| /* |
| * 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_receiver_video.h" |
| |
| #include <math.h> |
| |
| #include <cassert> // assert |
| #include <cstring> // memcpy() |
| |
| #include "webrtc/modules/rtp_rtcp/source/receiver_fec.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| #include "webrtc/system_wrappers/interface/trace_event.h" |
| |
| namespace webrtc { |
| uint32_t BitRateBPS(uint16_t x) { |
| return (x & 0x3fff) * uint32_t(pow(10.0f, (2 + (x >> 14)))); |
| } |
| |
| RTPReceiverVideo::RTPReceiverVideo( |
| const int32_t id, |
| const RTPPayloadRegistry* rtp_rtp_payload_registry, |
| RtpData* data_callback) |
| : RTPReceiverStrategy(data_callback), |
| id_(id), |
| rtp_rtp_payload_registry_(rtp_rtp_payload_registry), |
| critical_section_receiver_video_( |
| CriticalSectionWrapper::CreateCriticalSection()), |
| current_fec_frame_decoded_(false), |
| receive_fec_(NULL) { |
| } |
| |
| RTPReceiverVideo::~RTPReceiverVideo() { |
| delete critical_section_receiver_video_; |
| delete receive_fec_; |
| } |
| |
| bool RTPReceiverVideo::ShouldReportCsrcChanges( |
| uint8_t payload_type) const { |
| // Always do this for video packets. |
| return true; |
| } |
| |
| int32_t RTPReceiverVideo::OnNewPayloadTypeCreated( |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const int8_t payload_type, |
| const uint32_t frequency) { |
| if (ModuleRTPUtility::StringCompare(payload_name, "ULPFEC", 6)) { |
| // Enable FEC if not enabled. |
| if (receive_fec_ == NULL) { |
| receive_fec_ = new ReceiverFEC(id_, this); |
| } |
| receive_fec_->SetPayloadTypeFEC(payload_type); |
| } |
| return 0; |
| } |
| |
| int32_t RTPReceiverVideo::ParseRtpPacket( |
| WebRtcRTPHeader* rtp_header, |
| const ModuleRTPUtility::PayloadUnion& specific_payload, |
| const bool is_red, |
| const uint8_t* packet, |
| const uint16_t packet_length, |
| const int64_t timestamp_ms, |
| const bool is_first_packet) { |
| TRACE_EVENT2("webrtc_rtp", "Video::ParseRtp", |
| "seqnum", rtp_header->header.sequenceNumber, |
| "timestamp", rtp_header->header.timestamp); |
| const uint8_t* payload_data = |
| ModuleRTPUtility::GetPayloadData(rtp_header->header, packet); |
| const uint16_t payload_data_length = |
| ModuleRTPUtility::GetPayloadDataLength(rtp_header->header, packet_length); |
| return ParseVideoCodecSpecific(rtp_header, |
| payload_data, |
| payload_data_length, |
| specific_payload.Video.videoCodecType, |
| is_red, |
| packet, |
| packet_length, |
| timestamp_ms, |
| is_first_packet); |
| } |
| |
| int32_t RTPReceiverVideo::GetFrequencyHz() const { |
| return kDefaultVideoFrequency; |
| } |
| |
| RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive( |
| uint16_t last_payload_length) const { |
| return kRtpDead; |
| } |
| |
| int32_t RTPReceiverVideo::InvokeOnInitializeDecoder( |
| RtpFeedback* callback, |
| const int32_t id, |
| const int8_t payload_type, |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const ModuleRTPUtility::PayloadUnion& specific_payload) const { |
| // For video we just go with default values. |
| if (-1 == callback->OnInitializeDecoder( |
| id, payload_type, payload_name, kDefaultVideoFrequency, 1, 0)) { |
| WEBRTC_TRACE(kTraceError, |
| kTraceRtpRtcp, |
| id, |
| "Failed to create video decoder for payload type:%d", |
| payload_type); |
| return -1; |
| } |
| return 0; |
| } |
| |
| // we have no critext when calling this |
| // we are not allowed to have any critsects when calling |
| // CallbackOfReceivedPayloadData |
| int32_t RTPReceiverVideo::ParseVideoCodecSpecific( |
| WebRtcRTPHeader* rtp_header, |
| const uint8_t* payload_data, |
| const uint16_t payload_data_length, |
| const RtpVideoCodecTypes video_type, |
| const bool is_red, |
| const uint8_t* incoming_rtp_packet, |
| const uint16_t incoming_rtp_packet_size, |
| const int64_t now_ms, |
| const bool is_first_packet) { |
| int32_t ret_val = 0; |
| |
| critical_section_receiver_video_->Enter(); |
| |
| if (is_red) { |
| if (receive_fec_ == NULL) { |
| critical_section_receiver_video_->Leave(); |
| return -1; |
| } |
| bool FECpacket = false; |
| ret_val = receive_fec_->AddReceivedFECPacket( |
| rtp_header, incoming_rtp_packet, payload_data_length, FECpacket); |
| if (ret_val != -1) { |
| ret_val = receive_fec_->ProcessReceivedFEC(); |
| } |
| critical_section_receiver_video_->Leave(); |
| |
| if (ret_val == 0 && FECpacket) { |
| // Callback with the received FEC packet. |
| // The normal packets are delivered after parsing. |
| // This contains the original RTP packet header but with |
| // empty payload and data length. |
| rtp_header->frameType = kFrameEmpty; |
| // We need this for the routing. |
| int32_t ret_val = SetCodecType(video_type, rtp_header); |
| if (ret_val != 0) { |
| return ret_val; |
| } |
| // Pass the length of FEC packets so that they can be accounted for in |
| // the bandwidth estimator. |
| ret_val = data_callback_->OnReceivedPayloadData( |
| NULL, payload_data_length, rtp_header); |
| } |
| } else { |
| // will leave the critical_section_receiver_video_ critsect |
| ret_val = ParseVideoCodecSpecificSwitch(rtp_header, |
| payload_data, |
| payload_data_length, |
| video_type, |
| is_first_packet); |
| } |
| return ret_val; |
| } |
| |
| int32_t RTPReceiverVideo::BuildRTPheader( |
| const WebRtcRTPHeader* rtp_header, |
| uint8_t* data_buffer) const { |
| data_buffer[0] = static_cast<uint8_t>(0x80); // version 2 |
| data_buffer[1] = static_cast<uint8_t>(rtp_header->header.payloadType); |
| if (rtp_header->header.markerBit) { |
| data_buffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1 |
| } |
| ModuleRTPUtility::AssignUWord16ToBuffer(data_buffer + 2, |
| rtp_header->header.sequenceNumber); |
| ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 4, |
| rtp_header->header.timestamp); |
| ModuleRTPUtility::AssignUWord32ToBuffer(data_buffer + 8, |
| rtp_header->header.ssrc); |
| |
| int32_t rtp_header_length = 12; |
| |
| // Add the CSRCs if any |
| if (rtp_header->header.numCSRCs > 0) { |
| if (rtp_header->header.numCSRCs > 16) { |
| // error |
| assert(false); |
| } |
| uint8_t* ptr = &data_buffer[rtp_header_length]; |
| for (uint32_t i = 0; i < rtp_header->header.numCSRCs; ++i) { |
| ModuleRTPUtility::AssignUWord32ToBuffer(ptr, |
| rtp_header->header.arrOfCSRCs[i]); |
| ptr += 4; |
| } |
| data_buffer[0] = (data_buffer[0] & 0xf0) | rtp_header->header.numCSRCs; |
| // Update length of header |
| rtp_header_length += sizeof(uint32_t) * rtp_header->header.numCSRCs; |
| } |
| return rtp_header_length; |
| } |
| |
| int32_t RTPReceiverVideo::ReceiveRecoveredPacketCallback( |
| WebRtcRTPHeader* rtp_header, |
| const uint8_t* payload_data, |
| const uint16_t payload_data_length) { |
| // TODO(pwestin) Re-factor this to avoid the messy critsect handling. |
| critical_section_receiver_video_->Enter(); |
| |
| current_fec_frame_decoded_ = true; |
| |
| ModuleRTPUtility::Payload* payload = NULL; |
| if (rtp_rtp_payload_registry_->PayloadTypeToPayload( |
| rtp_header->header.payloadType, payload) != 0) { |
| critical_section_receiver_video_->Leave(); |
| return -1; |
| } |
| // here we can re-create the original lost packet so that we can use it for |
| // the relay we need to re-create the RED header too |
| uint8_t recovered_packet[IP_PACKET_SIZE]; |
| uint16_t rtp_header_length = |
| (uint16_t) BuildRTPheader(rtp_header, recovered_packet); |
| |
| const uint8_t kREDForFECHeaderLength = 1; |
| |
| // replace pltype |
| recovered_packet[1] &= 0x80; // Reset. |
| recovered_packet[1] += rtp_rtp_payload_registry_->red_payload_type(); |
| |
| // add RED header |
| recovered_packet[rtp_header_length] = rtp_header->header.payloadType; |
| // f-bit always 0 |
| |
| memcpy(recovered_packet + rtp_header_length + kREDForFECHeaderLength, |
| payload_data, |
| payload_data_length); |
| |
| // A recovered packet can be the first packet, but we lack the ability to |
| // detect it at the moment since we do not store the history of recently |
| // received packets. Most codecs like VP8 deal with this in other ways. |
| bool is_first_packet = false; |
| |
| return ParseVideoCodecSpecificSwitch( |
| rtp_header, |
| payload_data, |
| payload_data_length, |
| payload->typeSpecific.Video.videoCodecType, |
| is_first_packet); |
| } |
| |
| int32_t RTPReceiverVideo::SetCodecType( |
| const RtpVideoCodecTypes video_type, |
| WebRtcRTPHeader* rtp_header) const { |
| switch (video_type) { |
| case kRtpGenericVideo: |
| rtp_header->type.Video.codec = kRTPVideoGeneric; |
| break; |
| case kRtpVp8Video: |
| rtp_header->type.Video.codec = kRTPVideoVP8; |
| break; |
| case kRtpFecVideo: |
| rtp_header->type.Video.codec = kRTPVideoFEC; |
| break; |
| } |
| return 0; |
| } |
| |
| int32_t RTPReceiverVideo::ParseVideoCodecSpecificSwitch( |
| WebRtcRTPHeader* rtp_header, |
| const uint8_t* payload_data, |
| const uint16_t payload_data_length, |
| const RtpVideoCodecTypes video_type, |
| const bool is_first_packet) { |
| int32_t ret_val = SetCodecType(video_type, rtp_header); |
| if (ret_val != 0) { |
| critical_section_receiver_video_->Leave(); |
| return ret_val; |
| } |
| WEBRTC_TRACE(kTraceStream, |
| kTraceRtpRtcp, |
| id_, |
| "%s(timestamp:%u)", |
| __FUNCTION__, |
| rtp_header->header.timestamp); |
| |
| // All receive functions release critical_section_receiver_video_ before |
| // returning. |
| switch (video_type) { |
| case kRtpGenericVideo: |
| rtp_header->type.Video.isFirstPacket = is_first_packet; |
| return ReceiveGenericCodec(rtp_header, payload_data, payload_data_length); |
| case kRtpVp8Video: |
| return ReceiveVp8Codec(rtp_header, payload_data, payload_data_length); |
| case kRtpFecVideo: |
| break; |
| } |
| critical_section_receiver_video_->Leave(); |
| return -1; |
| } |
| |
| int32_t RTPReceiverVideo::ReceiveVp8Codec( |
| WebRtcRTPHeader* rtp_header, |
| const uint8_t* payload_data, |
| const uint16_t payload_data_length) { |
| bool success; |
| ModuleRTPUtility::RTPPayload parsed_packet; |
| if (payload_data_length == 0) { |
| success = true; |
| parsed_packet.info.VP8.dataLength = 0; |
| } else { |
| ModuleRTPUtility::RTPPayloadParser rtp_payload_parser( |
| kRtpVp8Video, payload_data, payload_data_length, id_); |
| |
| success = rtp_payload_parser.Parse(parsed_packet); |
| } |
| // from here down we only work on local data |
| critical_section_receiver_video_->Leave(); |
| |
| if (!success) { |
| return -1; |
| } |
| if (parsed_packet.info.VP8.dataLength == 0) { |
| // we have an "empty" VP8 packet, it's ok, could be one way video |
| // Inform the jitter buffer about this packet. |
| rtp_header->frameType = kFrameEmpty; |
| if (data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) != 0) { |
| return -1; |
| } |
| return 0; |
| } |
| rtp_header->frameType = (parsed_packet.frameType == ModuleRTPUtility::kIFrame) |
| ? kVideoFrameKey : kVideoFrameDelta; |
| |
| RTPVideoHeaderVP8* to_header = &rtp_header->type.Video.codecHeader.VP8; |
| ModuleRTPUtility::RTPPayloadVP8* from_header = &parsed_packet.info.VP8; |
| |
| rtp_header->type.Video.isFirstPacket = |
| from_header->beginningOfPartition && (from_header->partitionID == 0); |
| to_header->nonReference = from_header->nonReferenceFrame; |
| to_header->pictureId = |
| from_header->hasPictureID ? from_header->pictureID : kNoPictureId; |
| to_header->tl0PicIdx = |
| from_header->hasTl0PicIdx ? from_header->tl0PicIdx : kNoTl0PicIdx; |
| if (from_header->hasTID) { |
| to_header->temporalIdx = from_header->tID; |
| to_header->layerSync = from_header->layerSync; |
| } else { |
| to_header->temporalIdx = kNoTemporalIdx; |
| to_header->layerSync = false; |
| } |
| to_header->keyIdx = from_header->hasKeyIdx ? from_header->keyIdx : kNoKeyIdx; |
| |
| rtp_header->type.Video.width = from_header->frameWidth; |
| rtp_header->type.Video.height = from_header->frameHeight; |
| |
| to_header->partitionId = from_header->partitionID; |
| to_header->beginningOfPartition = from_header->beginningOfPartition; |
| |
| if (data_callback_->OnReceivedPayloadData(parsed_packet.info.VP8.data, |
| parsed_packet.info.VP8.dataLength, |
| rtp_header) != 0) { |
| return -1; |
| } |
| return 0; |
| } |
| |
| int32_t RTPReceiverVideo::ReceiveGenericCodec( |
| WebRtcRTPHeader* rtp_header, |
| const uint8_t* payload_data, |
| uint16_t payload_data_length) { |
| uint8_t generic_header = *payload_data++; |
| --payload_data_length; |
| |
| rtp_header->frameType = |
| ((generic_header & RtpFormatVideoGeneric::kKeyFrameBit) != 0) ? |
| kVideoFrameKey : kVideoFrameDelta; |
| rtp_header->type.Video.isFirstPacket = |
| (generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0; |
| |
| critical_section_receiver_video_->Leave(); |
| |
| if (data_callback_->OnReceivedPayloadData( |
| payload_data, payload_data_length, rtp_header) != 0) { |
| return -1; |
| } |
| return 0; |
| } |
| } // namespace webrtc |
