| /* |
| * 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 "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" |
| |
| #include <assert.h> |
| #include <stdlib.h> |
| #include <string.h> |
| |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| #include "webrtc/modules/rtp_rtcp/source/producer_fec.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/logging.h" |
| #include "webrtc/system_wrappers/interface/trace_event.h" |
| |
| namespace webrtc { |
| enum { REDForFECHeaderLength = 1 }; |
| |
| struct RtpPacket { |
| uint16_t rtpHeaderLength; |
| ForwardErrorCorrection::Packet* pkt; |
| }; |
| |
| RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender) |
| : _rtpSender(*rtpSender), |
| _sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()), |
| _videoType(kRtpVideoGeneric), |
| _videoCodecInformation(NULL), |
| _maxBitrate(0), |
| _retransmissionSettings(kRetransmitBaseLayer), |
| |
| // Generic FEC |
| _fec(), |
| _fecEnabled(false), |
| _payloadTypeRED(-1), |
| _payloadTypeFEC(-1), |
| _numberFirstPartition(0), |
| delta_fec_params_(), |
| key_fec_params_(), |
| producer_fec_(&_fec), |
| _fecOverheadRate(clock, NULL), |
| _videoBitrate(clock, NULL) { |
| memset(&delta_fec_params_, 0, sizeof(delta_fec_params_)); |
| memset(&key_fec_params_, 0, sizeof(key_fec_params_)); |
| delta_fec_params_.max_fec_frames = key_fec_params_.max_fec_frames = 1; |
| delta_fec_params_.fec_mask_type = key_fec_params_.fec_mask_type = |
| kFecMaskRandom; |
| } |
| |
| RTPSenderVideo::~RTPSenderVideo() { |
| if (_videoCodecInformation) { |
| delete _videoCodecInformation; |
| } |
| delete _sendVideoCritsect; |
| } |
| |
| void RTPSenderVideo::SetVideoCodecType(RtpVideoCodecTypes videoType) { |
| CriticalSectionScoped cs(_sendVideoCritsect); |
| _videoType = videoType; |
| } |
| |
| RtpVideoCodecTypes RTPSenderVideo::VideoCodecType() const { |
| return _videoType; |
| } |
| |
| int32_t RTPSenderVideo::RegisterVideoPayload( |
| const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| const int8_t payloadType, |
| const uint32_t maxBitRate, |
| RtpUtility::Payload*& payload) { |
| CriticalSectionScoped cs(_sendVideoCritsect); |
| |
| RtpVideoCodecTypes videoType = kRtpVideoGeneric; |
| if (RtpUtility::StringCompare(payloadName, "VP8", 3)) { |
| videoType = kRtpVideoVp8; |
| } else if (RtpUtility::StringCompare(payloadName, "H264", 4)) { |
| videoType = kRtpVideoH264; |
| } else if (RtpUtility::StringCompare(payloadName, "I420", 4)) { |
| videoType = kRtpVideoGeneric; |
| } else { |
| videoType = kRtpVideoGeneric; |
| } |
| payload = new RtpUtility::Payload; |
| payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; |
| strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1); |
| payload->typeSpecific.Video.videoCodecType = videoType; |
| payload->typeSpecific.Video.maxRate = maxBitRate; |
| payload->audio = false; |
| return 0; |
| } |
| |
| int32_t RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer, |
| const size_t payload_length, |
| const size_t rtp_header_length, |
| const uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| StorageType storage, |
| bool protect) { |
| if (_fecEnabled) { |
| int ret = 0; |
| size_t fec_overhead_sent = 0; |
| size_t video_sent = 0; |
| |
| RedPacket* red_packet = producer_fec_.BuildRedPacket( |
| data_buffer, payload_length, rtp_header_length, _payloadTypeRED); |
| TRACE_EVENT_INSTANT2("webrtc_rtp", |
| "Video::PacketRed", |
| "timestamp", |
| capture_timestamp, |
| "seqnum", |
| _rtpSender.SequenceNumber()); |
| // Sending the media packet with RED header. |
| int packet_success = |
| _rtpSender.SendToNetwork(red_packet->data(), |
| red_packet->length() - rtp_header_length, |
| rtp_header_length, |
| capture_time_ms, |
| storage, |
| PacedSender::kNormalPriority); |
| |
| ret |= packet_success; |
| |
| if (packet_success == 0) { |
| video_sent += red_packet->length(); |
| } |
| delete red_packet; |
| red_packet = NULL; |
| |
| if (protect) { |
| ret = producer_fec_.AddRtpPacketAndGenerateFec( |
| data_buffer, payload_length, rtp_header_length); |
| if (ret != 0) |
| return ret; |
| } |
| |
| while (producer_fec_.FecAvailable()) { |
| red_packet = |
| producer_fec_.GetFecPacket(_payloadTypeRED, |
| _payloadTypeFEC, |
| _rtpSender.IncrementSequenceNumber(), |
| rtp_header_length); |
| StorageType storage = kDontRetransmit; |
| if (_retransmissionSettings & kRetransmitFECPackets) { |
| storage = kAllowRetransmission; |
| } |
| TRACE_EVENT_INSTANT2("webrtc_rtp", |
| "Video::PacketFec", |
| "timestamp", |
| capture_timestamp, |
| "seqnum", |
| _rtpSender.SequenceNumber()); |
| // Sending FEC packet with RED header. |
| int packet_success = |
| _rtpSender.SendToNetwork(red_packet->data(), |
| red_packet->length() - rtp_header_length, |
| rtp_header_length, |
| capture_time_ms, |
| storage, |
| PacedSender::kNormalPriority); |
| |
| ret |= packet_success; |
| |
| if (packet_success == 0) { |
| fec_overhead_sent += red_packet->length(); |
| } |
| delete red_packet; |
| red_packet = NULL; |
| } |
| _videoBitrate.Update(video_sent); |
| _fecOverheadRate.Update(fec_overhead_sent); |
| return ret; |
| } |
| TRACE_EVENT_INSTANT2("webrtc_rtp", |
| "Video::PacketNormal", |
| "timestamp", |
| capture_timestamp, |
| "seqnum", |
| _rtpSender.SequenceNumber()); |
| int ret = _rtpSender.SendToNetwork(data_buffer, |
| payload_length, |
| rtp_header_length, |
| capture_time_ms, |
| storage, |
| PacedSender::kNormalPriority); |
| if (ret == 0) { |
| _videoBitrate.Update(payload_length + rtp_header_length); |
| } |
| return ret; |
| } |
| |
| int32_t RTPSenderVideo::SendRTPIntraRequest() { |
| // RFC 2032 |
| // 5.2.1. Full intra-frame Request (FIR) packet |
| |
| size_t length = 8; |
| uint8_t data[8]; |
| data[0] = 0x80; |
| data[1] = 192; |
| data[2] = 0; |
| data[3] = 1; // length |
| |
| RtpUtility::AssignUWord32ToBuffer(data + 4, _rtpSender.SSRC()); |
| |
| TRACE_EVENT_INSTANT1("webrtc_rtp", |
| "Video::IntraRequest", |
| "seqnum", |
| _rtpSender.SequenceNumber()); |
| return _rtpSender.SendToNetwork( |
| data, 0, length, -1, kDontStore, PacedSender::kNormalPriority); |
| } |
| |
| int32_t RTPSenderVideo::SetGenericFECStatus(const bool enable, |
| const uint8_t payloadTypeRED, |
| const uint8_t payloadTypeFEC) { |
| _fecEnabled = enable; |
| _payloadTypeRED = payloadTypeRED; |
| _payloadTypeFEC = payloadTypeFEC; |
| memset(&delta_fec_params_, 0, sizeof(delta_fec_params_)); |
| memset(&key_fec_params_, 0, sizeof(key_fec_params_)); |
| delta_fec_params_.max_fec_frames = key_fec_params_.max_fec_frames = 1; |
| delta_fec_params_.fec_mask_type = key_fec_params_.fec_mask_type = |
| kFecMaskRandom; |
| return 0; |
| } |
| |
| int32_t RTPSenderVideo::GenericFECStatus(bool& enable, |
| uint8_t& payloadTypeRED, |
| uint8_t& payloadTypeFEC) const { |
| enable = _fecEnabled; |
| payloadTypeRED = _payloadTypeRED; |
| payloadTypeFEC = _payloadTypeFEC; |
| return 0; |
| } |
| |
| size_t RTPSenderVideo::FECPacketOverhead() const { |
| if (_fecEnabled) { |
| // Overhead is FEC headers plus RED for FEC header plus anything in RTP |
| // header beyond the 12 bytes base header (CSRC list, extensions...) |
| // This reason for the header extensions to be included here is that |
| // from an FEC viewpoint, they are part of the payload to be protected. |
| // (The base RTP header is already protected by the FEC header.) |
| return ForwardErrorCorrection::PacketOverhead() + REDForFECHeaderLength + |
| (_rtpSender.RTPHeaderLength() - kRtpHeaderSize); |
| } |
| return 0; |
| } |
| |
| int32_t RTPSenderVideo::SetFecParameters( |
| const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params) { |
| assert(delta_params); |
| assert(key_params); |
| delta_fec_params_ = *delta_params; |
| key_fec_params_ = *key_params; |
| return 0; |
| } |
| |
| int32_t RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType, |
| const FrameType frameType, |
| const int8_t payloadType, |
| const uint32_t captureTimeStamp, |
| int64_t capture_time_ms, |
| const uint8_t* payloadData, |
| const size_t payloadSize, |
| const RTPFragmentationHeader* fragmentation, |
| VideoCodecInformation* codecInfo, |
| const RTPVideoTypeHeader* rtpTypeHdr) { |
| if (payloadSize == 0) { |
| return -1; |
| } |
| |
| if (frameType == kVideoFrameKey) { |
| producer_fec_.SetFecParameters(&key_fec_params_, _numberFirstPartition); |
| } else { |
| producer_fec_.SetFecParameters(&delta_fec_params_, _numberFirstPartition); |
| } |
| |
| // Default setting for number of first partition packets: |
| // Will be extracted in SendVP8 for VP8 codec; other codecs use 0 |
| _numberFirstPartition = 0; |
| |
| return Send(videoType, |
| frameType, |
| payloadType, |
| captureTimeStamp, |
| capture_time_ms, |
| payloadData, |
| payloadSize, |
| fragmentation, |
| rtpTypeHdr) |
| ? 0 |
| : -1; |
| } |
| |
| VideoCodecInformation* RTPSenderVideo::CodecInformationVideo() { |
| return _videoCodecInformation; |
| } |
| |
| void RTPSenderVideo::SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate) { |
| _maxBitrate = maxBitrate; |
| } |
| |
| uint32_t RTPSenderVideo::MaxConfiguredBitrateVideo() const { |
| return _maxBitrate; |
| } |
| |
| bool RTPSenderVideo::Send(const RtpVideoCodecTypes videoType, |
| const FrameType frameType, |
| const int8_t payloadType, |
| const uint32_t captureTimeStamp, |
| int64_t capture_time_ms, |
| const uint8_t* payloadData, |
| const size_t payloadSize, |
| const RTPFragmentationHeader* fragmentation, |
| const RTPVideoTypeHeader* rtpTypeHdr) { |
| uint16_t rtp_header_length = _rtpSender.RTPHeaderLength(); |
| size_t payload_bytes_to_send = payloadSize; |
| const uint8_t* data = payloadData; |
| size_t max_payload_length = _rtpSender.MaxDataPayloadLength(); |
| |
| scoped_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create( |
| videoType, max_payload_length, rtpTypeHdr, frameType)); |
| |
| // TODO(changbin): we currently don't support to configure the codec to |
| // output multiple partitions for VP8. Should remove below check after the |
| // issue is fixed. |
| const RTPFragmentationHeader* frag = |
| (videoType == kRtpVideoVp8) ? NULL : fragmentation; |
| |
| packetizer->SetPayloadData(data, payload_bytes_to_send, frag); |
| |
| bool last = false; |
| while (!last) { |
| uint8_t dataBuffer[IP_PACKET_SIZE] = {0}; |
| size_t payload_bytes_in_packet = 0; |
| if (!packetizer->NextPacket( |
| &dataBuffer[rtp_header_length], &payload_bytes_in_packet, &last)) { |
| return false; |
| } |
| |
| // Write RTP header. |
| // Set marker bit true if this is the last packet in frame. |
| _rtpSender.BuildRTPheader( |
| dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms); |
| if (SendVideoPacket(dataBuffer, |
| payload_bytes_in_packet, |
| rtp_header_length, |
| captureTimeStamp, |
| capture_time_ms, |
| packetizer->GetStorageType(_retransmissionSettings), |
| packetizer->GetProtectionType() == kProtectedPacket)) { |
| LOG(LS_WARNING) << packetizer->ToString() |
| << " failed to send packet number " |
| << _rtpSender.SequenceNumber(); |
| } |
| } |
| |
| TRACE_EVENT_ASYNC_END1( |
| "webrtc", "Video", capture_time_ms, "timestamp", _rtpSender.Timestamp()); |
| return true; |
| } |
| |
| void RTPSenderVideo::ProcessBitrate() { |
| _videoBitrate.Process(); |
| _fecOverheadRate.Process(); |
| } |
| |
| uint32_t RTPSenderVideo::VideoBitrateSent() const { |
| return _videoBitrate.BitrateLast(); |
| } |
| |
| uint32_t RTPSenderVideo::FecOverheadRate() const { |
| return _fecOverheadRate.BitrateLast(); |
| } |
| |
| int RTPSenderVideo::SelectiveRetransmissions() const { |
| return _retransmissionSettings; |
| } |
| |
| int RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) { |
| _retransmissionSettings = settings; |
| return 0; |
| } |
| |
| } // namespace webrtc |