Google Git
Sign in
webrtc / src / a0b52b5708534943aa66bef724c438244cc84eac / . / video / rtp_video_stream_receiver.cc
blob: 696aa2c7a20a2f03f2567372a2e78cdf0251bc01 [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 "video/rtp_video_stream_receiver.h"
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "media/base/media_constants.h"
#include "modules/pacing/packet_router.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
#include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/video_coding/frame_object.h"
#include "modules/video_coding/h264_sprop_parameter_sets.h"
#include "modules/video_coding/h264_sps_pps_tracker.h"
#include "modules/video_coding/nack_module.h"
#include "modules/video_coding/packet_buffer.h"
#include "modules/video_coding/video_coding_impl.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/fallthrough.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "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 kPacketBufferMaxSize = 2048;
} // namespace
std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
Clock* clock,
ReceiveStatistics* receive_statistics,
Transport* outgoing_transport,
RtcpRttStats* rtt_stats,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
uint32_t local_ssrc) {
RtpRtcp::Configuration configuration;
configuration.clock = clock;
configuration.audio = false;
configuration.receiver_only = true;
configuration.receive_statistics = receive_statistics;
configuration.outgoing_transport = outgoing_transport;
configuration.rtt_stats = rtt_stats;
configuration.rtcp_packet_type_counter_observer =
rtcp_packet_type_counter_observer;
configuration.media_send_ssrc = local_ssrc;
std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
return rtp_rtcp;
}
static const int kPacketLogIntervalMs = 10000;
RtpVideoStreamReceiver::RtcpFeedbackBuffer::RtcpFeedbackBuffer(
KeyFrameRequestSender* key_frame_request_sender,
NackSender* nack_sender,
LossNotificationSender* loss_notification_sender)
: key_frame_request_sender_(key_frame_request_sender),
nack_sender_(nack_sender),
loss_notification_sender_(loss_notification_sender),
request_key_frame_(false) {
RTC_DCHECK(key_frame_request_sender_);
RTC_DCHECK(nack_sender_);
RTC_DCHECK(loss_notification_sender_);
}
void RtpVideoStreamReceiver::RtcpFeedbackBuffer::RequestKeyFrame() {
rtc::CritScope lock(&cs_);
request_key_frame_ = true;
}
void RtpVideoStreamReceiver::RtcpFeedbackBuffer::SendNack(
const std::vector<uint16_t>& sequence_numbers,
bool buffering_allowed) {
RTC_DCHECK(!sequence_numbers.empty());
rtc::CritScope lock(&cs_);
nack_sequence_numbers_.insert(nack_sequence_numbers_.end(),
sequence_numbers.cbegin(),
sequence_numbers.cend());
if (!buffering_allowed) {
// Note that while *buffering* is not allowed, *batching* is, meaning that
// previously buffered messages may be sent along with the current message.
SendBufferedRtcpFeedback();
}
}
void RtpVideoStreamReceiver::RtcpFeedbackBuffer::SendLossNotification(
uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed) {
RTC_DCHECK(buffering_allowed);
rtc::CritScope lock(&cs_);
RTC_DCHECK(!lntf_state_)
<< "SendLossNotification() called twice in a row with no call to "
"SendBufferedRtcpFeedback() in between.";
lntf_state_ = absl::make_optional<LossNotificationState>(
last_decoded_seq_num, last_received_seq_num, decodability_flag);
}
void RtpVideoStreamReceiver::RtcpFeedbackBuffer::SendBufferedRtcpFeedback() {
bool request_key_frame = false;
std::vector<uint16_t> nack_sequence_numbers;
absl::optional<LossNotificationState> lntf_state;
{
rtc::CritScope lock(&cs_);
std::swap(request_key_frame, request_key_frame_);
std::swap(nack_sequence_numbers, nack_sequence_numbers_);
std::swap(lntf_state, lntf_state_);
}
if (lntf_state) {
// If either a NACK or a key frame request is sent, we should buffer
// the LNTF and wait for them (NACK or key frame request) to trigger
// the compound feedback message.
// Otherwise, the LNTF should be sent out immediately.
const bool buffering_allowed =
request_key_frame || !nack_sequence_numbers.empty();
loss_notification_sender_->SendLossNotification(
lntf_state->last_decoded_seq_num, lntf_state->last_received_seq_num,
lntf_state->decodability_flag, buffering_allowed);
}
if (request_key_frame) {
key_frame_request_sender_->RequestKeyFrame();
} else if (!nack_sequence_numbers.empty()) {
nack_sender_->SendNack(nack_sequence_numbers, true);
}
}
RtpVideoStreamReceiver::RtpVideoStreamReceiver(
Clock* clock,
Transport* transport,
RtcpRttStats* rtt_stats,
PacketRouter* packet_router,
const VideoReceiveStream::Config* config,
ReceiveStatistics* rtp_receive_statistics,
ReceiveStatisticsProxy* receive_stats_proxy,
ProcessThread* process_thread,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender,
video_coding::OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor)
: clock_(clock),
config_(*config),
packet_router_(packet_router),
process_thread_(process_thread),
ntp_estimator_(clock),
rtp_header_extensions_(config_.rtp.extensions),
rtp_receive_statistics_(rtp_receive_statistics),
ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc,
this,
config->rtp.extensions)),
receiving_(false),
last_packet_log_ms_(-1),
rtp_rtcp_(CreateRtpRtcpModule(clock,
rtp_receive_statistics_,
transport,
rtt_stats,
receive_stats_proxy,
config_.rtp.local_ssrc)),
complete_frame_callback_(complete_frame_callback),
keyframe_request_sender_(keyframe_request_sender),
// TODO(bugs.webrtc.org/10336): Let |rtcp_feedback_buffer_| communicate
// directly with |rtp_rtcp_|.
rtcp_feedback_buffer_(this, nack_sender, this),
has_received_frame_(false),
frames_decryptable_(false) {
constexpr bool remb_candidate = true;
if (packet_router_)
packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
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_->SetRemoteSSRC(config_.rtp.remote_ssrc);
static const int kMaxPacketAgeToNack = 450;
const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
? kMaxPacketAgeToNack
: kDefaultMaxReorderingThreshold;
rtp_receive_statistics_->SetMaxReorderingThreshold(config_.rtp.remote_ssrc,
max_reordering_threshold);
// TODO(nisse): For historic reasons, we applied the above
// max_reordering_threshold also for RTX stats, which makes little sense since
// we don't NACK rtx packets. Consider deleting the below block, and rely on
// the default threshold.
if (config_.rtp.rtx_ssrc) {
rtp_receive_statistics_->SetMaxReorderingThreshold(
config_.rtp.rtx_ssrc, max_reordering_threshold);
}
if (config_.rtp.rtcp_xr.receiver_reference_time_report)
rtp_rtcp_->SetRtcpXrRrtrStatus(true);
// Stats callback for CNAME changes.
rtp_rtcp_->RegisterRtcpCnameCallback(receive_stats_proxy);
process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
if (config_.rtp.lntf.enabled) {
loss_notification_controller_ =
absl::make_unique<LossNotificationController>(&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_);
}
if (config_.rtp.nack.rtp_history_ms != 0) {
nack_module_ = absl::make_unique<NackModule>(clock_, &rtcp_feedback_buffer_,
&rtcp_feedback_buffer_);
process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE);
}
// The group here must be a positive power of 2, in which case that is used as
// size. All other values shall result in the default value being used.
const std::string group_name =
webrtc::field_trial::FindFullName("WebRTC-PacketBufferMaxSize");
int packet_buffer_max_size = kPacketBufferMaxSize;
if (!group_name.empty() &&
(sscanf(group_name.c_str(), "%d", &packet_buffer_max_size) != 1 ||
packet_buffer_max_size <= 0 ||
// Verify that the number is a positive power of 2.
(packet_buffer_max_size & (packet_buffer_max_size - 1)) != 0)) {
RTC_LOG(LS_WARNING) << "Invalid packet buffer max size: " << group_name;
packet_buffer_max_size = kPacketBufferMaxSize;
}
packet_buffer_ = video_coding::PacketBuffer::Create(
clock_, kPacketBufferStartSize, packet_buffer_max_size, this);
reference_finder_ =
absl::make_unique<video_coding::RtpFrameReferenceFinder>(this);
// Only construct the encrypted receiver if frame encryption is enabled.
if (config_.crypto_options.sframe.require_frame_encryption) {
buffered_frame_decryptor_ =
absl::make_unique<BufferedFrameDecryptor>(this, this);
if (frame_decryptor != nullptr) {
buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
}
}
}
RtpVideoStreamReceiver::~RtpVideoStreamReceiver() {
RTC_DCHECK(secondary_sinks_.empty());
if (nack_module_) {
process_thread_->DeRegisterModule(nack_module_.get());
}
process_thread_->DeRegisterModule(rtp_rtcp_.get());
if (packet_router_)
packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
UpdateHistograms();
}
void RtpVideoStreamReceiver::AddReceiveCodec(
const VideoCodec& video_codec,
const std::map<std::string, std::string>& codec_params,
bool raw_payload) {
absl::optional<VideoCodecType> video_type;
if (!raw_payload) {
video_type = video_codec.codecType;
}
payload_type_map_.emplace(video_codec.plType, video_type);
pt_codec_params_.emplace(video_codec.plType, codec_params);
}
absl::optional<Syncable::Info> RtpVideoStreamReceiver::GetSyncInfo() const {
Syncable::Info info;
if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs,
&info.capture_time_ntp_frac, nullptr, nullptr,
&info.capture_time_source_clock) != 0) {
return absl::nullopt;
}
{
rtc::CritScope lock(&sync_info_lock_);
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
return absl::nullopt;
}
info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
}
// Leaves info.current_delay_ms uninitialized.
return info;
}
int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
const uint8_t* payload_data,
size_t payload_size,
const RTPHeader& rtp_header,
const RTPVideoHeader& video_header,
const absl::optional<RtpGenericFrameDescriptor>& generic_descriptor,
bool is_recovered) {
VCMPacket packet(payload_data, payload_size, rtp_header, video_header,
ntp_estimator_.Estimate(rtp_header.timestamp),
clock_->TimeInMilliseconds());
packet.generic_descriptor = generic_descriptor;
if (loss_notification_controller_) {
if (is_recovered) {
// TODO(bugs.webrtc.org/10336): Implement support for reordering.
RTC_LOG(LS_WARNING)
<< "LossNotificationController does not support reordering.";
} else {
loss_notification_controller_->OnReceivedPacket(packet);
}
}
if (nack_module_) {
const bool is_keyframe =
video_header.is_first_packet_in_frame &&
video_header.frame_type == VideoFrameType::kVideoFrameKey;
packet.timesNacked = nack_module_->OnReceivedPacket(
rtp_header.sequenceNumber, is_keyframe, is_recovered);
} else {
packet.timesNacked = -1;
}
if (packet.sizeBytes == 0) {
NotifyReceiverOfEmptyPacket(packet.seqNum);
rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
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:
rtcp_feedback_buffer_.RequestKeyFrame();
rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
RTC_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;
}
rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
if (!packet_buffer_->InsertPacket(&packet)) {
RequestKeyFrame();
}
return 0;
}
void RtpVideoStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
size_t rtp_packet_length) {
RtpPacketReceived packet;
if (!packet.Parse(rtp_packet, rtp_packet_length))
return;
if (packet.PayloadType() == config_.rtp.red_payload_type) {
RTC_LOG(LS_WARNING) << "Discarding recovered packet with RED encapsulation";
return;
}
packet.IdentifyExtensions(rtp_header_extensions_);
packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
// TODO(nisse): UlpfecReceiverImpl::ProcessReceivedFec passes both
// original (decapsulated) media packets and recovered packets to
// this callback. We need a way to distinguish, for setting
// packet.recovered() correctly. Ideally, move RED decapsulation out
// of the Ulpfec implementation.
ReceivePacket(packet);
}
// This method handles both regular RTP packets and packets recovered
// via FlexFEC.
void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
if (!receiving_) {
return;
}
if (!packet.recovered()) {
// TODO(nisse): Exclude out-of-order packets?
int64_t now_ms = clock_->TimeInMilliseconds();
{
rtc::CritScope cs(&sync_info_lock_);
last_received_rtp_timestamp_ = packet.Timestamp();
last_received_rtp_system_time_ms_ = now_ms;
}
// Periodically log the RTP header of incoming packets.
if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
rtc::StringBuilder 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;
}
RTC_LOG(LS_INFO) << ss.str();
last_packet_log_ms_ = now_ms;
}
}
ReceivePacket(packet);
// 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()) {
rtp_receive_statistics_->OnRtpPacket(packet);
}
for (RtpPacketSinkInterface* secondary_sink : secondary_sinks_) {
secondary_sink->OnRtpPacket(packet);
}
}
void RtpVideoStreamReceiver::RequestKeyFrame() {
// TODO(bugs.webrtc.org/10336): Allow the sender to ignore key frame requests
// issued by anything other than the LossNotificationController if it (the
// sender) is relying on LNTF alone.
if (keyframe_request_sender_) {
keyframe_request_sender_->RequestKeyFrame();
} else {
rtp_rtcp_->SendPictureLossIndication();
}
}
void RtpVideoStreamReceiver::SendLossNotification(
uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed) {
RTC_DCHECK(config_.rtp.lntf.enabled);
rtp_rtcp_->SendLossNotification(last_decoded_seq_num, last_received_seq_num,
decodability_flag, buffering_allowed);
}
bool RtpVideoStreamReceiver::IsUlpfecEnabled() const {
return config_.rtp.ulpfec_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);
}
bool RtpVideoStreamReceiver::IsDecryptable() const {
return frames_decryptable_.load();
}
void RtpVideoStreamReceiver::OnAssembledFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&network_tc_);
RTC_DCHECK(frame);
absl::optional<RtpGenericFrameDescriptor> descriptor =
frame->GetGenericFrameDescriptor();
if (loss_notification_controller_ && descriptor) {
loss_notification_controller_->OnAssembledFrame(
frame->first_seq_num(), descriptor->FrameId(),
descriptor->Discardable().value_or(false),
descriptor->FrameDependenciesDiffs());
}
// If frames arrive before a key frame, they would not be decodable.
// In that case, request a key frame ASAP.
if (!has_received_frame_) {
if (frame->FrameType() != VideoFrameType::kVideoFrameKey) {
// |loss_notification_controller_|, if present, would have already
// requested a key frame when the first packet for the non-key frame
// had arrived, so no need to replicate the request.
if (!loss_notification_controller_) {
RequestKeyFrame();
}
}
has_received_frame_ = true;
}
if (buffered_frame_decryptor_ == nullptr) {
reference_finder_->ManageFrame(std::move(frame));
} else {
buffered_frame_decryptor_->ManageEncryptedFrame(std::move(frame));
}
}
void RtpVideoStreamReceiver::OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> 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->id.picture_id] =
rtp_frame->last_seq_num();
}
complete_frame_callback_->OnCompleteFrame(std::move(frame));
}
void RtpVideoStreamReceiver::OnDecryptedFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) {
reference_finder_->ManageFrame(std::move(frame));
}
void RtpVideoStreamReceiver::OnDecryptionStatusChange(
FrameDecryptorInterface::Status status) {
frames_decryptable_.store(
(status == FrameDecryptorInterface::Status::kOk) ||
(status == FrameDecryptorInterface::Status::kRecoverable));
}
void RtpVideoStreamReceiver::SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
RTC_DCHECK_RUN_ON(&network_tc_);
if (buffered_frame_decryptor_ == nullptr) {
buffered_frame_decryptor_ =
absl::make_unique<BufferedFrameDecryptor>(this, this);
}
buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
}
void RtpVideoStreamReceiver::UpdateRtt(int64_t max_rtt_ms) {
if (nack_module_)
nack_module_->UpdateRtt(max_rtt_ms);
}
absl::optional<int64_t> RtpVideoStreamReceiver::LastReceivedPacketMs() const {
return packet_buffer_->LastReceivedPacketMs();
}
absl::optional<int64_t> RtpVideoStreamReceiver::LastReceivedKeyframePacketMs()
const {
return packet_buffer_->LastReceivedKeyframePacketMs();
}
void RtpVideoStreamReceiver::AddSecondarySink(RtpPacketSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
RTC_DCHECK(!absl::c_linear_search(secondary_sinks_, sink));
secondary_sinks_.push_back(sink);
}
void RtpVideoStreamReceiver::RemoveSecondarySink(
const RtpPacketSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
auto it = absl::c_find(secondary_sinks_, 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.
RTC_LOG(LS_WARNING) << "Removal of unknown sink.";
return;
}
secondary_sinks_.erase(it);
}
void RtpVideoStreamReceiver::ReceivePacket(const RtpPacketReceived& packet) {
if (packet.payload_size() == 0) {
// Padding or keep-alive packet.
// TODO(nisse): Could drop empty packets earlier, but need to figure out how
// they should be counted in stats.
NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
return;
}
if (packet.PayloadType() == config_.rtp.red_payload_type) {
ParseAndHandleEncapsulatingHeader(packet);
return;
}
const auto type_it = payload_type_map_.find(packet.PayloadType());
if (type_it == payload_type_map_.end()) {
return;
}
auto depacketizer =
absl::WrapUnique(RtpDepacketizer::Create(type_it->second));
if (!depacketizer) {
RTC_LOG(LS_ERROR) << "Failed to create depacketizer.";
return;
}
RtpDepacketizer::ParsedPayload parsed_payload;
if (!depacketizer->Parse(&parsed_payload, packet.payload().data(),
packet.payload().size())) {
RTC_LOG(LS_WARNING) << "Failed parsing payload.";
return;
}
RTPHeader rtp_header;
packet.GetHeader(&rtp_header);
RTPVideoHeader video_header = parsed_payload.video_header();
video_header.rotation = kVideoRotation_0;
video_header.content_type = VideoContentType::UNSPECIFIED;
video_header.video_timing.flags = VideoSendTiming::kInvalid;
video_header.is_last_packet_in_frame = rtp_header.markerBit;
video_header.frame_marking.temporal_id = kNoTemporalIdx;
if (parsed_payload.video_header().codec == kVideoCodecVP9) {
const RTPVideoHeaderVP9& codec_header = absl::get<RTPVideoHeaderVP9>(
parsed_payload.video_header().video_type_header);
video_header.is_last_packet_in_frame |= codec_header.end_of_frame;
video_header.is_first_packet_in_frame |= codec_header.beginning_of_frame;
}
packet.GetExtension<VideoOrientation>(&video_header.rotation);
packet.GetExtension<VideoContentTypeExtension>(&video_header.content_type);
packet.GetExtension<VideoTimingExtension>(&video_header.video_timing);
packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay);
packet.GetExtension<FrameMarkingExtension>(&video_header.frame_marking);
// Color space should only be transmitted in the last packet of a frame,
// therefore, neglect it otherwise so that last_color_space_ is not reset by
// mistake.
if (video_header.is_last_packet_in_frame) {
video_header.color_space = packet.GetExtension<ColorSpaceExtension>();
if (video_header.color_space ||
video_header.frame_type == VideoFrameType::kVideoFrameKey) {
// Store color space since it's only transmitted when changed or for key
// frames. Color space will be cleared if a key frame is transmitted
// without color space information.
last_color_space_ = video_header.color_space;
} else if (last_color_space_) {
video_header.color_space = last_color_space_;
}
}
absl::optional<RtpGenericFrameDescriptor> generic_descriptor_wire;
generic_descriptor_wire.emplace();
const bool generic_descriptor_v00 =
packet.GetExtension<RtpGenericFrameDescriptorExtension00>(
&generic_descriptor_wire.value());
const bool generic_descriptor_v01 =
packet.GetExtension<RtpGenericFrameDescriptorExtension01>(
&generic_descriptor_wire.value());
if (generic_descriptor_v00 && generic_descriptor_v01) {
RTC_LOG(LS_WARNING) << "RTP packet had two different GFD versions.";
return;
}
if (generic_descriptor_v00 || generic_descriptor_v01) {
if (generic_descriptor_v00) {
generic_descriptor_wire->SetByteRepresentation(
packet.GetRawExtension<RtpGenericFrameDescriptorExtension00>());
} else {
generic_descriptor_wire->SetByteRepresentation(
packet.GetRawExtension<RtpGenericFrameDescriptorExtension01>());
}
video_header.is_first_packet_in_frame =
generic_descriptor_wire->FirstPacketInSubFrame();
video_header.is_last_packet_in_frame =
rtp_header.markerBit || generic_descriptor_wire->LastPacketInSubFrame();
if (generic_descriptor_wire->FirstPacketInSubFrame()) {
video_header.frame_type =
generic_descriptor_wire->FrameDependenciesDiffs().empty()
? VideoFrameType::kVideoFrameKey
: VideoFrameType::kVideoFrameDelta;
}
video_header.width = generic_descriptor_wire->Width();
video_header.height = generic_descriptor_wire->Height();
} else {
generic_descriptor_wire.reset();
}
OnReceivedPayloadData(parsed_payload.payload, parsed_payload.payload_length,
rtp_header, video_header, generic_descriptor_wire,
packet.recovered());
}
void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
if (packet.PayloadType() == config_.rtp.red_payload_type &&
packet.payload_size() > 0) {
if (packet.payload()[0] == config_.rtp.ulpfec_payload_type) {
rtp_receive_statistics_->FecPacketReceived(packet);
// Notify video_receiver about received FEC packets to avoid NACKing these
// packets.
NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
}
RTPHeader header;
packet.GetHeader(&header);
if (ulpfec_receiver_->AddReceivedRedPacket(
header, packet.data(), packet.size(),
config_.rtp.ulpfec_payload_type) != 0) {
return;
}
ulpfec_receiver_->ProcessReceivedFec();
}
}
// 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.
void RtpVideoStreamReceiver::NotifyReceiverOfEmptyPacket(uint16_t seq_num) {
reference_finder_->PaddingReceived(seq_num);
packet_buffer_->PaddingReceived(seq_num);
if (nack_module_) {
nack_module_->OnReceivedPacket(seq_num, /* is_keyframe = */ false,
/* is _recovered = */ false);
}
if (loss_notification_controller_) {
// TODO(bugs.webrtc.org/10336): Handle empty packets.
RTC_LOG(LS_WARNING)
<< "LossNotificationController does not expect empty packets.";
}
}
bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
size_t rtcp_packet_length) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
if (!receiving_) {
return false;
}
rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
int64_t rtt = 0;
rtp_rtcp_->RTT(config_.rtp.remote_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;
uint32_t recieved_ntp_secs = 0;
uint32_t recieved_ntp_frac = 0;
if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &recieved_ntp_secs,
&recieved_ntp_frac, &rtp_timestamp) != 0) {
// Waiting for RTCP.
return true;
}
NtpTime recieved_ntp(recieved_ntp_secs, recieved_ntp_frac);
int64_t time_since_recieved =
clock_->CurrentNtpInMilliseconds() - recieved_ntp.ToMs();
// Don't use old SRs to estimate time.
if (time_since_recieved <= 1) {
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);
}
int RtpVideoStreamReceiver::GetUniqueFramesSeen() const {
return packet_buffer_->GetUniqueFramesSeen();
}
void RtpVideoStreamReceiver::StartReceive() {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
receiving_ = true;
}
void RtpVideoStreamReceiver::StopReceive() {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
receiving_ = false;
}
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::InsertSpsPpsIntoTracker(uint8_t payload_type) {
auto codec_params_it = pt_codec_params_.find(payload_type);
if (codec_params_it == pt_codec_params_.end())
return;
RTC_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