| /* |
| * 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 <assert.h> |
| #include <string.h> |
| |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_format.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/logging.h" |
| #include "webrtc/system_wrappers/interface/trace_event.h" |
| |
| namespace webrtc { |
| |
| RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy( |
| RtpData* data_callback) { |
| return new RTPReceiverVideo(data_callback); |
| } |
| |
| RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback) |
| : RTPReceiverStrategy(data_callback) { |
| } |
| |
| 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, |
| size_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 size_t payload_data_length = |
| payload_length - rtp_header->header.paddingLength; |
| |
| if (payload == NULL || payload_data_length == 0) { |
| return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0 |
| : -1; |
| } |
| |
| // We are not allowed to hold a critical section when calling below functions. |
| scoped_ptr<RtpDepacketizer> depacketizer( |
| RtpDepacketizer::Create(rtp_header->type.Video.codec)); |
| if (depacketizer.get() == NULL) { |
| LOG(LS_ERROR) << "Failed to create depacketizer."; |
| return -1; |
| } |
| |
| rtp_header->type.Video.isFirstPacket = is_first_packet; |
| RtpDepacketizer::ParsedPayload parsed_payload; |
| if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length)) |
| return -1; |
| |
| rtp_header->frameType = parsed_payload.frame_type; |
| rtp_header->type = parsed_payload.type; |
| return data_callback_->OnReceivedPayloadData(parsed_payload.payload, |
| parsed_payload.payload_length, |
| rtp_header) == 0 |
| ? 0 |
| : -1; |
| } |
| |
| 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)) { |
| LOG(LS_ERROR) << "Failed to created decoder for payload type: " |
| << payload_type; |
| return -1; |
| } |
| return 0; |
| } |
| |
| 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 |
| } |
| RtpUtility::AssignUWord16ToBuffer(data_buffer + 2, |
| rtp_header->header.sequenceNumber); |
| RtpUtility::AssignUWord32ToBuffer(data_buffer + 4, |
| rtp_header->header.timestamp); |
| RtpUtility::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) { |
| RtpUtility::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; |
| } |
| |
| } // namespace webrtc |