| Index: webrtc/video/vie_receiver.cc
|
| diff --git a/webrtc/video/vie_receiver.cc b/webrtc/video/vie_receiver.cc
|
| deleted file mode 100644
|
| index 98c8c5dbc09dfb50fd891e7528bff1e2d2ca42a2..0000000000000000000000000000000000000000
|
| --- a/webrtc/video/vie_receiver.cc
|
| +++ /dev/null
|
| @@ -1,482 +0,0 @@
|
| -/*
|
| - * 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/vie_receiver.h"
|
| -
|
| -#include <vector>
|
| -
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
|
| -#include "webrtc/modules/video_coding/include/video_coding.h"
|
| -#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
|
| -#include "webrtc/system_wrappers/include/metrics.h"
|
| -#include "webrtc/system_wrappers/include/tick_util.h"
|
| -#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
|
| -#include "webrtc/system_wrappers/include/trace.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -static const int kPacketLogIntervalMs = 10000;
|
| -
|
| -ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm,
|
| - RemoteBitrateEstimator* remote_bitrate_estimator,
|
| - RtpFeedback* rtp_feedback)
|
| - : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
|
| - clock_(Clock::GetRealTimeClock()),
|
| - rtp_header_parser_(RtpHeaderParser::Create()),
|
| - rtp_payload_registry_(
|
| - new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
|
| - rtp_receiver_(
|
| - RtpReceiver::CreateVideoReceiver(clock_,
|
| - this,
|
| - rtp_feedback,
|
| - rtp_payload_registry_.get())),
|
| - rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
|
| - fec_receiver_(FecReceiver::Create(this)),
|
| - rtp_rtcp_(NULL),
|
| - vcm_(module_vcm),
|
| - remote_bitrate_estimator_(remote_bitrate_estimator),
|
| - ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
|
| - receiving_(false),
|
| - restored_packet_in_use_(false),
|
| - receiving_ast_enabled_(false),
|
| - receiving_cvo_enabled_(false),
|
| - receiving_tsn_enabled_(false),
|
| - last_packet_log_ms_(-1) {
|
| - assert(remote_bitrate_estimator);
|
| -}
|
| -
|
| -ViEReceiver::~ViEReceiver() {
|
| - UpdateHistograms();
|
| -}
|
| -
|
| -void ViEReceiver::UpdateHistograms() {
|
| - FecPacketCounter counter = fec_receiver_->GetPacketCounter();
|
| - if (counter.num_packets > 0) {
|
| - RTC_HISTOGRAM_PERCENTAGE(
|
| - "WebRTC.Video.ReceivedFecPacketsInPercent",
|
| - static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
|
| - }
|
| - if (counter.num_fec_packets > 0) {
|
| - RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
|
| - static_cast<int>(counter.num_recovered_packets *
|
| - 100 / counter.num_fec_packets));
|
| - }
|
| -}
|
| -
|
| -bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
|
| - int8_t old_pltype = -1;
|
| - if (rtp_payload_registry_->ReceivePayloadType(video_codec.plName,
|
| - kVideoPayloadTypeFrequency,
|
| - 0,
|
| - video_codec.maxBitrate,
|
| - &old_pltype) != -1) {
|
| - rtp_payload_registry_->DeRegisterReceivePayload(old_pltype);
|
| - }
|
| -
|
| - return RegisterPayload(video_codec);
|
| -}
|
| -
|
| -bool ViEReceiver::RegisterPayload(const VideoCodec& video_codec) {
|
| - return rtp_receiver_->RegisterReceivePayload(video_codec.plName,
|
| - video_codec.plType,
|
| - kVideoPayloadTypeFrequency,
|
| - 0,
|
| - video_codec.maxBitrate) == 0;
|
| -}
|
| -
|
| -void ViEReceiver::SetNackStatus(bool enable,
|
| - int max_nack_reordering_threshold) {
|
| - if (!enable) {
|
| - // Reset the threshold back to the lower default threshold when NACK is
|
| - // disabled since we no longer will be receiving retransmissions.
|
| - max_nack_reordering_threshold = kDefaultMaxReorderingThreshold;
|
| - }
|
| - rtp_receive_statistics_->SetMaxReorderingThreshold(
|
| - max_nack_reordering_threshold);
|
| - rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
|
| -}
|
| -
|
| -void ViEReceiver::SetRtxPayloadType(int payload_type,
|
| - int associated_payload_type) {
|
| - rtp_payload_registry_->SetRtxPayloadType(payload_type,
|
| - associated_payload_type);
|
| -}
|
| -
|
| -void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) {
|
| - rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(val);
|
| -}
|
| -
|
| -void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
|
| - rtp_payload_registry_->SetRtxSsrc(ssrc);
|
| -}
|
| -
|
| -bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const {
|
| - return rtp_payload_registry_->GetRtxSsrc(ssrc);
|
| -}
|
| -
|
| -bool ViEReceiver::IsFecEnabled() const {
|
| - return rtp_payload_registry_->ulpfec_payload_type() > -1;
|
| -}
|
| -
|
| -uint32_t ViEReceiver::GetRemoteSsrc() const {
|
| - return rtp_receiver_->SSRC();
|
| -}
|
| -
|
| -int ViEReceiver::GetCsrcs(uint32_t* csrcs) const {
|
| - return rtp_receiver_->CSRCs(csrcs);
|
| -}
|
| -
|
| -void ViEReceiver::SetRtpRtcpModule(RtpRtcp* module) {
|
| - rtp_rtcp_ = module;
|
| -}
|
| -
|
| -RtpReceiver* ViEReceiver::GetRtpReceiver() const {
|
| - return rtp_receiver_.get();
|
| -}
|
| -
|
| -void ViEReceiver::RegisterRtpRtcpModules(
|
| - const std::vector<RtpRtcp*>& rtp_modules) {
|
| - CriticalSectionScoped cs(receive_cs_.get());
|
| - // Only change the "simulcast" modules, the base module can be accessed
|
| - // without a lock whereas the simulcast modules require locking as they can be
|
| - // changed in runtime.
|
| - rtp_rtcp_simulcast_ =
|
| - std::vector<RtpRtcp*>(rtp_modules.begin() + 1, 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) {
|
| - if (rtp_header_parser_->RegisterRtpHeaderExtension(
|
| - kRtpExtensionAbsoluteSendTime, id)) {
|
| - receiving_ast_enabled_ = true;
|
| - return true;
|
| - } else {
|
| - return false;
|
| - }
|
| - } else {
|
| - receiving_ast_enabled_ = false;
|
| - return rtp_header_parser_->DeregisterRtpHeaderExtension(
|
| - kRtpExtensionAbsoluteSendTime);
|
| - }
|
| -}
|
| -
|
| -bool ViEReceiver::SetReceiveVideoRotationStatus(bool enable, int id) {
|
| - if (enable) {
|
| - if (rtp_header_parser_->RegisterRtpHeaderExtension(
|
| - kRtpExtensionVideoRotation, id)) {
|
| - receiving_cvo_enabled_ = true;
|
| - return true;
|
| - } else {
|
| - return false;
|
| - }
|
| - } else {
|
| - receiving_cvo_enabled_ = false;
|
| - return rtp_header_parser_->DeregisterRtpHeaderExtension(
|
| - kRtpExtensionVideoRotation);
|
| - }
|
| -}
|
| -
|
| -bool ViEReceiver::SetReceiveTransportSequenceNumber(bool enable, int id) {
|
| - if (enable) {
|
| - if (rtp_header_parser_->RegisterRtpHeaderExtension(
|
| - kRtpExtensionTransportSequenceNumber, id)) {
|
| - receiving_tsn_enabled_ = true;
|
| - return true;
|
| - } else {
|
| - return false;
|
| - }
|
| - } else {
|
| - receiving_tsn_enabled_ = false;
|
| - return rtp_header_parser_->DeregisterRtpHeaderExtension(
|
| - kRtpExtensionTransportSequenceNumber);
|
| - }
|
| -}
|
| -
|
| -int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
|
| - size_t rtp_packet_length,
|
| - const PacketTime& packet_time) {
|
| - return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet),
|
| - rtp_packet_length, packet_time);
|
| -}
|
| -
|
| -int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet,
|
| - size_t rtcp_packet_length) {
|
| - return InsertRTCPPacket(static_cast<const uint8_t*>(rtcp_packet),
|
| - rtcp_packet_length);
|
| -}
|
| -
|
| -int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
|
| - const size_t payload_size,
|
| - const WebRtcRTPHeader* rtp_header) {
|
| - WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
|
| - rtp_header_with_ntp.ntp_time_ms =
|
| - ntp_estimator_->Estimate(rtp_header->header.timestamp);
|
| - if (vcm_->IncomingPacket(payload_data,
|
| - payload_size,
|
| - rtp_header_with_ntp) != 0) {
|
| - // Check this...
|
| - return -1;
|
| - }
|
| - return 0;
|
| -}
|
| -
|
| -bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
|
| - size_t rtp_packet_length) {
|
| - RTPHeader header;
|
| - if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
|
| - return false;
|
| - }
|
| - header.payload_type_frequency = kVideoPayloadTypeFrequency;
|
| - bool in_order = IsPacketInOrder(header);
|
| - return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
|
| -}
|
| -
|
| -int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
|
| - size_t rtp_packet_length,
|
| - const PacketTime& packet_time) {
|
| - {
|
| - CriticalSectionScoped cs(receive_cs_.get());
|
| - if (!receiving_) {
|
| - return -1;
|
| - }
|
| - }
|
| -
|
| - RTPHeader header;
|
| - if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
|
| - &header)) {
|
| - return -1;
|
| - }
|
| - size_t payload_length = rtp_packet_length - header.headerLength;
|
| - int64_t arrival_time_ms;
|
| - int64_t now_ms = clock_->TimeInMilliseconds();
|
| - if (packet_time.timestamp != -1)
|
| - arrival_time_ms = (packet_time.timestamp + 500) / 1000;
|
| - else
|
| - arrival_time_ms = now_ms;
|
| -
|
| - {
|
| - // Periodically log the RTP header of incoming packets.
|
| - CriticalSectionScoped cs(receive_cs_.get());
|
| - if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
|
| - std::stringstream ss;
|
| - ss << "Packet received on SSRC: " << header.ssrc << " with payload type: "
|
| - << static_cast<int>(header.payloadType) << ", timestamp: "
|
| - << header.timestamp << ", sequence number: " << header.sequenceNumber
|
| - << ", arrival time: " << arrival_time_ms;
|
| - if (header.extension.hasTransmissionTimeOffset)
|
| - ss << ", toffset: " << header.extension.transmissionTimeOffset;
|
| - if (header.extension.hasAbsoluteSendTime)
|
| - ss << ", abs send time: " << header.extension.absoluteSendTime;
|
| - LOG(LS_INFO) << ss.str();
|
| - last_packet_log_ms_ = now_ms;
|
| - }
|
| - }
|
| -
|
| - remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length,
|
| - header, true);
|
| - header.payload_type_frequency = kVideoPayloadTypeFrequency;
|
| -
|
| - bool in_order = IsPacketInOrder(header);
|
| - rtp_payload_registry_->SetIncomingPayloadType(header);
|
| - int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order)
|
| - ? 0
|
| - : -1;
|
| - // Update receive statistics after ReceivePacket.
|
| - // Receive statistics will be reset if the payload type changes (make sure
|
| - // that the first packet is included in the stats).
|
| - rtp_receive_statistics_->IncomingPacket(
|
| - header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
|
| - return ret;
|
| -}
|
| -
|
| -bool ViEReceiver::ReceivePacket(const uint8_t* packet,
|
| - size_t packet_length,
|
| - const RTPHeader& header,
|
| - bool in_order) {
|
| - if (rtp_payload_registry_->IsEncapsulated(header)) {
|
| - return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
|
| - }
|
| - const uint8_t* payload = packet + header.headerLength;
|
| - assert(packet_length >= header.headerLength);
|
| - size_t payload_length = packet_length - header.headerLength;
|
| - PayloadUnion payload_specific;
|
| - if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
|
| - &payload_specific)) {
|
| - return false;
|
| - }
|
| - return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
|
| - payload_specific, in_order);
|
| -}
|
| -
|
| -bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
|
| - size_t packet_length,
|
| - const RTPHeader& header) {
|
| - if (rtp_payload_registry_->IsRed(header)) {
|
| - int8_t ulpfec_pt = rtp_payload_registry_->ulpfec_payload_type();
|
| - if (packet[header.headerLength] == ulpfec_pt) {
|
| - rtp_receive_statistics_->FecPacketReceived(header, packet_length);
|
| - // Notify vcm about received FEC packets to avoid NACKing these packets.
|
| - NotifyReceiverOfFecPacket(header);
|
| - }
|
| - if (fec_receiver_->AddReceivedRedPacket(
|
| - header, packet, packet_length, ulpfec_pt) != 0) {
|
| - return false;
|
| - }
|
| - return fec_receiver_->ProcessReceivedFec() == 0;
|
| - } else if (rtp_payload_registry_->IsRtx(header)) {
|
| - if (header.headerLength + header.paddingLength == packet_length) {
|
| - // This is an empty packet and should be silently dropped before trying to
|
| - // parse the RTX header.
|
| - return true;
|
| - }
|
| - // Remove the RTX header and parse the original RTP header.
|
| - if (packet_length < header.headerLength)
|
| - return false;
|
| - if (packet_length > sizeof(restored_packet_))
|
| - return false;
|
| - CriticalSectionScoped cs(receive_cs_.get());
|
| - if (restored_packet_in_use_) {
|
| - LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
|
| - return false;
|
| - }
|
| - if (!rtp_payload_registry_->RestoreOriginalPacket(
|
| - restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
|
| - header)) {
|
| - LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header";
|
| - return false;
|
| - }
|
| - restored_packet_in_use_ = true;
|
| - bool ret = OnRecoveredPacket(restored_packet_, packet_length);
|
| - restored_packet_in_use_ = false;
|
| - return ret;
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
|
| - int8_t last_media_payload_type =
|
| - rtp_payload_registry_->last_received_media_payload_type();
|
| - if (last_media_payload_type < 0) {
|
| - LOG(LS_WARNING) << "Failed to get last media payload type.";
|
| - return;
|
| - }
|
| - // Fake an empty media packet.
|
| - WebRtcRTPHeader rtp_header = {};
|
| - rtp_header.header = header;
|
| - rtp_header.header.payloadType = last_media_payload_type;
|
| - rtp_header.header.paddingLength = 0;
|
| - PayloadUnion payload_specific;
|
| - if (!rtp_payload_registry_->GetPayloadSpecifics(last_media_payload_type,
|
| - &payload_specific)) {
|
| - LOG(LS_WARNING) << "Failed to get payload specifics.";
|
| - return;
|
| - }
|
| - rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
|
| - rtp_header.type.Video.rotation = kVideoRotation_0;
|
| - if (header.extension.hasVideoRotation) {
|
| - rtp_header.type.Video.rotation =
|
| - ConvertCVOByteToVideoRotation(header.extension.videoRotation);
|
| - }
|
| - OnReceivedPayloadData(NULL, 0, &rtp_header);
|
| -}
|
| -
|
| -int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
|
| - size_t rtcp_packet_length) {
|
| - {
|
| - CriticalSectionScoped cs(receive_cs_.get());
|
| - if (!receiving_) {
|
| - return -1;
|
| - }
|
| -
|
| - for (RtpRtcp* rtp_rtcp : rtp_rtcp_simulcast_)
|
| - rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
|
| - }
|
| - assert(rtp_rtcp_); // Should be set by owner at construction time.
|
| - int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
|
| - if (ret != 0) {
|
| - return ret;
|
| - }
|
| -
|
| - int64_t rtt = 0;
|
| - rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
|
| - if (rtt == 0) {
|
| - // Waiting for valid rtt.
|
| - return 0;
|
| - }
|
| - uint32_t ntp_secs = 0;
|
| - uint32_t ntp_frac = 0;
|
| - uint32_t rtp_timestamp = 0;
|
| - if (0 != rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
|
| - &rtp_timestamp)) {
|
| - // Waiting for RTCP.
|
| - return 0;
|
| - }
|
| - ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
|
| -
|
| - return 0;
|
| -}
|
| -
|
| -void ViEReceiver::StartReceive() {
|
| - CriticalSectionScoped cs(receive_cs_.get());
|
| - receiving_ = true;
|
| -}
|
| -
|
| -void ViEReceiver::StopReceive() {
|
| - CriticalSectionScoped cs(receive_cs_.get());
|
| - receiving_ = false;
|
| -}
|
| -
|
| -ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const {
|
| - return rtp_receive_statistics_.get();
|
| -}
|
| -
|
| -bool ViEReceiver::IsPacketInOrder(const RTPHeader& header) const {
|
| - StreamStatistician* statistician =
|
| - rtp_receive_statistics_->GetStatistician(header.ssrc);
|
| - if (!statistician)
|
| - return false;
|
| - return statistician->IsPacketInOrder(header.sequenceNumber);
|
| -}
|
| -
|
| -bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header,
|
| - bool in_order) const {
|
| - // Retransmissions are handled separately if RTX is enabled.
|
| - if (rtp_payload_registry_->RtxEnabled())
|
| - return false;
|
| - StreamStatistician* statistician =
|
| - rtp_receive_statistics_->GetStatistician(header.ssrc);
|
| - if (!statistician)
|
| - return false;
|
| - // Check if this is a retransmission.
|
| - int64_t min_rtt = 0;
|
| - rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
|
| - return !in_order &&
|
| - statistician->IsRetransmitOfOldPacket(header, min_rtt);
|
| -}
|
| -} // namespace webrtc
|
|
|