtp_header.headerLength <
1233 block_pos + kTransmissionTimeOffsetLength) {
1234 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1235 "Failed to update transmission time offset, invalid length.");
1238 // Verify that header contains extension.
1239 if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
1240 (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
1242 kTraceStream, kTraceRtpRtcp, id_,
1243 "Failed to update transmission time offset, hdr extension not found.");
1248 if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
1250 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1251 "Failed to update transmission time offset, no id.");
1254 // Verify first byte in block.
1255 const uint8_t first_block_byte = (id << 4) + 2;
1256 if (rtp_packet[block_pos] != first_block_byte) {
1257 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1258 "Failed to update transmission time offset.");
1261 // Update transmission offset field (converting to a 90 kHz timestamp).
1262 ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
1263 time_diff_ms * 90); // RTP timestamp.
1267 bool RTPSender::UpdateAbsoluteSendTime(
1268 uint8_t *rtp_packet, const uint16_t rtp_packet_length,
1269 const RTPHeader &rtp_header, const int64_t now_ms) const {
1270 CriticalSectionScoped cs(send_critsect_);
1272 // Get length until start of header extension block.
1273 int extension_block_pos =
1274 rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
1275 kRtpExtensionAbsoluteSendTime);
1276 if (extension_block_pos < 0) {
1277 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1278 "Failed to update absolute send time, not registered.");
1281 int block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
1282 if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength ||
1283 rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) {
1284 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1285 "Failed to update absolute send time, invalid length.");
1288 // Verify that header contains extension.
1289 if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
1290 (rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
1292 kTraceStream, kTraceRtpRtcp, id_,
1293 "Failed to update absolute send time, hdr extension not found.");
1298 if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
1300 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1301 "Failed to update absolute send time, no id.");
1304 // Verify first byte in block.
1305 const uint8_t first_block_byte = (id << 4) + 2;
1306 if (rtp_packet[block_pos] != first_block_byte) {
1307 WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_,
1308 "Failed to update absolute send time.");
1311 // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
1312 // fractional part).
1313 ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
1314 ((now_ms << 18) / 1000) & 0x00ffffff);
1318 void RTPSender::SetSendingStatus(bool enabled) {
1320 uint32_t frequency_hz = SendPayloadFrequency();
1321 uint32_t RTPtime = ModuleRTPUtility::GetCurrentRTP(clock_, frequency_hz);
1323 // Will be ignored if it's already configured via API.
1324 SetStartTimestamp(RTPtime, false);
1326 if (!ssrc_forced_) {
1327 // Generate a new SSRC.
1328 ssrc_db_.ReturnSSRC(ssrc_);
1329 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
1331 // Don't initialize seq number if SSRC passed externally.
1332 if (!sequence_number_forced_ && !ssrc_forced_) {
1333 // Generate a new sequence number.
1335 rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
1340 void RTPSender::SetSendingMediaStatus(const bool enabled) {
1341 CriticalSectionScoped cs(send_critsect_);
1342 sending_media_ = enabled;
1345 bool RTPSender::SendingMedia() const {
1346 CriticalSectionScoped cs(send_critsect_);
1347 return sending_media_;
1350 uint32_t RTPSender::Timestamp() const {
1351 CriticalSectionScoped cs(send_critsect_);
1355 void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) {
1356 CriticalSectionScoped cs(send_critsect_);
1358 start_time_stamp_forced_ = force;
1359 start_time_stamp_ = timestamp;
1361 if (!start_time_stamp_forced_) {
1362 start_time_stamp_ = timestamp;
1367 uint32_t RTPSender::StartTimestamp() const {
1368 CriticalSectionScoped cs(send_critsect_);
1369 return start_time_stamp_;
1372 uint32_t RTPSender::GenerateNewSSRC() {
1373 // If configured via API, return 0.
1374 CriticalSectionScoped cs(send_critsect_);
1379 ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
1383 void RTPSender::SetSSRC(uint32_t ssrc) {
1384 // This is configured via the API.
1385 CriticalSectionScoped cs(send_critsect_);
1387 if (ssrc_ == ssrc && ssrc_forced_) {
1388 return; // Since it's same ssrc, don't reset anything.
1390 ssrc_forced_ = true;
1391 ssrc_db_.ReturnSSRC(ssrc_);
1392 ssrc_db_.RegisterSSRC(ssrc);
1394 if (!sequence_number_forced_) {
1396 rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
1400 uint32_t RTPSender::SSRC() const {
1401 CriticalSectionScoped cs(send_critsect_);
1405 void RTPSender::SetCSRCStatus(const bool include) {
1406 include_csrcs_ = include;
1409 void RTPSender::SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize],
1410 const uint8_t arr_length) {
1411 assert(arr_length <= kRtpCsrcSize);
1412 CriticalSectionScoped cs(send_critsect_);
1414 for (int i = 0; i < arr_length; i++) {
1415 csrcs_[i] = arr_of_csrc[i];
1417 num_csrcs_ = arr_length;
1420 int32_t RTPSender::CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const {
1421 assert(arr_of_csrc);
1422 CriticalSectionScoped cs(send_critsect_);
1423 for (int i = 0; i < num_csrcs_ && i < kRtpCsrcSize; i++) {
1424 arr_of_csrc[i] = csrcs_[i];
1429 void RTPSender::SetSequenceNumber(uint16_t seq) {
1430 CriticalSectionScoped cs(send_critsect_);
1431 sequence_number_forced_ = true;
1432 sequence_number_ = seq;
1435 uint16_t RTPSender::SequenceNumber() const {
1436 CriticalSectionScoped cs(send_critsect_);
1437 return sequence_number_;
1441 int32_t RTPSender::SendTelephoneEvent(const uint8_t key,
1442 const uint16_t time_ms,
1443 const uint8_t level) {
1444 if (!audio_configured_) {
1447 return audio_->SendTelephoneEvent(key, time_ms, level);
1450 bool RTPSender::SendTelephoneEventActive(int8_t *telephone_event) const {
1451 if (!audio_configured_) {
1454 return audio_->SendTelephoneEventActive(*telephone_event);
1457 int32_t RTPSender::SetAudioPacketSize(
1458 const uint16_t packet_size_samples) {
1459 if (!audio_configured_) {
1462 return audio_->SetAudioPacketSize(packet_size_samples);
1465 int32_t RTPSender::SetAudioLevelIndicationStatus(const bool enable,
1467 if (!audio_configured_) {
1470 return audio_->SetAudioLevelIndicationStatus(enable, ID);
1473 int32_t RTPSender::AudioLevelIndicationStatus(bool *enable,
1474 uint8_t* id) const {
1475 return audio_->AudioLevelIndicationStatus(*enable, *id);
1478 int32_t RTPSender::SetAudioLevel(const uint8_t level_d_bov) {
1479 return audio_->SetAudioLevel(level_d_bov);
1482 int32_t RTPSender::SetRED(const int8_t payload_type) {
1483 if (!audio_configured_) {
1486 return audio_->SetRED(payload_type);
1489 int32_t RTPSender::RED(int8_t *payload_type) const {
1490 if (!audio_configured_) {
1493 return audio_->RED(*payload_type);
1497 VideoCodecInformation *RTPSender::CodecInformationVideo() {
1498 if (audio_configured_) {
1501 return video_->CodecInformationVideo();
1504 RtpVideoCodecTypes RTPSender::VideoCodecType() const {
1505 assert(!audio_configured_ && "Sender is an audio stream!");
1506 return video_->VideoCodecType();
1509 uint32_t RTPSender::MaxConfiguredBitrateVideo() const {
1510 if (audio_configured_) {
1513 return video_->MaxConfiguredBitrateVideo();
1516 int32_t RTPSender::SendRTPIntraRequest() {
1517 if (audio_configured_) {
1520 return video_->SendRTPIntraRequest();
1523 int32_t RTPSender::SetGenericFECStatus(
1524 const bool enable, const uint8_t payload_type_red,
1525 const uint8_t payload_type_fec) {
1526 if (audio_configured_) {
1529 return video_->SetGenericFECStatus(enable, payload_type_red,
1533 int32_t RTPSender::GenericFECStatus(
1534 bool *enable, uint8_t *payload_type_red,
1535 uint8_t *payload_type_fec) const {
1536 if (audio_configured_) {
1539 return video_->GenericFECStatus(
1540 *enable, *payload_type_red, *payload_type_fec);
1543 int32_t RTPSender::SetFecParameters(
1544 const FecProtectionParams *delta_params,
1545 const FecProtectionParams *key_params) {
1546 if (audio_configured_) {
1549 return video_->SetFecParameters(delta_params, key_params);
1552 void RTPSender::BuildRtxPacket(uint8_t* buffer, uint16_t* length,
1553 uint8_t* buffer_rtx) {
1554 CriticalSectionScoped cs(send_critsect_);
1555 uint8_t* data_buffer_rtx = buffer_rtx;
1557 ModuleRTPUtility::RTPHeaderParser rtp_parser(
1558 reinterpret_cast<const uint8_t *>(buffer), *length);
1560 RTPHeader rtp_header;
1561 rtp_parser.Parse(rtp_header);
1563 // Add original RTP header.
1564 memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
1566 // Replace payload type, if a specific type is set for RTX.
1567 if (payload_type_rtx_ != -1) {
1568 data_buffer_rtx[1] = static_cast<uint8_t>(payload_type_rtx_);
1569 if (rtp_header.markerBit)
1570 data_buffer_rtx[1] |= kRtpMarkerBitMask;
1573 // Replace sequence number.
1574 uint8_t *ptr = data_buffer_rtx + 2;
1575 ModuleRTPUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++);
1579 ModuleRTPUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_);
1581 // Add OSN (original sequence number).
1582 ptr = data_buffer_rtx + rtp_header.headerLength;
1583 ModuleRTPUtility::AssignUWord16ToBuffer(ptr, rtp_header.sequenceNumber);
1586 // Add original payload data.
1587 memcpy(ptr, buffer + rtp_header.headerLength,
1588 *length - rtp_header.headerLength);
1592 void RTPSender::RegisterFrameCountObserver(FrameCountObserver* observer) {
1593 CriticalSectionScoped cs(statistics_crit_.get());
1594 if (observer != NULL)
1595 assert(frame_count_observer_ == NULL);
1596 frame_count_observer_ = observer;
1599 FrameCountObserver* RTPSender::GetFrameCountObserver() const {
1600 CriticalSectionScoped cs(statistics_crit_.get());
1601 return frame_count_observer_;
1604 void RTPSender::RegisterRtpStatisticsCallback(
1605 StreamDataCountersCallback* callback) {
1606 CriticalSectionScoped cs(statistics_crit_.get());
1607 if (callback != NULL)
1608 assert(rtp_stats_callback_ == NULL);
1609 rtp_stats_callback_ = callback;
1612 StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
1613 CriticalSectionScoped cs(statistics_crit_.get());
1614 return rtp_stats_callback_;
1617 void RTPSender::RegisterBitrateObserver(BitrateStatisticsObserver* observer) {
1618 CriticalSectionScoped cs(statistics_crit_.get());
1619 if (observer != NULL)
1620 assert(bitrate_callback_ == NULL);
1621 bitrate_callback_ = observer;
1624 BitrateStatisticsObserver* RTPSender::GetBitrateObserver() const {
1625 CriticalSectionScoped cs(statistics_crit_.get());
1626 return bitrate_callback_;
1629 uint32_t RTPSender::BitrateSent() const { return bitrate_sent_.BitrateLast(); }
1631 void RTPSender::BitrateUpdated(const BitrateStatistics& stats) {
1632 CriticalSectionScoped cs(statistics_crit_.get());
1633 if (bitrate_callback_) {
1634 bitrate_callback_->Notify(stats, ssrc_);
1637 } // namespace webrtc