| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <cstdlib> // srand |
| |
| #include "rtp_sender.h" |
| |
| #include "critical_section_wrapper.h" |
| #include "trace.h" |
| |
| #include "rtp_packet_history.h" |
| #include "rtp_sender_audio.h" |
| #include "rtp_sender_video.h" |
| |
| namespace webrtc { |
| RTPSender::RTPSender(const WebRtc_Word32 id, |
| const bool audio, |
| RtpRtcpClock* clock) : |
| Bitrate(clock), |
| _id(id), |
| _audioConfigured(audio), |
| _audio(NULL), |
| _video(NULL), |
| _sendCritsect(CriticalSectionWrapper::CreateCriticalSection()), |
| _transportCritsect(CriticalSectionWrapper::CreateCriticalSection()), |
| |
| _transport(NULL), |
| |
| _sendingMedia(true), // Default to sending media |
| |
| _maxPayloadLength(IP_PACKET_SIZE-28), // default is IP/UDP |
| _targetSendBitrate(0), |
| _packetOverHead(28), |
| |
| _payloadType(-1), |
| _payloadTypeMap(), |
| |
| _rtpHeaderExtensionMap(), |
| _transmissionTimeOffset(0), |
| |
| // NACK |
| _nackByteCountTimes(), |
| _nackByteCount(), |
| _nackBitrate(clock), |
| |
| _packetHistory(new RTPPacketHistory(clock)), |
| _sendBucket(clock), |
| _timeLastSendToNetworkUpdate(clock->GetTimeInMS()), |
| _transmissionSmoothing(false), |
| |
| // statistics |
| _packetsSent(0), |
| _payloadBytesSent(0), |
| |
| // RTP variables |
| _startTimeStampForced(false), |
| _startTimeStamp(0), |
| _ssrcDB(*SSRCDatabase::GetSSRCDatabase()), |
| _remoteSSRC(0), |
| _sequenceNumberForced(false), |
| _sequenceNumber(0), |
| _sequenceNumberRTX(0), |
| _ssrcForced(false), |
| _ssrc(0), |
| _timeStamp(0), |
| _CSRCs(0), |
| _CSRC(), |
| _includeCSRCs(true), |
| _RTX(false), |
| _ssrcRTX(0) |
| { |
| memset(_nackByteCountTimes, 0, sizeof(_nackByteCountTimes)); |
| memset(_nackByteCount, 0, sizeof(_nackByteCount)); |
| |
| memset(_CSRC, 0, sizeof(_CSRC)); |
| |
| // we need to seed the random generator, otherwise we get 26500 each time, hardly a random value :) |
| srand( (WebRtc_UWord32)_clock.GetTimeInMS() ); |
| |
| _ssrc = _ssrcDB.CreateSSRC(); // can't be 0 |
| |
| if(audio) |
| { |
| _audio = new RTPSenderAudio(id, &_clock, this); |
| } else |
| { |
| _video = new RTPSenderVideo(id, &_clock, this); |
| } |
| WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); |
| } |
| |
| RTPSender::~RTPSender() { |
| if(_remoteSSRC != 0) { |
| _ssrcDB.ReturnSSRC(_remoteSSRC); |
| } |
| _ssrcDB.ReturnSSRC(_ssrc); |
| |
| SSRCDatabase::ReturnSSRCDatabase(); |
| delete _sendCritsect; |
| delete _transportCritsect; |
| while (!_payloadTypeMap.empty()) { |
| std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it = |
| _payloadTypeMap.begin(); |
| delete it->second; |
| _payloadTypeMap.erase(it); |
| } |
| delete _packetHistory; |
| delete _audio; |
| delete _video; |
| |
| WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__); |
| } |
| /* |
| WebRtc_Word32 |
| RTPSender::Init(const WebRtc_UWord32 remoteSSRC) |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| // reset to default generation |
| _ssrcForced = false; |
| _startTimeStampForced = false; |
| |
| // register a remote SSRC if we have it to avoid collisions |
| if(remoteSSRC != 0) |
| { |
| if(_ssrc == remoteSSRC) |
| { |
| // collision detected |
| _ssrc = _ssrcDB.CreateSSRC(); // can't be 0 |
| } |
| _remoteSSRC = remoteSSRC; |
| _ssrcDB.RegisterSSRC(remoteSSRC); |
| } |
| _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); |
| _sequenceNumberRTX = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); |
| _packetsSent = 0; |
| _payloadBytesSent = 0; |
| _packetOverHead = 28; |
| |
| _rtpHeaderExtensionMap.Erase(); |
| |
| while (!_payloadTypeMap.empty()) { |
| std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it = |
| _payloadTypeMap.begin(); |
| delete it->second; |
| _payloadTypeMap.erase(it); |
| } |
| |
| memset(_CSRC, 0, sizeof(_CSRC)); |
| |
| memset(_nackByteCount, 0, sizeof(_nackByteCount)); |
| memset(_nackByteCountTimes, 0, sizeof(_nackByteCountTimes)); |
| _nackBitrate.Init(); |
| |
| SetStorePacketsStatus(false, 0); |
| _sendBucket.Reset(); |
| |
| Bitrate::Init(); |
| |
| if(_audioConfigured) |
| { |
| _audio->Init(); |
| } else |
| { |
| _video->Init(); |
| } |
| return(0); |
| } |
| */ |
| |
| void RTPSender::SetTargetSendBitrate(const WebRtc_UWord32 bits) { |
| _targetSendBitrate = static_cast<uint16_t>(bits / 1000); |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::ActualSendBitrateKbit() const |
| { |
| return (WebRtc_UWord16) (Bitrate::BitrateNow()/1000); |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::VideoBitrateSent() const { |
| if (_video) |
| return _video->VideoBitrateSent(); |
| else |
| return 0; |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::FecOverheadRate() const { |
| if (_video) |
| return _video->FecOverheadRate(); |
| else |
| return 0; |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::NackOverheadRate() const { |
| return _nackBitrate.BitrateLast(); |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SetTransmissionTimeOffset( |
| const WebRtc_Word32 transmissionTimeOffset) |
| { |
| if (transmissionTimeOffset > (0x800000 - 1) || |
| transmissionTimeOffset < -(0x800000 - 1)) // Word24 |
| { |
| return -1; |
| } |
| CriticalSectionScoped cs(_sendCritsect); |
| _transmissionTimeOffset = transmissionTimeOffset; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::RegisterRtpHeaderExtension(const RTPExtensionType type, |
| const WebRtc_UWord8 id) |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _rtpHeaderExtensionMap.Register(type, id); |
| } |
| |
| WebRtc_Word32 |
| RTPSender::DeregisterRtpHeaderExtension(const RTPExtensionType type) |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _rtpHeaderExtensionMap.Deregister(type); |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::RtpHeaderExtensionTotalLength() const |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _rtpHeaderExtensionMap.GetTotalLengthInBytes(); |
| } |
| |
| //can be called multiple times |
| WebRtc_Word32 RTPSender::RegisterPayload( |
| const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| const WebRtc_Word8 payloadNumber, |
| const WebRtc_UWord32 frequency, |
| const WebRtc_UWord8 channels, |
| const WebRtc_UWord32 rate) { |
| assert(payloadName); |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it = |
| _payloadTypeMap.find(payloadNumber); |
| |
| if (_payloadTypeMap.end() != it) { |
| // we already use this payload type |
| ModuleRTPUtility::Payload* payload = it->second; |
| assert(payload); |
| |
| // check if it's the same as we already have |
| if (ModuleRTPUtility::StringCompare(payload->name, payloadName, |
| RTP_PAYLOAD_NAME_SIZE - 1)) { |
| if (_audioConfigured && payload->audio && |
| payload->typeSpecific.Audio.frequency == frequency && |
| (payload->typeSpecific.Audio.rate == rate || |
| payload->typeSpecific.Audio.rate == 0 || rate == 0)) { |
| payload->typeSpecific.Audio.rate = rate; |
| // Ensure that we update the rate if new or old is zero |
| return 0; |
| } |
| if(!_audioConfigured && !payload->audio) { |
| return 0; |
| } |
| } |
| return -1; |
| } |
| WebRtc_Word32 retVal = -1; |
| ModuleRTPUtility::Payload* payload = NULL; |
| if (_audioConfigured) { |
| retVal = _audio->RegisterAudioPayload(payloadName, payloadNumber, frequency, |
| channels, rate, payload); |
| } else { |
| retVal = _video->RegisterVideoPayload(payloadName, payloadNumber, rate, |
| payload); |
| } |
| if(payload) { |
| _payloadTypeMap[payloadNumber] = payload; |
| } |
| return retVal; |
| } |
| |
| WebRtc_Word32 RTPSender::DeRegisterSendPayload(const WebRtc_Word8 payloadType) { |
| CriticalSectionScoped lock(_sendCritsect); |
| |
| std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it = |
| _payloadTypeMap.find(payloadType); |
| |
| if (_payloadTypeMap.end() == it) return -1; |
| |
| ModuleRTPUtility::Payload* payload = it->second; |
| delete payload; |
| _payloadTypeMap.erase(it); |
| return 0; |
| } |
| |
| WebRtc_Word8 RTPSender::SendPayloadType() const |
| { |
| return _payloadType; |
| } |
| |
| |
| int RTPSender::SendPayloadFrequency() const |
| { |
| return _audio->AudioFrequency(); |
| } |
| |
| |
| WebRtc_Word32 |
| RTPSender::SetMaxPayloadLength(const WebRtc_UWord16 maxPayloadLength, const WebRtc_UWord16 packetOverHead) |
| { |
| // sanity check |
| if(maxPayloadLength < 100 || maxPayloadLength > IP_PACKET_SIZE) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| |
| CriticalSectionScoped cs(_sendCritsect); |
| _maxPayloadLength = maxPayloadLength; |
| _packetOverHead = packetOverHead; |
| |
| WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, _id, "SetMaxPayloadLength to %d.", maxPayloadLength); |
| return 0; |
| } |
| |
| WebRtc_UWord16 RTPSender::MaxDataPayloadLength() const { |
| if(_audioConfigured) { |
| return _maxPayloadLength - RTPHeaderLength(); |
| } else { |
| return _maxPayloadLength - RTPHeaderLength() - |
| _video->FECPacketOverhead() - ((_RTX) ? 2 : 0); |
| // Include the FEC/ULP/RED overhead. |
| } |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::MaxPayloadLength() const |
| { |
| return _maxPayloadLength; |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::PacketOverHead() const |
| { |
| return _packetOverHead; |
| } |
| |
| void RTPSender::SetTransmissionSmoothingStatus(const bool enable) { |
| CriticalSectionScoped cs(_sendCritsect); |
| _transmissionSmoothing = enable; |
| } |
| |
| bool RTPSender::TransmissionSmoothingStatus() const { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _transmissionSmoothing; |
| } |
| |
| void RTPSender::SetRTXStatus(const bool enable, |
| const bool setSSRC, |
| const WebRtc_UWord32 SSRC) { |
| CriticalSectionScoped cs(_sendCritsect); |
| _RTX = enable; |
| if (enable) { |
| if (setSSRC) { |
| _ssrcRTX = SSRC; |
| } else { |
| _ssrcRTX = _ssrcDB.CreateSSRC(); // can't be 0 |
| } |
| } |
| } |
| |
| void RTPSender::RTXStatus(bool* enable, |
| WebRtc_UWord32* SSRC) const { |
| CriticalSectionScoped cs(_sendCritsect); |
| *enable = _RTX; |
| *SSRC = _ssrcRTX; |
| } |
| |
| WebRtc_Word32 RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType, |
| RtpVideoCodecTypes& videoType) { |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| if (payloadType < 0) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "\tinvalid payloadType (%d)", payloadType); |
| return -1; |
| } |
| if (_audioConfigured) { |
| WebRtc_Word8 redPlType = -1; |
| if (_audio->RED(redPlType) == 0) { |
| // We have configured RED. |
| if(redPlType == payloadType) { |
| // And it's a match... |
| return 0; |
| } |
| } |
| } |
| if (_payloadType == payloadType) { |
| if (!_audioConfigured) { |
| videoType = _video->VideoCodecType(); |
| } |
| return 0; |
| } |
| std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it = |
| _payloadTypeMap.find(payloadType); |
| if (it == _payloadTypeMap.end()) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "\tpayloadType:%d not registered", payloadType); |
| return -1; |
| } |
| _payloadType = payloadType; |
| ModuleRTPUtility::Payload* payload = it->second; |
| assert(payload); |
| if (!payload->audio && !_audioConfigured) { |
| _video->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType); |
| videoType = payload->typeSpecific.Video.videoCodecType; |
| _video->SetMaxConfiguredBitrateVideo( |
| payload->typeSpecific.Video.maxRate); |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SendOutgoingData(const FrameType frame_type, |
| const WebRtc_Word8 payload_type, |
| const WebRtc_UWord32 capture_timestamp, |
| int64_t capture_time_ms, |
| const WebRtc_UWord8* payload_data, |
| const WebRtc_UWord32 payload_size, |
| const RTPFragmentationHeader* fragmentation, |
| VideoCodecInformation* codec_info, |
| const RTPVideoTypeHeader* rtp_type_hdr) |
| { |
| { |
| // Drop this packet if we're not sending media packets. |
| CriticalSectionScoped cs(_sendCritsect); |
| if (!_sendingMedia) |
| { |
| return 0; |
| } |
| } |
| RtpVideoCodecTypes video_type = kRtpNoVideo; |
| if (CheckPayloadType(payload_type, video_type) != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s invalid argument failed to find payloadType:%d", |
| __FUNCTION__, payload_type); |
| return -1; |
| } |
| |
| if (_audioConfigured) |
| { |
| assert(frame_type == kAudioFrameSpeech || |
| frame_type == kAudioFrameCN || |
| frame_type == kFrameEmpty); |
| |
| return _audio->SendAudio(frame_type, payload_type, capture_timestamp, |
| payload_data, payload_size,fragmentation); |
| } else { |
| assert(frame_type != kAudioFrameSpeech && |
| frame_type != kAudioFrameCN); |
| |
| if (frame_type == kFrameEmpty) { |
| return SendPaddingAccordingToBitrate(payload_type, capture_timestamp, |
| capture_time_ms); |
| } |
| return _video->SendVideo(video_type, |
| frame_type, |
| payload_type, |
| capture_timestamp, |
| capture_time_ms, |
| payload_data, |
| payload_size, |
| fragmentation, |
| codec_info, |
| rtp_type_hdr); |
| } |
| } |
| |
| WebRtc_Word32 RTPSender::SendPaddingAccordingToBitrate( |
| WebRtc_Word8 payload_type, |
| WebRtc_UWord32 capture_timestamp, |
| int64_t capture_time_ms) { |
| // Current bitrate since last estimate(1 second) averaged with the |
| // estimate since then, to get the most up to date bitrate. |
| uint32_t current_bitrate = BitrateNow(); |
| int bitrate_diff = _targetSendBitrate * 1000 - current_bitrate; |
| if (bitrate_diff > 0) { |
| int bytes = 0; |
| if (current_bitrate == 0) { |
| // Start up phase. Send one 33.3 ms batch to start with. |
| bytes = (bitrate_diff / 8) / 30; |
| } else { |
| bytes = (bitrate_diff / 8); |
| // Cap at 200 ms of target send data. |
| int bytes_cap = _targetSendBitrate * 25; // 1000 / 8 / 5 |
| if (bytes > bytes_cap) { |
| bytes = bytes_cap; |
| } |
| } |
| // Send padding data. |
| return SendPadData(payload_type, capture_timestamp, capture_time_ms, bytes); |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 RTPSender::SendPadData(WebRtc_Word8 payload_type, |
| WebRtc_UWord32 capture_timestamp, |
| int64_t capture_time_ms, |
| WebRtc_Word32 bytes) { |
| // Drop this packet if we're not sending media packets |
| if (!_sendingMedia) { |
| return 0; |
| } |
| // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP. |
| int max_length = 224; |
| WebRtc_UWord8 data_buffer[IP_PACKET_SIZE]; |
| |
| for (; bytes > 0; bytes -= max_length) { |
| int padding_bytes_in_packet = max_length; |
| if (bytes < max_length) { |
| padding_bytes_in_packet = (bytes + 16) & 0xffe0; // Keep our modulus 32. |
| } |
| if (padding_bytes_in_packet < 32) { |
| // Sanity don't send empty packets. |
| break; |
| } |
| |
| WebRtc_Word32 header_length; |
| { |
| // Correct seq num, timestamp and payload type. |
| header_length = BuildRTPheader(data_buffer, |
| payload_type, |
| false, // No markerbit. |
| capture_timestamp, |
| true, // Timestamp provided. |
| true); // Increment sequence number. |
| } |
| data_buffer[0] |= 0x20; // Set padding bit. |
| WebRtc_Word32* data = |
| reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length])); |
| |
| // Fill data buffer with random data. |
| for(int j = 0; j < (padding_bytes_in_packet >> 2); j++) { |
| data[j] = rand(); |
| } |
| // Set number of padding bytes in the last byte of the packet. |
| data_buffer[header_length + padding_bytes_in_packet - 1] = |
| padding_bytes_in_packet; |
| // Send the packet |
| if (0 > SendToNetwork(data_buffer, |
| padding_bytes_in_packet, |
| header_length, |
| capture_time_ms, |
| kDontRetransmit)) { |
| // Error sending the packet. |
| break; |
| } |
| } |
| if (bytes > 31) { // 31 due to our modulus 32. |
| // We did not manage to send all bytes. |
| return -1; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 RTPSender::SetStorePacketsStatus( |
| const bool enable, |
| const WebRtc_UWord16 numberToStore) { |
| _packetHistory->SetStorePacketsStatus(enable, numberToStore); |
| return 0; |
| } |
| |
| bool RTPSender::StorePackets() const { |
| return _packetHistory->StorePackets(); |
| } |
| |
| WebRtc_Word32 RTPSender::ReSendPacket(WebRtc_UWord16 packet_id, |
| WebRtc_UWord32 min_resend_time) { |
| |
| WebRtc_UWord16 length = IP_PACKET_SIZE; |
| WebRtc_UWord8 data_buffer[IP_PACKET_SIZE]; |
| WebRtc_UWord8* buffer_to_send_ptr = data_buffer; |
| |
| int64_t stored_time_in_ms; |
| StorageType type; |
| bool found = _packetHistory->GetRTPPacket(packet_id, |
| min_resend_time, data_buffer, &length, &stored_time_in_ms, &type); |
| if (!found) { |
| // Packet not found. |
| return 0; |
| } |
| |
| if (length == 0 || type == kDontRetransmit) { |
| // No bytes copied (packet recently resent, skip resending) or |
| // packet should not be retransmitted. |
| return 0; |
| } |
| |
| WebRtc_UWord8 data_buffer_rtx[IP_PACKET_SIZE]; |
| if (_RTX) { |
| buffer_to_send_ptr = data_buffer_rtx; |
| |
| CriticalSectionScoped cs(_sendCritsect); |
| // Add RTX header. |
| ModuleRTPUtility::RTPHeaderParser rtpParser( |
| reinterpret_cast<const WebRtc_UWord8*>(data_buffer), |
| length); |
| |
| WebRtcRTPHeader rtp_header; |
| rtpParser.Parse(rtp_header); |
| |
| // Add original RTP header. |
| memcpy(data_buffer_rtx, data_buffer, rtp_header.header.headerLength); |
| |
| // Replace sequence number. |
| WebRtc_UWord8* ptr = data_buffer_rtx + 2; |
| ModuleRTPUtility::AssignUWord16ToBuffer(ptr, _sequenceNumberRTX++); |
| |
| // Replace SSRC. |
| ptr += 6; |
| ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _ssrcRTX); |
| |
| // Add OSN (original sequence number). |
| ptr = data_buffer_rtx + rtp_header.header.headerLength; |
| ModuleRTPUtility::AssignUWord16ToBuffer( |
| ptr, rtp_header.header.sequenceNumber); |
| ptr += 2; |
| |
| // Add original payload data. |
| memcpy(ptr, |
| data_buffer + rtp_header.header.headerLength, |
| length - rtp_header.header.headerLength); |
| length += 2; |
| } |
| |
| WebRtc_Word32 bytes_sent = ReSendToNetwork(buffer_to_send_ptr, length); |
| if (bytes_sent <= 0) { |
| WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, |
| "Transport failed to resend packet_id %u", packet_id); |
| return -1; |
| } |
| |
| // Store the time when the packet was last resent. |
| _packetHistory->UpdateResendTime(packet_id); |
| |
| return bytes_sent; |
| } |
| |
| WebRtc_Word32 RTPSender::ReSendToNetwork(const WebRtc_UWord8* packet, |
| const WebRtc_UWord32 size) { |
| WebRtc_Word32 bytes_sent = -1; |
| { |
| CriticalSectionScoped lock(_transportCritsect); |
| if (_transport) { |
| bytes_sent = _transport->SendPacket(_id, packet, size); |
| } |
| } |
| |
| if (bytes_sent <= 0) { |
| return -1; |
| } |
| |
| // Update send statistics |
| CriticalSectionScoped cs(_sendCritsect); |
| Bitrate::Update(bytes_sent); |
| _packetsSent++; |
| // We on purpose don't add to _payloadBytesSent since this is a |
| // re-transmit and not new payload data. |
| return bytes_sent; |
| } |
| |
| int RTPSender::SelectiveRetransmissions() const { |
| if (!_video) return -1; |
| return _video->SelectiveRetransmissions(); |
| } |
| |
| int RTPSender::SetSelectiveRetransmissions(uint8_t settings) { |
| if (!_video) return -1; |
| return _video->SetSelectiveRetransmissions(settings); |
| } |
| |
| void |
| RTPSender::OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength, |
| const WebRtc_UWord16* nackSequenceNumbers, |
| const WebRtc_UWord16 avgRTT) { |
| const WebRtc_Word64 now = _clock.GetTimeInMS(); |
| WebRtc_UWord32 bytesReSent = 0; |
| |
| // Enough bandwidth to send NACK? |
| if (!ProcessNACKBitRate(now)) { |
| WEBRTC_TRACE(kTraceStream, |
| kTraceRtpRtcp, |
| _id, |
| "NACK bitrate reached. Skip sending NACK response. Target %d", |
| _targetSendBitrate); |
| return; |
| } |
| |
| for (WebRtc_UWord16 i = 0; i < nackSequenceNumbersLength; ++i) { |
| const WebRtc_Word32 bytesSent = ReSendPacket(nackSequenceNumbers[i], |
| 5+avgRTT); |
| if (bytesSent > 0) { |
| bytesReSent += bytesSent; |
| } else if (bytesSent == 0) { |
| // The packet has previously been resent. |
| // Try resending next packet in the list. |
| continue; |
| } else if (bytesSent < 0) { |
| // Failed to send one Sequence number. Give up the rest in this nack. |
| WEBRTC_TRACE(kTraceWarning, |
| kTraceRtpRtcp, |
| _id, |
| "Failed resending RTP packet %d, Discard rest of packets", |
| nackSequenceNumbers[i]); |
| break; |
| } |
| // delay bandwidth estimate (RTT * BW) |
| if (_targetSendBitrate != 0 && avgRTT) { |
| // kbits/s * ms = bits => bits/8 = bytes |
| WebRtc_UWord32 targetBytes = |
| (static_cast<WebRtc_UWord32>(_targetSendBitrate) * avgRTT) >> 3; |
| if (bytesReSent > targetBytes) { |
| break; // ignore the rest of the packets in the list |
| } |
| } |
| } |
| if (bytesReSent > 0) { |
| // TODO(pwestin) consolidate these two methods. |
| UpdateNACKBitRate(bytesReSent, now); |
| _nackBitrate.Update(bytesReSent); |
| } |
| } |
| |
| /** |
| * @return true if the nack bitrate is lower than the requested max bitrate |
| */ |
| bool RTPSender::ProcessNACKBitRate(const WebRtc_UWord32 now) { |
| WebRtc_UWord32 num = 0; |
| WebRtc_Word32 byteCount = 0; |
| const WebRtc_UWord32 avgInterval=1000; |
| |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| if (_targetSendBitrate == 0) { |
| return true; |
| } |
| for (num = 0; num < NACK_BYTECOUNT_SIZE; num++) { |
| if ((now - _nackByteCountTimes[num]) > avgInterval) { |
| // don't use data older than 1sec |
| break; |
| } else { |
| byteCount += _nackByteCount[num]; |
| } |
| } |
| WebRtc_Word32 timeInterval = avgInterval; |
| if (num == NACK_BYTECOUNT_SIZE) { |
| // More than NACK_BYTECOUNT_SIZE nack messages has been received |
| // during the last msgInterval |
| timeInterval = now - _nackByteCountTimes[num-1]; |
| if(timeInterval < 0) { |
| timeInterval = avgInterval; |
| } |
| } |
| return (byteCount*8) < (_targetSendBitrate * timeInterval); |
| } |
| |
| void RTPSender::UpdateNACKBitRate(const WebRtc_UWord32 bytes, |
| const WebRtc_UWord32 now) { |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| // save bitrate statistics |
| if(bytes > 0) { |
| if(now == 0) { |
| // add padding length |
| _nackByteCount[0] += bytes; |
| } else { |
| if(_nackByteCountTimes[0] == 0) { |
| // first no shift |
| } else { |
| // shift |
| for(int i = (NACK_BYTECOUNT_SIZE-2); i >= 0 ; i--) { |
| _nackByteCount[i+1] = _nackByteCount[i]; |
| _nackByteCountTimes[i+1] = _nackByteCountTimes[i]; |
| } |
| } |
| _nackByteCount[0] = bytes; |
| _nackByteCountTimes[0] = now; |
| } |
| } |
| } |
| |
| // Function triggered by timer. |
| void RTPSender::ProcessSendToNetwork() { |
| WebRtc_Word64 delta_time_ms; |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| if (!_transmissionSmoothing) { |
| return; |
| } |
| WebRtc_Word64 now = _clock.GetTimeInMS(); |
| delta_time_ms = now - _timeLastSendToNetworkUpdate; |
| _timeLastSendToNetworkUpdate = now; |
| } |
| _sendBucket.UpdateBytesPerInterval(delta_time_ms, _targetSendBitrate); |
| |
| while (!_sendBucket.Empty()) { |
| |
| WebRtc_Word32 seq_num = _sendBucket.GetNextPacket(); |
| if (seq_num < 0) { |
| break; |
| } |
| |
| WebRtc_UWord8 data_buffer[IP_PACKET_SIZE]; |
| WebRtc_UWord16 length = IP_PACKET_SIZE; |
| int64_t stored_time_ms; |
| StorageType type; |
| bool found = _packetHistory->GetRTPPacket(seq_num, 0, data_buffer, &length, |
| &stored_time_ms, &type); |
| if (!found) { |
| assert(false); |
| return; |
| } |
| assert(length > 0); |
| |
| WebRtc_Word64 diff_ms = _clock.GetTimeInMS() - stored_time_ms; |
| |
| ModuleRTPUtility::RTPHeaderParser rtpParser(data_buffer, length); |
| WebRtcRTPHeader rtp_header; |
| rtpParser.Parse(rtp_header); |
| |
| UpdateTransmissionTimeOffset(data_buffer, length, rtp_header, diff_ms); |
| |
| // Send packet |
| WebRtc_Word32 bytes_sent = -1; |
| { |
| CriticalSectionScoped cs(_transportCritsect); |
| if (_transport) { |
| bytes_sent = _transport->SendPacket(_id, data_buffer, length); |
| } |
| } |
| |
| // Update send statistics |
| if (bytes_sent > 0) { |
| CriticalSectionScoped cs(_sendCritsect); |
| Bitrate::Update(bytes_sent); |
| _packetsSent++; |
| if (bytes_sent > rtp_header.header.headerLength) { |
| _payloadBytesSent += bytes_sent - rtp_header.header.headerLength; |
| } |
| } |
| } |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SendToNetwork(WebRtc_UWord8* buffer, |
| const WebRtc_UWord16 length, |
| const WebRtc_UWord16 rtpLength, |
| int64_t capture_time_ms, |
| const StorageType storage) |
| { |
| // Used for NACK or to spead out the transmission of packets. |
| if (_packetHistory->PutRTPPacket( |
| buffer, rtpLength + length, _maxPayloadLength, capture_time_ms, storage) |
| != 0) { |
| return -1; |
| } |
| |
| if (_transmissionSmoothing) { |
| const WebRtc_UWord16 sequenceNumber = (buffer[2] << 8) + buffer[3]; |
| const WebRtc_UWord32 timestamp = (buffer[4] << 24) + (buffer[5] << 16) + |
| (buffer[6] << 8) + buffer[7]; |
| _sendBucket.Fill(sequenceNumber, timestamp, rtpLength + length); |
| // Packet will be sent at a later time. |
| return 0; |
| } |
| |
| // |capture_time_ms| <= 0 is considered invalid. |
| // TODO(holmer): This should be changed all over Video Engine so that negative |
| // time is consider invalid, while 0 is considered a valid time. |
| if (capture_time_ms > 0) { |
| ModuleRTPUtility::RTPHeaderParser rtpParser(buffer, length); |
| WebRtcRTPHeader rtp_header; |
| rtpParser.Parse(rtp_header); |
| int64_t time_now = _clock.GetTimeInMS(); |
| UpdateTransmissionTimeOffset(buffer, length, rtp_header, |
| time_now - capture_time_ms); |
| } |
| |
| // Send packet |
| WebRtc_Word32 bytes_sent = -1; |
| { |
| CriticalSectionScoped cs(_transportCritsect); |
| if (_transport) { |
| bytes_sent = _transport->SendPacket(_id, buffer, length + rtpLength); |
| } |
| } |
| |
| if (bytes_sent <= 0) { |
| return -1; |
| } |
| |
| // Update send statistics |
| CriticalSectionScoped cs(_sendCritsect); |
| Bitrate::Update(bytes_sent); |
| _packetsSent++; |
| if (bytes_sent > rtpLength) { |
| _payloadBytesSent += bytes_sent - rtpLength; |
| } |
| return 0; |
| } |
| |
| void |
| RTPSender::ProcessBitrate() |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| Bitrate::Process(); |
| _nackBitrate.Process(); |
| |
| if (_audioConfigured) |
| return; |
| _video->ProcessBitrate(); |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::RTPHeaderLength() const |
| { |
| WebRtc_UWord16 rtpHeaderLength = 12; |
| |
| if(_includeCSRCs) |
| { |
| rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs; |
| } |
| rtpHeaderLength += RtpHeaderExtensionTotalLength(); |
| |
| return rtpHeaderLength; |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::IncrementSequenceNumber() |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _sequenceNumber++; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::ResetDataCounters() |
| { |
| _packetsSent = 0; |
| _payloadBytesSent = 0; |
| |
| return 0; |
| } |
| |
| // number of sent RTP packets |
| // dont use critsect to avoid potental deadlock |
| WebRtc_UWord32 |
| RTPSender::Packets() const |
| { |
| return _packetsSent; |
| } |
| |
| // number of sent RTP bytes |
| // dont use critsect to avoid potental deadlock |
| WebRtc_UWord32 |
| RTPSender::Bytes() const |
| { |
| return _payloadBytesSent; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::BuildRTPheader(WebRtc_UWord8* dataBuffer, |
| const WebRtc_Word8 payloadType, |
| const bool markerBit, |
| const WebRtc_UWord32 captureTimeStamp, |
| const bool timeStampProvided, |
| const bool incSequenceNumber) |
| { |
| assert(payloadType>=0); |
| |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2 |
| dataBuffer[1] = static_cast<WebRtc_UWord8>(payloadType); |
| if (markerBit) |
| { |
| dataBuffer[1] |= kRtpMarkerBitMask; // MarkerBit is set |
| } |
| |
| if(timeStampProvided) |
| { |
| _timeStamp = _startTimeStamp + captureTimeStamp; |
| } else |
| { |
| // make a unique time stamp |
| // used for inband signaling |
| // we can't inc by the actual time, since then we increase the risk of back timing |
| _timeStamp++; |
| } |
| |
| ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _sequenceNumber); |
| ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, _timeStamp); |
| ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, _ssrc); |
| |
| WebRtc_Word32 rtpHeaderLength = 12; |
| |
| // Add the CSRCs if any |
| if (_includeCSRCs && _CSRCs > 0) |
| { |
| if(_CSRCs > kRtpCsrcSize) |
| { |
| // error |
| assert(false); |
| return -1; |
| } |
| WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength]; |
| for (WebRtc_UWord32 i = 0; i < _CSRCs; ++i) |
| { |
| ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _CSRC[i]); |
| ptr +=4; |
| } |
| dataBuffer[0] = (dataBuffer[0]&0xf0) | _CSRCs; |
| |
| // Update length of header |
| rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs; |
| } |
| { |
| _sequenceNumber++; // prepare for next packet |
| } |
| |
| WebRtc_UWord16 len = BuildRTPHeaderExtension(dataBuffer + rtpHeaderLength); |
| if (len) |
| { |
| dataBuffer[0] |= 0x10; // set eXtension bit |
| rtpHeaderLength += len; |
| } |
| |
| return rtpHeaderLength; |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const |
| { |
| if (_rtpHeaderExtensionMap.Size() <= 0) { |
| return 0; |
| } |
| |
| /* RTP header extension, RFC 3550. |
| 0 1 2 3 |
| 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 |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | defined by profile | length | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | header extension | |
| | .... | |
| */ |
| |
| const WebRtc_UWord32 kPosLength = 2; |
| const WebRtc_UWord32 kHeaderLength = RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES; |
| |
| // Add extension ID (0xBEDE). |
| ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer, |
| RTP_ONE_BYTE_HEADER_EXTENSION); |
| |
| // Add extensions. |
| WebRtc_UWord16 total_block_length = 0; |
| |
| RTPExtensionType type = _rtpHeaderExtensionMap.First(); |
| while (type != kRtpExtensionNone) |
| { |
| WebRtc_UWord8 block_length = 0; |
| if (type == kRtpExtensionTransmissionTimeOffset) |
| { |
| block_length = BuildTransmissionTimeOffsetExtension( |
| dataBuffer + kHeaderLength + total_block_length); |
| } |
| total_block_length += block_length; |
| type = _rtpHeaderExtensionMap.Next(type); |
| } |
| |
| if (total_block_length == 0) |
| { |
| // No extension added. |
| return 0; |
| } |
| |
| // Set header length (in number of Word32, header excluded). |
| assert(total_block_length % 4 == 0); |
| ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer + kPosLength, |
| total_block_length / 4); |
| |
| // Total added length. |
| return kHeaderLength + total_block_length; |
| } |
| |
| WebRtc_UWord8 |
| RTPSender::BuildTransmissionTimeOffsetExtension(WebRtc_UWord8* dataBuffer) const |
| { |
| // From RFC 5450: Transmission Time Offsets in RTP Streams. |
| // |
| // The transmission time is signaled to the receiver in-band using the |
| // general mechanism for RTP header extensions [RFC5285]. The payload |
| // of this extension (the transmitted value) is a 24-bit signed integer. |
| // When added to the RTP timestamp of the packet, it represents the |
| // "effective" RTP transmission time of the packet, on the RTP |
| // timescale. |
| // |
| // The form of the transmission offset extension block: |
| // |
| // 0 1 2 3 |
| // 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 |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // | ID | len=2 | transmission offset | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
| // Get id defined by user. |
| WebRtc_UWord8 id; |
| if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset, &id) |
| != 0) { |
| // Not registered. |
| return 0; |
| } |
| |
| int pos = 0; |
| const WebRtc_UWord8 len = 2; |
| dataBuffer[pos++] = (id << 4) + len; |
| ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + pos, |
| _transmissionTimeOffset); |
| pos += 3; |
| assert(pos == TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES); |
| return TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES; |
| } |
| |
| void RTPSender::UpdateTransmissionTimeOffset( |
| WebRtc_UWord8* rtp_packet, |
| const WebRtc_UWord16 rtp_packet_length, |
| const WebRtcRTPHeader& rtp_header, |
| const WebRtc_Word64 time_diff_ms) const { |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| // Get length until start of transmission block. |
| int transmission_block_pos = |
| _rtpHeaderExtensionMap.GetLengthUntilBlockStartInBytes( |
| kRtpExtensionTransmissionTimeOffset); |
| if (transmission_block_pos < 0) { |
| WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, |
| "Failed to update transmission time offset, not registered."); |
| return; |
| } |
| |
| int block_pos = 12 + rtp_header.header.numCSRCs + transmission_block_pos; |
| if (rtp_packet_length < block_pos + 4 || |
| rtp_header.header.headerLength < block_pos + 4) { |
| WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, |
| "Failed to update transmission time offset, invalid length."); |
| return; |
| } |
| |
| // Verify that header contains extension. |
| if (!((rtp_packet[12 + rtp_header.header.numCSRCs] == 0xBE) && |
| (rtp_packet[12 + rtp_header.header.numCSRCs + 1] == 0xDE))) { |
| WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, |
| "Failed to update transmission time offset, hdr extension not found."); |
| return; |
| } |
| |
| // Get id. |
| WebRtc_UWord8 id = 0; |
| if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset, |
| &id) != 0) { |
| WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, |
| "Failed to update transmission time offset, no id."); |
| return; |
| } |
| |
| // Verify first byte in block. |
| const WebRtc_UWord8 first_block_byte = (id << 4) + 2; |
| if (rtp_packet[block_pos] != first_block_byte) { |
| WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, |
| "Failed to update transmission time offset."); |
| return; |
| } |
| |
| // Update transmission offset field. |
| ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1, |
| time_diff_ms * 90); // RTP timestamp. |
| } |
| |
| WebRtc_Word32 |
| RTPSender::RegisterSendTransport(Transport* transport) |
| { |
| CriticalSectionScoped cs(_transportCritsect); |
| _transport = transport; |
| return 0; |
| } |
| |
| void |
| RTPSender::SetSendingStatus(const bool enabled) |
| { |
| if(enabled) |
| { |
| WebRtc_UWord32 freq; |
| if(_audioConfigured) |
| { |
| WebRtc_UWord32 frequency = _audio->AudioFrequency(); |
| |
| // sanity |
| switch(frequency) |
| { |
| case 8000: |
| case 12000: |
| case 16000: |
| case 24000: |
| case 32000: |
| break; |
| default: |
| assert(false); |
| return; |
| } |
| freq = frequency; |
| } else |
| { |
| freq = 90000; // 90 KHz for all video |
| } |
| WebRtc_UWord32 RTPtime = ModuleRTPUtility::GetCurrentRTP(&_clock, freq); |
| |
| SetStartTimestamp(RTPtime); // will be ignored if it's already configured via API |
| |
| } else |
| { |
| if(!_ssrcForced) |
| { |
| // generate a new SSRC |
| _ssrcDB.ReturnSSRC(_ssrc); |
| _ssrc = _ssrcDB.CreateSSRC(); // can't be 0 |
| |
| } |
| if(!_sequenceNumberForced && !_ssrcForced) // don't initialize seq number if SSRC passed externally |
| { |
| // generate a new sequence number |
| _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); |
| } |
| } |
| } |
| |
| void |
| RTPSender::SetSendingMediaStatus(const bool enabled) |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| _sendingMedia = enabled; |
| } |
| |
| bool |
| RTPSender::SendingMedia() const |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _sendingMedia; |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::Timestamp() const |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _timeStamp; |
| } |
| |
| |
| WebRtc_Word32 |
| RTPSender::SetStartTimestamp( const WebRtc_UWord32 timestamp, const bool force) |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| if(force) |
| { |
| _startTimeStampForced = force; |
| _startTimeStamp = timestamp; |
| } else |
| { |
| if(!_startTimeStampForced) |
| { |
| _startTimeStamp = timestamp; |
| } |
| } |
| return 0; |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::StartTimestamp() const |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _startTimeStamp; |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::GenerateNewSSRC() |
| { |
| // if configured via API, return 0 |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| if(_ssrcForced) |
| { |
| return 0; |
| } |
| _ssrc = _ssrcDB.CreateSSRC(); // can't be 0 |
| return _ssrc; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SetSSRC(WebRtc_UWord32 ssrc) |
| { |
| // this is configured via the API |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| if (_ssrc == ssrc && _ssrcForced) |
| { |
| return 0; // since it's same ssrc, don't reset anything |
| } |
| |
| _ssrcForced = true; |
| |
| _ssrcDB.ReturnSSRC(_ssrc); |
| _ssrcDB.RegisterSSRC(ssrc); |
| _ssrc = ssrc; |
| |
| if(!_sequenceNumberForced) |
| { |
| _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); |
| } |
| return 0; |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::SSRC() const |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _ssrc; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SetCSRCStatus(const bool include) |
| { |
| _includeCSRCs = include; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], |
| const WebRtc_UWord8 arrLength) |
| { |
| if(arrLength > kRtpCsrcSize) |
| { |
| assert(false); |
| return -1; |
| } |
| |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| for(int i = 0; i < arrLength;i++) |
| { |
| _CSRC[i] = arrOfCSRC[i]; |
| } |
| _CSRCs = arrLength; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| |
| if(arrOfCSRC == NULL) |
| { |
| assert(false); |
| return -1; |
| } |
| for(int i = 0; i < _CSRCs && i < kRtpCsrcSize;i++) |
| { |
| arrOfCSRC[i] = _CSRC[i]; |
| } |
| return _CSRCs; |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SetSequenceNumber(WebRtc_UWord16 seq) |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| _sequenceNumberForced = true; |
| _sequenceNumber = seq; |
| return 0; |
| } |
| |
| WebRtc_UWord16 |
| RTPSender::SequenceNumber() const |
| { |
| CriticalSectionScoped cs(_sendCritsect); |
| return _sequenceNumber; |
| } |
| |
| |
| /* |
| * Audio |
| */ |
| WebRtc_Word32 |
| RTPSender::RegisterAudioCallback(RtpAudioFeedback* messagesCallback) |
| { |
| if(!_audioConfigured) |
| { |
| return -1; |
| } |
| return _audio->RegisterAudioCallback(messagesCallback); |
| } |
| |
| // Send a DTMF tone, RFC 2833 (4733) |
| WebRtc_Word32 |
| RTPSender::SendTelephoneEvent(const WebRtc_UWord8 key, |
| const WebRtc_UWord16 time_ms, |
| const WebRtc_UWord8 level) |
| { |
| if(!_audioConfigured) |
| { |
| return -1; |
| } |
| return _audio->SendTelephoneEvent(key, time_ms, level); |
| } |
| |
| bool |
| RTPSender::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const |
| { |
| if(!_audioConfigured) |
| { |
| return false; |
| } |
| return _audio->SendTelephoneEventActive(telephoneEvent); |
| } |
| |
| // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG) |
| WebRtc_Word32 |
| RTPSender::SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples) |
| { |
| if(!_audioConfigured) |
| { |
| return -1; |
| } |
| return _audio->SetAudioPacketSize(packetSizeSamples); |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SetAudioLevelIndicationStatus(const bool enable, |
| const WebRtc_UWord8 ID) |
| { |
| if(!_audioConfigured) |
| { |
| return -1; |
| } |
| return _audio->SetAudioLevelIndicationStatus(enable, ID); |
| } |
| |
| WebRtc_Word32 |
| RTPSender::AudioLevelIndicationStatus(bool& enable, |
| WebRtc_UWord8& ID) const |
| { |
| return _audio->AudioLevelIndicationStatus(enable, ID); |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SetAudioLevel(const WebRtc_UWord8 level_dBov) |
| { |
| return _audio->SetAudioLevel(level_dBov); |
| } |
| |
| // Set payload type for Redundant Audio Data RFC 2198 |
| WebRtc_Word32 |
| RTPSender::SetRED(const WebRtc_Word8 payloadType) |
| { |
| if(!_audioConfigured) |
| { |
| return -1; |
| } |
| return _audio->SetRED(payloadType); |
| } |
| |
| // Get payload type for Redundant Audio Data RFC 2198 |
| WebRtc_Word32 |
| RTPSender::RED(WebRtc_Word8& payloadType) const |
| { |
| if(!_audioConfigured) |
| { |
| return -1; |
| } |
| return _audio->RED(payloadType); |
| } |
| |
| /* |
| * Video |
| */ |
| VideoCodecInformation* |
| RTPSender::CodecInformationVideo() |
| { |
| if(_audioConfigured) |
| { |
| return NULL; |
| } |
| return _video->CodecInformationVideo(); |
| } |
| |
| RtpVideoCodecTypes |
| RTPSender::VideoCodecType() const |
| { |
| if(_audioConfigured) |
| { |
| return kRtpNoVideo; |
| } |
| return _video->VideoCodecType(); |
| } |
| |
| WebRtc_UWord32 |
| RTPSender::MaxConfiguredBitrateVideo() const |
| { |
| if(_audioConfigured) |
| { |
| return 0; |
| } |
| return _video->MaxConfiguredBitrateVideo(); |
| } |
| |
| WebRtc_Word32 |
| RTPSender::SendRTPIntraRequest() |
| { |
| if(_audioConfigured) |
| { |
| return -1; |
| } |
| return _video->SendRTPIntraRequest(); |
| } |
| |
| // FEC |
| WebRtc_Word32 |
| RTPSender::SetGenericFECStatus(const bool enable, |
| const WebRtc_UWord8 payloadTypeRED, |
| const WebRtc_UWord8 payloadTypeFEC) |
| { |
| if(_audioConfigured) |
| { |
| return -1; |
| } |
| return _video->SetGenericFECStatus(enable, payloadTypeRED, payloadTypeFEC); |
| } |
| |
| WebRtc_Word32 |
| RTPSender::GenericFECStatus(bool& enable, |
| WebRtc_UWord8& payloadTypeRED, |
| WebRtc_UWord8& payloadTypeFEC) const |
| { |
| if(_audioConfigured) |
| { |
| return -1; |
| } |
| return _video->GenericFECStatus(enable, payloadTypeRED, payloadTypeFEC); |
| } |
| |
| WebRtc_Word32 RTPSender::SetFecParameters( |
| const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params) { |
| if (_audioConfigured) { |
| return -1; |
| } |
| return _video->SetFecParameters(delta_params, key_params); |
| } |
| } // namespace webrtc |
