/* * 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 #include #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.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 { RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy( int32_t id, RtpData* data_callback) { return new RTPReceiverVideo(id, data_callback); } RTPReceiverVideo::RTPReceiverVideo(int32_t id, RtpData* data_callback) : RTPReceiverStrategy(data_callback), id_(id) {} RTPReceiverVideo::~RTPReceiverVideo() { } 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], int8_t payload_type, uint32_t frequency) { return 0; } int32_t RTPReceiverVideo::ParseRtpPacket( WebRtcRTPHeader* rtp_header, const PayloadUnion& specific_payload, bool is_red, const uint8_t* payload, uint16_t payload_length, int64_t timestamp_ms, bool is_first_packet) { TRACE_EVENT2("webrtc_rtp", "Video::ParseRtp", "seqnum", rtp_header->header.sequenceNumber, "timestamp", rtp_header->header.timestamp); rtp_header->type.Video.codec = specific_payload.Video.videoCodecType; const uint16_t payload_data_length = payload_length - rtp_header->header.paddingLength; if (payload_data_length == 0) return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0 : -1; return ParseVideoCodecSpecific(rtp_header, payload, payload_data_length, specific_payload.Video.videoCodecType, timestamp_ms, is_first_packet); } int RTPReceiverVideo::GetPayloadTypeFrequency() const { return kVideoPayloadTypeFrequency; } RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive( uint16_t last_payload_length) const { return kRtpDead; } int32_t RTPReceiverVideo::InvokeOnInitializeDecoder( RtpFeedback* callback, int32_t id, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion& specific_payload) const { // For video we just go with default values. if (-1 == callback->OnInitializeDecoder( id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) { WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, id, "Failed to create video decoder for payload type:%d", payload_type); return -1; } return 0; } // We are not allowed to hold a critical section when calling this function. int32_t RTPReceiverVideo::ParseVideoCodecSpecific( WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_data_length, RtpVideoCodecTypes video_type, int64_t now_ms, bool is_first_packet) { WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, id_, "%s(timestamp:%u)", __FUNCTION__, rtp_header->header.timestamp); switch (rtp_header->type.Video.codec) { case kRtpVideoGeneric: rtp_header->type.Video.isFirstPacket = is_first_packet; return ReceiveGenericCodec(rtp_header, payload_data, payload_data_length); case kRtpVideoVp8: return ReceiveVp8Codec(rtp_header, payload_data, payload_data_length); case kRtpVideoNone: break; } return -1; } int32_t RTPReceiverVideo::BuildRTPheader( const WebRtcRTPHeader* rtp_header, uint8_t* data_buffer) const { data_buffer[0] = static_cast(0x80); // version 2 data_buffer[1] = static_cast(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::ReceiveVp8Codec(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_data_length) { ModuleRTPUtility::RTPPayload parsed_packet; uint32_t id; { CriticalSectionScoped cs(crit_sect_.get()); id = id_; } ModuleRTPUtility::RTPPayloadParser rtp_payload_parser( kRtpVideoVp8, payload_data, payload_data_length, id); if (!rtp_payload_parser.Parse(parsed_packet)) return -1; if (parsed_packet.info.VP8.dataLength == 0) 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; if (data_callback_->OnReceivedPayloadData( payload_data, payload_data_length, rtp_header) != 0) { return -1; } return 0; } } // namespace webrtc