blob: a47055abf08465a2ed0e03886c2770e636a62af0 [file] [log] [blame]
/*
* 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/rtp_video_stream_receiver.h"
#include <algorithm>
#include <utility>
#include <vector>
#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
#include "webrtc/media/base/mediaconstants.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/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/rtp_rtcp/include/ulpfec_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/modules/video_coding/frame_object.h"
#include "webrtc/modules/video_coding/h264_sprop_parameter_sets.h"
#include "webrtc/modules/video_coding/h264_sps_pps_tracker.h"
#include "webrtc/modules/video_coding/packet_buffer.h"
#include "webrtc/modules/video_coding/video_coding_impl.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/location.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
#include "webrtc/video/receive_statistics_proxy.h"
namespace webrtc {
namespace {
// TODO(philipel): Change kPacketBufferStartSize back to 32 in M63 see:
// crbug.com/752886
constexpr int kPacketBufferStartSize = 512;
constexpr int kPacketBufferMaxSixe = 2048;
}
std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
ReceiveStatistics* receive_statistics,
Transport* outgoing_transport,
RtcpRttStats* rtt_stats,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
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.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.send_packet_observer = nullptr;
configuration.bandwidth_callback = nullptr;
configuration.transport_feedback_callback = nullptr;
std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
return rtp_rtcp;
}
static const int kPacketLogIntervalMs = 10000;
RtpVideoStreamReceiver::RtpVideoStreamReceiver(
Transport* transport,
RtcpRttStats* rtt_stats,
PacketRouter* packet_router,
const VideoReceiveStream::Config* config,
ReceiveStatisticsProxy* receive_stats_proxy,
ProcessThread* process_thread,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender,
video_coding::OnCompleteFrameCallback* complete_frame_callback,
VCMTiming* timing)
: clock_(Clock::GetRealTimeClock()),
config_(*config),
packet_router_(packet_router),
process_thread_(process_thread),
ntp_estimator_(clock_),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
this,
this,
&rtp_payload_registry_)),
rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, this)),
receiving_(false),
restored_packet_in_use_(false),
last_packet_log_ms_(-1),
rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
transport,
rtt_stats,
receive_stats_proxy,
packet_router)),
complete_frame_callback_(complete_frame_callback),
keyframe_request_sender_(keyframe_request_sender),
timing_(timing),
has_received_frame_(false) {
constexpr bool remb_candidate = true;
packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy);
rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
<< "A stream should not be configured with RTCP disabled. This value is "
"reserved for internal usage.";
RTC_DCHECK(config_.rtp.remote_ssrc != 0);
// TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
RTC_DCHECK(config_.rtp.local_ssrc != 0);
RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc);
rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc);
rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
EnableReceiveRtpHeaderExtension(config_.rtp.extensions[i].uri,
config_.rtp.extensions[i].id);
}
static const int kMaxPacketAgeToNack = 450;
const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
? kMaxPacketAgeToNack
: kDefaultMaxReorderingThreshold;
rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
if (config_.rtp.rtx_ssrc) {
rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc);
for (const auto& kv : config_.rtp.rtx_associated_payload_types) {
RTC_DCHECK_NE(kv.first, 0);
rtp_payload_registry_.SetRtxPayloadType(kv.first, kv.second);
}
}
if (IsUlpfecEnabled()) {
VideoCodec ulpfec_codec = {};
ulpfec_codec.codecType = kVideoCodecULPFEC;
strncpy(ulpfec_codec.plName, "ulpfec", sizeof(ulpfec_codec.plName));
ulpfec_codec.plType = config_.rtp.ulpfec.ulpfec_payload_type;
RTC_CHECK(AddReceiveCodec(ulpfec_codec));
}
if (IsRedEnabled()) {
VideoCodec red_codec = {};
red_codec.codecType = kVideoCodecRED;
strncpy(red_codec.plName, "red", sizeof(red_codec.plName));
red_codec.plType = config_.rtp.ulpfec.red_payload_type;
RTC_CHECK(AddReceiveCodec(red_codec));
if (config_.rtp.ulpfec.red_rtx_payload_type != -1) {
rtp_payload_registry_.SetRtxPayloadType(
config_.rtp.ulpfec.red_rtx_payload_type,
config_.rtp.ulpfec.red_payload_type);
}
}
if (config_.rtp.rtcp_xr.receiver_reference_time_report)
rtp_rtcp_->SetRtcpXrRrtrStatus(true);
// Stats callback for CNAME changes.
rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
if (config_.rtp.nack.rtp_history_ms != 0) {
nack_module_.reset(
new NackModule(clock_, nack_sender, keyframe_request_sender));
process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE);
}
packet_buffer_ = video_coding::PacketBuffer::Create(
clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this);
reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this));
}
RtpVideoStreamReceiver::~RtpVideoStreamReceiver() {
RTC_DCHECK(secondary_sinks_.empty());
if (nack_module_) {
process_thread_->DeRegisterModule(nack_module_.get());
}
process_thread_->DeRegisterModule(rtp_rtcp_.get());
packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
UpdateHistograms();
}
bool RtpVideoStreamReceiver::AddReceiveCodec(
const VideoCodec& video_codec,
const std::map<std::string, std::string>& codec_params) {
pt_codec_params_.insert(make_pair(video_codec.plType, codec_params));
return AddReceiveCodec(video_codec);
}
bool RtpVideoStreamReceiver::AddReceiveCodec(const VideoCodec& video_codec) {
int8_t old_pltype = -1;
if (rtp_payload_registry_.ReceivePayloadType(video_codec, &old_pltype) !=
-1) {
rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
}
return rtp_payload_registry_.RegisterReceivePayload(video_codec) == 0;
}
uint32_t RtpVideoStreamReceiver::GetRemoteSsrc() const {
return config_.rtp.remote_ssrc;
}
int RtpVideoStreamReceiver::GetCsrcs(uint32_t* csrcs) const {
return rtp_receiver_->CSRCs(csrcs);
}
RtpReceiver* RtpVideoStreamReceiver::GetRtpReceiver() const {
return rtp_receiver_.get();
}
int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
const uint8_t* payload_data,
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);
VCMPacket packet(payload_data, payload_size, rtp_header_with_ntp);
packet.timesNacked =
nack_module_ ? nack_module_->OnReceivedPacket(packet) : -1;
packet.receive_time_ms = clock_->TimeInMilliseconds();
// In the case of a video stream without picture ids and no rtx the
// RtpFrameReferenceFinder will need to know about padding to
// correctly calculate frame references.
if (packet.sizeBytes == 0) {
reference_finder_->PaddingReceived(packet.seqNum);
packet_buffer_->PaddingReceived(packet.seqNum);
return 0;
}
if (packet.codec == kVideoCodecH264) {
// Only when we start to receive packets will we know what payload type
// that will be used. When we know the payload type insert the correct
// sps/pps into the tracker.
if (packet.payloadType != last_payload_type_) {
last_payload_type_ = packet.payloadType;
InsertSpsPpsIntoTracker(packet.payloadType);
}
switch (tracker_.CopyAndFixBitstream(&packet)) {
case video_coding::H264SpsPpsTracker::kRequestKeyframe:
keyframe_request_sender_->RequestKeyFrame();
FALLTHROUGH();
case video_coding::H264SpsPpsTracker::kDrop:
return 0;
case video_coding::H264SpsPpsTracker::kInsert:
break;
}
} else {
uint8_t* data = new uint8_t[packet.sizeBytes];
memcpy(data, packet.dataPtr, packet.sizeBytes);
packet.dataPtr = data;
}
packet_buffer_->InsertPacket(&packet);
return 0;
}
// TODO(nisse): Try to delete this method. Obstacles: It is used by
// ParseAndHandleEncapsulatingHeader, for handling Rtx packets, and
// for callbacks from |ulpfec_receiver_|.
void RtpVideoStreamReceiver::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;
}
header.payload_type_frequency = kVideoPayloadTypeFrequency;
bool in_order = IsPacketInOrder(header);
ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
}
// TODO(pbos): Remove as soon as audio can handle a changing payload type
// without this callback.
int32_t RtpVideoStreamReceiver::OnInitializeDecoder(
const int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const size_t channels,
const uint32_t rate) {
RTC_NOTREACHED();
return 0;
}
// This method handles both regular RTP packets and packets recovered
// via FlexFEC.
void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
if (!receiving_) {
return;
}
if (!packet.recovered()) {
int64_t now_ms = clock_->TimeInMilliseconds();
// Periodically log the RTP header of incoming packets.
if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
std::stringstream ss;
ss << "Packet received on SSRC: " << packet.Ssrc()
<< " with payload type: " << static_cast<int>(packet.PayloadType())
<< ", timestamp: " << packet.Timestamp()
<< ", sequence number: " << packet.SequenceNumber()
<< ", arrival time: " << packet.arrival_time_ms();
int32_t time_offset;
if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
ss << ", toffset: " << time_offset;
}
uint32_t send_time;
if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
ss << ", abs send time: " << send_time;
}
LOG(LS_INFO) << ss.str();
last_packet_log_ms_ = now_ms;
}
}
// TODO(nisse): Delete use of GetHeader, but needs refactoring of
// ReceivePacket and IncomingPacket methods below.
RTPHeader header;
packet.GetHeader(&header);
header.payload_type_frequency = kVideoPayloadTypeFrequency;
bool in_order = IsPacketInOrder(header);
if (!packet.recovered()) {
// TODO(nisse): Why isn't this done for recovered packets?
rtp_payload_registry_.SetIncomingPayloadType(header);
}
ReceivePacket(packet.data(), packet.size(), 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).
if (!packet.recovered()) {
// TODO(nisse): We should pass a recovered flag to stats, to aid
// fixing bug bugs.webrtc.org/6339.
rtp_receive_statistics_->IncomingPacket(
header, packet.size(), IsPacketRetransmitted(header, in_order));
}
for (RtpPacketSinkInterface* secondary_sink : secondary_sinks_) {
secondary_sink->OnRtpPacket(packet);
}
}
int32_t RtpVideoStreamReceiver::RequestKeyFrame() {
return rtp_rtcp_->RequestKeyFrame();
}
bool RtpVideoStreamReceiver::IsUlpfecEnabled() const {
return config_.rtp.ulpfec.ulpfec_payload_type != -1;
}
bool RtpVideoStreamReceiver::IsRedEnabled() const {
return config_.rtp.ulpfec.red_payload_type != -1;
}
bool RtpVideoStreamReceiver::IsRetransmissionsEnabled() const {
return config_.rtp.nack.rtp_history_ms > 0;
}
void RtpVideoStreamReceiver::RequestPacketRetransmit(
const std::vector<uint16_t>& sequence_numbers) {
rtp_rtcp_->SendNack(sequence_numbers);
}
int32_t RtpVideoStreamReceiver::ResendPackets(const uint16_t* sequence_numbers,
uint16_t length) {
return rtp_rtcp_->SendNACK(sequence_numbers, length);
}
void RtpVideoStreamReceiver::OnReceivedFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) {
if (!has_received_frame_) {
has_received_frame_ = true;
if (frame->FrameType() != kVideoFrameKey)
keyframe_request_sender_->RequestKeyFrame();
}
if (!frame->delayed_by_retransmission())
timing_->IncomingTimestamp(frame->timestamp, clock_->TimeInMilliseconds());
reference_finder_->ManageFrame(std::move(frame));
}
void RtpVideoStreamReceiver::OnCompleteFrame(
std::unique_ptr<video_coding::FrameObject> frame) {
{
rtc::CritScope lock(&last_seq_num_cs_);
video_coding::RtpFrameObject* rtp_frame =
static_cast<video_coding::RtpFrameObject*>(frame.get());
last_seq_num_for_pic_id_[rtp_frame->picture_id] = rtp_frame->last_seq_num();
}
complete_frame_callback_->OnCompleteFrame(std::move(frame));
}
void RtpVideoStreamReceiver::OnRttUpdate(int64_t avg_rtt_ms,
int64_t max_rtt_ms) {
if (nack_module_)
nack_module_->UpdateRtt(max_rtt_ms);
}
rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedPacketMs() const {
return packet_buffer_->LastReceivedPacketMs();
}
rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedKeyframePacketMs()
const {
return packet_buffer_->LastReceivedKeyframePacketMs();
}
void RtpVideoStreamReceiver::AddSecondarySink(RtpPacketSinkInterface* sink) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
RTC_DCHECK(std::find(secondary_sinks_.cbegin(), secondary_sinks_.cend(),
sink) == secondary_sinks_.cend());
secondary_sinks_.push_back(sink);
}
void RtpVideoStreamReceiver::RemoveSecondarySink(
const RtpPacketSinkInterface* sink) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
auto it = std::find(secondary_sinks_.begin(), secondary_sinks_.end(), sink);
if (it == secondary_sinks_.end()) {
// We might be rolling-back a call whose setup failed mid-way. In such a
// case, it's simpler to remove "everything" rather than remember what
// has already been added.
LOG(LS_WARNING) << "Removal of unknown sink.";
return;
}
secondary_sinks_.erase(it);
}
void RtpVideoStreamReceiver::ReceivePacket(const uint8_t* packet,
size_t packet_length,
const RTPHeader& header,
bool in_order) {
if (rtp_payload_registry_.IsEncapsulated(header)) {
ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
return;
}
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;
}
rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
payload_specific, in_order);
}
void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
const uint8_t* packet, size_t packet_length, const RTPHeader& header) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
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 (ulpfec_receiver_->AddReceivedRedPacket(header, packet, packet_length,
ulpfec_pt) != 0) {
return;
}
ulpfec_receiver_->ProcessReceivedFec();
} 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;
}
// Remove the RTX header and parse the original RTP header.
if (packet_length < header.headerLength)
return;
if (packet_length > sizeof(restored_packet_))
return;
if (restored_packet_in_use_) {
LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
return;
}
if (!rtp_payload_registry_.RestoreOriginalPacket(
restored_packet_, packet, &packet_length, config_.rtp.remote_ssrc,
header)) {
LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
<< header.ssrc << " payload type: "
<< static_cast<int>(header.payloadType);
return;
}
restored_packet_in_use_ = true;
OnRecoveredPacket(restored_packet_, packet_length);
restored_packet_in_use_ = false;
}
}
void RtpVideoStreamReceiver::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 = header.extension.videoRotation;
}
rtp_header.type.Video.content_type = VideoContentType::UNSPECIFIED;
if (header.extension.hasVideoContentType) {
rtp_header.type.Video.content_type = header.extension.videoContentType;
}
rtp_header.type.Video.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
if (header.extension.has_video_timing) {
rtp_header.type.Video.video_timing = header.extension.video_timing;
}
rtp_header.type.Video.playout_delay = header.extension.playout_delay;
OnReceivedPayloadData(nullptr, 0, &rtp_header);
}
bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
size_t rtcp_packet_length) {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
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 RtpVideoStreamReceiver::FrameContinuous(int64_t picture_id) {
if (!nack_module_)
return;
int seq_num = -1;
{
rtc::CritScope lock(&last_seq_num_cs_);
auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
if (seq_num_it != last_seq_num_for_pic_id_.end())
seq_num = seq_num_it->second;
}
if (seq_num != -1)
nack_module_->ClearUpTo(seq_num);
}
void RtpVideoStreamReceiver::FrameDecoded(int64_t picture_id) {
int seq_num = -1;
{
rtc::CritScope lock(&last_seq_num_cs_);
auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
if (seq_num_it != last_seq_num_for_pic_id_.end()) {
seq_num = seq_num_it->second;
last_seq_num_for_pic_id_.erase(last_seq_num_for_pic_id_.begin(),
++seq_num_it);
}
}
if (seq_num != -1) {
packet_buffer_->ClearTo(seq_num);
reference_finder_->ClearTo(seq_num);
}
}
void RtpVideoStreamReceiver::SignalNetworkState(NetworkState state) {
rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
: RtcpMode::kOff);
}
void RtpVideoStreamReceiver::StartReceive() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
receiving_ = true;
}
void RtpVideoStreamReceiver::StopReceive() {
RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
receiving_ = false;
}
bool RtpVideoStreamReceiver::IsPacketInOrder(const RTPHeader& header) const {
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(header.ssrc);
if (!statistician)
return false;
return statistician->IsPacketInOrder(header.sequenceNumber);
}
bool RtpVideoStreamReceiver::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(config_.rtp.remote_ssrc, nullptr, nullptr, &min_rtt, nullptr);
return !in_order &&
statistician->IsRetransmitOfOldPacket(header, min_rtt);
}
void RtpVideoStreamReceiver::UpdateHistograms() {
FecPacketCounter counter = ulpfec_receiver_->GetPacketCounter();
if (counter.first_packet_time_ms == -1)
return;
int64_t elapsed_sec =
(clock_->TimeInMilliseconds() - counter.first_packet_time_ms) / 1000;
if (elapsed_sec < metrics::kMinRunTimeInSeconds)
return;
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));
}
}
void RtpVideoStreamReceiver::EnableReceiveRtpHeaderExtension(
const std::string& extension, int id) {
// One-byte-extension local identifiers are in the range 1-14 inclusive.
RTC_DCHECK_GE(id, 1);
RTC_DCHECK_LE(id, 14);
RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
StringToRtpExtensionType(extension), id));
}
void RtpVideoStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) {
auto codec_params_it = pt_codec_params_.find(payload_type);
if (codec_params_it == pt_codec_params_.end())
return;
LOG(LS_INFO) << "Found out of band supplied codec parameters for"
<< " payload type: " << static_cast<int>(payload_type);
H264SpropParameterSets sprop_decoder;
auto sprop_base64_it =
codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets);
if (sprop_base64_it == codec_params_it->second.end())
return;
if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
return;
tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
sprop_decoder.pps_nalu());
}
} // namespace webrtc