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Unified Diff: webrtc/video/vie_receiver.cc

Issue 1917363005: Rename ViEReceiver and move ownership to VideoReceiveStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Changed comment. Created 4 years, 8 months ago
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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

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