| /* |
| * 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/video_engine/vie_receiver.h" |
| |
| #include <vector> |
| |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/utility/interface/rtp_dump.h" |
| #include "webrtc/modules/video_coding/main/interface/video_coding.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/tick_util.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| |
| namespace webrtc { |
| |
| ViEReceiver::ViEReceiver(const int32_t channel_id, |
| VideoCodingModule* module_vcm, |
| RemoteBitrateEstimator* remote_bitrate_estimator) |
| : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()), |
| channel_id_(channel_id), |
| rtp_header_parser_(RtpHeaderParser::Create()), |
| rtp_rtcp_(NULL), |
| vcm_(module_vcm), |
| remote_bitrate_estimator_(remote_bitrate_estimator), |
| external_decryption_(NULL), |
| decryption_buffer_(NULL), |
| rtp_dump_(NULL), |
| receiving_(false) { |
| assert(remote_bitrate_estimator); |
| } |
| |
| ViEReceiver::~ViEReceiver() { |
| if (decryption_buffer_) { |
| delete[] decryption_buffer_; |
| decryption_buffer_ = NULL; |
| } |
| if (rtp_dump_) { |
| rtp_dump_->Stop(); |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } |
| } |
| |
| int ViEReceiver::RegisterExternalDecryption(Encryption* decryption) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (external_decryption_) { |
| return -1; |
| } |
| decryption_buffer_ = new uint8_t[kViEMaxMtu]; |
| if (decryption_buffer_ == NULL) { |
| return -1; |
| } |
| external_decryption_ = decryption; |
| return 0; |
| } |
| |
| int ViEReceiver::DeregisterExternalDecryption() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (external_decryption_ == NULL) { |
| return -1; |
| } |
| external_decryption_ = NULL; |
| return 0; |
| } |
| |
| void ViEReceiver::SetRtpRtcpModule(RtpRtcp* module) { |
| rtp_rtcp_ = module; |
| } |
| |
| void ViEReceiver::RegisterSimulcastRtpRtcpModules( |
| const std::list<RtpRtcp*>& rtp_modules) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| rtp_rtcp_simulcast_.clear(); |
| |
| if (!rtp_modules.empty()) { |
| rtp_rtcp_simulcast_.insert(rtp_rtcp_simulcast_.begin(), |
| rtp_modules.begin(), |
| rtp_modules.end()); |
| } |
| } |
| |
| bool ViEReceiver::SetReceiveTimestampOffsetStatus(bool enable, int id) { |
| if (enable) { |
| return rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, id); |
| } else { |
| return rtp_header_parser_->DeregisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset); |
| } |
| } |
| |
| bool ViEReceiver::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) { |
| if (enable) { |
| return rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, id); |
| } else { |
| return rtp_header_parser_->DeregisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime); |
| } |
| } |
| |
| int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, |
| int rtp_packet_length) { |
| if (!receiving_) { |
| return -1; |
| } |
| return InsertRTPPacket(static_cast<const int8_t*>(rtp_packet), |
| rtp_packet_length); |
| } |
| |
| int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet, |
| int rtcp_packet_length) { |
| if (!receiving_) { |
| return -1; |
| } |
| return InsertRTCPPacket(static_cast<const int8_t*>(rtcp_packet), |
| rtcp_packet_length); |
| } |
| |
| int32_t ViEReceiver::OnReceivedPayloadData( |
| const uint8_t* payload_data, const uint16_t payload_size, |
| const WebRtcRTPHeader* rtp_header) { |
| if (rtp_header == NULL) { |
| return 0; |
| } |
| if (vcm_->IncomingPacket(payload_data, payload_size, *rtp_header) != 0) { |
| // Check this... |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViEReceiver::InsertRTPPacket(const int8_t* rtp_packet, |
| int rtp_packet_length) { |
| // TODO(mflodman) Change decrypt to get rid of this cast. |
| int8_t* tmp_ptr = const_cast<int8_t*>(rtp_packet); |
| unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr); |
| int received_packet_length = rtp_packet_length; |
| |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| |
| if (external_decryption_) { |
| int decrypted_length = kViEMaxMtu; |
| external_decryption_->decrypt(channel_id_, received_packet, |
| decryption_buffer_, received_packet_length, |
| &decrypted_length); |
| if (decrypted_length <= 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_, |
| "RTP decryption failed"); |
| return -1; |
| } else if (decrypted_length > kViEMaxMtu) { |
| WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo, channel_id_, |
| "InsertRTPPacket: %d bytes is allocated as RTP decrytption" |
| " output, external decryption used %d bytes. => memory is " |
| " now corrupted", kViEMaxMtu, decrypted_length); |
| return -1; |
| } |
| received_packet = decryption_buffer_; |
| received_packet_length = decrypted_length; |
| } |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket(received_packet, |
| static_cast<uint16_t>(received_packet_length)); |
| } |
| } |
| RTPHeader header; |
| if (!rtp_header_parser_->Parse(received_packet, received_packet_length, |
| &header)) { |
| WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, channel_id_, |
| "IncomingPacket invalid RTP header"); |
| return -1; |
| } |
| const int payload_size = received_packet_length - header.headerLength; |
| remote_bitrate_estimator_->IncomingPacket(TickTime::MillisecondTimestamp(), |
| payload_size, header); |
| assert(rtp_rtcp_); // Should be set by owner at construction time. |
| return rtp_rtcp_->IncomingRtpPacket(received_packet, received_packet_length, |
| header); |
| } |
| |
| int ViEReceiver::InsertRTCPPacket(const int8_t* rtcp_packet, |
| int rtcp_packet_length) { |
| // TODO(mflodman) Change decrypt to get rid of this cast. |
| int8_t* tmp_ptr = const_cast<int8_t*>(rtcp_packet); |
| unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr); |
| int received_packet_length = rtcp_packet_length; |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| |
| if (external_decryption_) { |
| int decrypted_length = kViEMaxMtu; |
| external_decryption_->decrypt_rtcp(channel_id_, received_packet, |
| decryption_buffer_, |
| received_packet_length, |
| &decrypted_length); |
| if (decrypted_length <= 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_, |
| "RTP decryption failed"); |
| return -1; |
| } else if (decrypted_length > kViEMaxMtu) { |
| WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo, channel_id_, |
| "InsertRTCPPacket: %d bytes is allocated as RTP " |
| " decrytption output, external decryption used %d bytes. " |
| " => memory is now corrupted", |
| kViEMaxMtu, decrypted_length); |
| return -1; |
| } |
| received_packet = decryption_buffer_; |
| received_packet_length = decrypted_length; |
| } |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket( |
| received_packet, static_cast<uint16_t>(received_packet_length)); |
| } |
| } |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin(); |
| while (it != rtp_rtcp_simulcast_.end()) { |
| RtpRtcp* rtp_rtcp = *it++; |
| rtp_rtcp->IncomingRtcpPacket(received_packet, received_packet_length); |
| } |
| } |
| assert(rtp_rtcp_); // Should be set by owner at construction time. |
| return rtp_rtcp_->IncomingRtcpPacket(received_packet, received_packet_length); |
| } |
| |
| void ViEReceiver::StartReceive() { |
| receiving_ = true; |
| } |
| |
| void ViEReceiver::StopReceive() { |
| receiving_ = false; |
| } |
| |
| int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (rtp_dump_) { |
| // Restart it if it already exists and is started |
| rtp_dump_->Stop(); |
| } else { |
| rtp_dump_ = RtpDump::CreateRtpDump(); |
| if (rtp_dump_ == NULL) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_, |
| "StartRTPDump: Failed to create RTP dump"); |
| return -1; |
| } |
| } |
| if (rtp_dump_->Start(file_nameUTF8) != 0) { |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_, |
| "StartRTPDump: Failed to start RTP dump"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViEReceiver::StopRTPDump() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (rtp_dump_) { |
| if (rtp_dump_->IsActive()) { |
| rtp_dump_->Stop(); |
| } else { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_, |
| "StopRTPDump: Dump not active"); |
| } |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } else { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, channel_id_, |
| "StopRTPDump: RTP dump not started"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| // TODO(holmer): To be moved to ViEChannelGroup. |
| void ViEReceiver::EstimatedReceiveBandwidth( |
| unsigned int* available_bandwidth) const { |
| std::vector<unsigned int> ssrcs; |
| |
| // LatestEstimate returns an error if there is no valid bitrate estimate, but |
| // ViEReceiver instead returns a zero estimate. |
| remote_bitrate_estimator_->LatestEstimate(&ssrcs, available_bandwidth); |
| if (std::find(ssrcs.begin(), ssrcs.end(), rtp_rtcp_->RemoteSSRC()) != |
| ssrcs.end()) { |
| *available_bandwidth /= ssrcs.size(); |
| } else { |
| *available_bandwidth = 0; |
| } |
| } |
| |
| } // namespace webrtc |