| 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 95d2f6fc8e3e3ec84486357b2e40a7a3402c0960..0000000000000000000000000000000000000000
|
| --- a/webrtc/video/vie_receiver.cc
|
| +++ /dev/null
|
| @@ -1,440 +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/config.h"
|
| -#include "webrtc/modules/pacing/packet_router.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/rtp_cvo.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
|
| -#include "webrtc/modules/video_coding/video_coding_impl.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 {
|
| -
|
| -std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
|
| - ReceiveStatistics* receive_statistics,
|
| - Transport* outgoing_transport,
|
| - RtcpRttStats* rtt_stats,
|
| - RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
|
| - RemoteBitrateEstimator* remote_bitrate_estimator,
|
| - RtpPacketSender* paced_sender,
|
| - TransportSequenceNumberAllocator* transport_sequence_number_allocator) {
|
| - RtpRtcp::Configuration configuration;
|
| - configuration.audio = false;
|
| - configuration.receiver_only = true;
|
| - configuration.receive_statistics = receive_statistics;
|
| - configuration.outgoing_transport = outgoing_transport;
|
| - configuration.intra_frame_callback = nullptr;
|
| - configuration.rtt_stats = rtt_stats;
|
| - configuration.rtcp_packet_type_counter_observer =
|
| - rtcp_packet_type_counter_observer;
|
| - configuration.paced_sender = paced_sender;
|
| - configuration.transport_sequence_number_allocator =
|
| - transport_sequence_number_allocator;
|
| - configuration.send_bitrate_observer = nullptr;
|
| - configuration.send_frame_count_observer = nullptr;
|
| - configuration.send_side_delay_observer = nullptr;
|
| - configuration.bandwidth_callback = nullptr;
|
| - configuration.transport_feedback_callback = nullptr;
|
| -
|
| - std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
|
| - rtp_rtcp->SetSendingStatus(false);
|
| - rtp_rtcp->SetSendingMediaStatus(false);
|
| - rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
|
| -
|
| - return rtp_rtcp;
|
| -}
|
| -
|
| -
|
| -static const int kPacketLogIntervalMs = 10000;
|
| -
|
| -ViEReceiver::ViEReceiver(vcm::VideoReceiver* video_receiver,
|
| - RemoteBitrateEstimator* remote_bitrate_estimator,
|
| - RtpFeedback* rtp_feedback,
|
| - Transport* transport,
|
| - RtcpRttStats* rtt_stats,
|
| - PacedSender* paced_sender,
|
| - PacketRouter* packet_router)
|
| - : clock_(Clock::GetRealTimeClock()),
|
| - video_receiver_(video_receiver),
|
| - remote_bitrate_estimator_(remote_bitrate_estimator),
|
| - packet_router_(packet_router),
|
| - ntp_estimator_(clock_),
|
| - rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
|
| - rtp_header_parser_(RtpHeaderParser::Create()),
|
| - rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
|
| - this,
|
| - rtp_feedback,
|
| - &rtp_payload_registry_)),
|
| - rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
|
| - fec_receiver_(FecReceiver::Create(this)),
|
| - receiving_(false),
|
| - restored_packet_in_use_(false),
|
| - last_packet_log_ms_(-1),
|
| - rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
|
| - transport,
|
| - rtt_stats,
|
| - &rtcp_packet_type_counter_observer_,
|
| - remote_bitrate_estimator_,
|
| - paced_sender,
|
| - packet_router)) {
|
| - packet_router_->AddRtpModule(rtp_rtcp_.get());
|
| - rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
|
| -}
|
| -
|
| -ViEReceiver::~ViEReceiver() {
|
| - packet_router_->RemoveRtpModule(rtp_rtcp_.get());
|
| - UpdateHistograms();
|
| -}
|
| -
|
| -void ViEReceiver::UpdateHistograms() {
|
| - FecPacketCounter counter = fec_receiver_->GetPacketCounter();
|
| - if (counter.num_packets > 0) {
|
| - RTC_LOGGED_HISTOGRAM_PERCENTAGE(
|
| - "WebRTC.Video.ReceivedFecPacketsInPercent",
|
| - static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
|
| - }
|
| - if (counter.num_fec_packets > 0) {
|
| - RTC_LOGGED_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 rtp_receiver_->RegisterReceivePayload(
|
| - video_codec.plName, video_codec.plType, kVideoPayloadTypeFrequency,
|
| - 0, 0) == 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);
|
| -}
|
| -
|
| -RtpReceiver* ViEReceiver::GetRtpReceiver() const {
|
| - return rtp_receiver_.get();
|
| -}
|
| -
|
| -void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension,
|
| - int id) {
|
| - RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
|
| - RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
|
| - StringToRtpExtensionType(extension), id));
|
| -}
|
| -
|
| -void ViEReceiver::RegisterRtcpPacketTypeCounterObserver(
|
| - RtcpPacketTypeCounterObserver* observer) {
|
| - rtcp_packet_type_counter_observer_.Set(observer);
|
| -}
|
| -
|
| -
|
| -int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
|
| - const size_t payload_size,
|
| - const WebRtcRTPHeader* rtp_header) {
|
| - RTC_DCHECK(video_receiver_);
|
| - WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
|
| - rtp_header_with_ntp.ntp_time_ms =
|
| - ntp_estimator_.Estimate(rtp_header->header.timestamp);
|
| - if (video_receiver_->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);
|
| -}
|
| -
|
| -bool ViEReceiver::DeliverRtp(const uint8_t* rtp_packet,
|
| - size_t rtp_packet_length,
|
| - const PacketTime& packet_time) {
|
| - RTC_DCHECK(remote_bitrate_estimator_);
|
| - {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - if (!receiving_) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - RTPHeader header;
|
| - if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
|
| - &header)) {
|
| - return false;
|
| - }
|
| - 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.
|
| - rtc::CritScope lock(&receive_cs_);
|
| - 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);
|
| - bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
|
| - // 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 video_receiver 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;
|
| - rtc::CritScope lock(&receive_cs_);
|
| - 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 ssrc: "
|
| - << header.ssrc << " payload type: "
|
| - << static_cast<int>(header.payloadType);
|
| - 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(nullptr, 0, &rtp_header);
|
| -}
|
| -
|
| -bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
|
| - size_t rtcp_packet_length) {
|
| - {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - if (!receiving_) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
|
| -
|
| - int64_t rtt = 0;
|
| - rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
|
| - if (rtt == 0) {
|
| - // Waiting for valid rtt.
|
| - return true;
|
| - }
|
| - uint32_t ntp_secs = 0;
|
| - uint32_t ntp_frac = 0;
|
| - uint32_t rtp_timestamp = 0;
|
| - if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
|
| - &rtp_timestamp) != 0) {
|
| - // Waiting for RTCP.
|
| - return true;
|
| - }
|
| - ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
|
| -
|
| - return true;
|
| -}
|
| -
|
| -void ViEReceiver::StartReceive() {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - receiving_ = true;
|
| -}
|
| -
|
| -void ViEReceiver::StopReceive() {
|
| - rtc::CritScope lock(&receive_cs_);
|
| - 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(), nullptr, nullptr, &min_rtt, nullptr);
|
| - return !in_order &&
|
| - statistician->IsRetransmitOfOldPacket(header, min_rtt);
|
| -}
|
| -} // namespace webrtc
|
|
|