blob: cdff8607c18463bc274b0707916a1af7790521ac [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_engine/vie_receiver.h"
#include <vector>
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.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/logging.h"
#include "webrtc/system_wrappers/interface/metrics.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
static const int kPacketLogIntervalMs = 10000;
ViEReceiver::ViEReceiver(const int32_t channel_id,
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(channel_id,
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_)),
rtp_dump_(NULL),
receiving_(false),
restored_packet_in_use_(false),
receiving_ast_enabled_(false),
last_packet_log_ms_(-1) {
assert(remote_bitrate_estimator);
}
ViEReceiver::~ViEReceiver() {
UpdateHistograms();
if (rtp_dump_) {
rtp_dump_->Stop();
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
}
}
void ViEReceiver::UpdateHistograms() {
FecPacketCounter counter = fec_receiver_->GetPacketCounter();
if (counter.num_packets > 0) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedFecPacketsInPercent",
counter.num_fec_packets * 100 / counter.num_packets);
}
if (counter.num_fec_packets > 0) {
RTC_HISTOGRAM_PERCENTAGE(
"WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
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) {
rtp_payload_registry_->SetRtxPayloadType(payload_type);
}
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::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) {
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);
}
}
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);
}
void ViEReceiver::ReceivedBWEPacket(
int64_t arrival_time_ms, size_t payload_size, const RTPHeader& header) {
// Only forward if the incoming packet *and* the channel are both configured
// to receive absolute sender time. RTP time stamps may have different rates
// for audio and video and shouldn't be mixed.
if (header.extension.hasAbsoluteSendTime && receiving_ast_enabled_) {
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
header);
}
}
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;
}
if (rtp_dump_) {
rtp_dump_->DumpPacket(rtp_packet, rtp_packet_length);
}
}
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);
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);
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;
}
uint8_t* restored_packet_ptr = restored_packet_;
if (!rtp_payload_registry_->RestoreOriginalPacket(
&restored_packet_ptr, 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_ptr, packet_length);
restored_packet_in_use_ = false;
return ret;
}
return false;
}
int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet,
size_t rtcp_packet_length) {
{
CriticalSectionScoped cs(receive_cs_.get());
if (!receiving_) {
return -1;
}
if (rtp_dump_) {
rtp_dump_->DumpPacket(rtcp_packet, rtcp_packet_length);
}
std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin();
while (it != rtp_rtcp_simulcast_.end()) {
RtpRtcp* rtp_rtcp = *it++;
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;
}
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) {
return -1;
}
}
if (rtp_dump_->Start(file_nameUTF8) != 0) {
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
return -1;
}
return 0;
}
int ViEReceiver::StopRTPDump() {
CriticalSectionScoped cs(receive_cs_.get());
if (rtp_dump_) {
if (rtp_dump_->IsActive()) {
rtp_dump_->Stop();
}
RtpDump::DestroyRtpDump(rtp_dump_);
rtp_dump_ = NULL;
} else {
return -1;
}
return 0;
}
void ViEReceiver::GetReceiveBandwidthEstimatorStats(
ReceiveBandwidthEstimatorStats* output) const {
remote_bitrate_estimator_->GetStats(output);
}
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