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webrtc/video/rtp_video_stream_receiver2.cc
Per K 217b384c1b Remove rtp header extension from config of Call audio and video receivers 
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.

Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
2023-01-31 11:58:43 +00:00

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/*
* 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_receiver2.h"
#include <algorithm>
#include <limits>
#include <memory>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/types/optional.h"
#include "api/video/video_codec_type.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/source/create_video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.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/rtp_rtcp/source/ulpfec_receiver.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.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_requester.h"
#include "modules/video_coding/packet_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/ntp_time.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;
constexpr int kMaxPacketAgeToNack = 450;
int PacketBufferMaxSize(const FieldTrialsView& field_trials) {
// 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 =
field_trials.Lookup("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;
}
return packet_buffer_max_size;
}
std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
Clock* clock,
ReceiveStatistics* receive_statistics,
Transport* outgoing_transport,
RtcpRttStats* rtt_stats,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
bool non_sender_rtt_measurement,
uint32_t local_ssrc,
RtcEventLog* rtc_event_log) {
RtpRtcpInterface::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.rtcp_cname_callback = rtcp_cname_callback;
configuration.local_media_ssrc = local_ssrc;
configuration.non_sender_rtt_measurement = non_sender_rtt_measurement;
configuration.event_log = rtc_event_log;
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp =
ModuleRtpRtcpImpl2::Create(configuration);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
return rtp_rtcp;
}
std::unique_ptr<NackRequester> MaybeConstructNackModule(
TaskQueueBase* current_queue,
NackPeriodicProcessor* nack_periodic_processor,
const NackConfig& nack,
Clock* clock,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender,
const FieldTrialsView& field_trials) {
if (nack.rtp_history_ms == 0)
return nullptr;
// TODO(bugs.webrtc.org/12420): pass rtp_history_ms to the nack module.
return std::make_unique<NackRequester>(current_queue, nack_periodic_processor,
clock, nack_sender,
keyframe_request_sender, field_trials);
}
std::unique_ptr<UlpfecReceiver> MaybeConstructUlpfecReceiver(
uint32_t remote_ssrc,
int red_payload_type,
int ulpfec_payload_type,
RecoveredPacketReceiver* callback,
Clock* clock) {
RTC_DCHECK_GE(red_payload_type, -1);
RTC_DCHECK_GE(ulpfec_payload_type, -1);
if (red_payload_type == -1)
return nullptr;
// TODO(tommi, brandtr): Consider including this check too once
// `UlpfecReceiver` has been updated to not consider both red and ulpfec
// payload ids.
// if (ulpfec_payload_type == -1)
// return nullptr;
return std::make_unique<UlpfecReceiver>(remote_ssrc, ulpfec_payload_type,
callback, clock);
}
static const int kPacketLogIntervalMs = 10000;
} // namespace
RtpVideoStreamReceiver2::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_);
packet_sequence_checker_.Detach();
}
void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RequestKeyFrame() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
request_key_frame_ = true;
}
void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendNack(
const std::vector<uint16_t>& sequence_numbers,
bool buffering_allowed) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(!sequence_numbers.empty());
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 RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendLossNotification(
uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(buffering_allowed);
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 RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendBufferedRtcpFeedback() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
bool request_key_frame = false;
std::vector<uint16_t> nack_sequence_numbers;
absl::optional<LossNotificationState> lntf_state;
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);
}
}
void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::ClearLossNotificationState() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
lntf_state_.reset();
}
RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
TaskQueueBase* current_queue,
Clock* clock,
Transport* transport,
RtcpRttStats* rtt_stats,
PacketRouter* packet_router,
const VideoReceiveStreamInterface::Config* config,
ReceiveStatistics* rtp_receive_statistics,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
NackPeriodicProcessor* nack_periodic_processor,
OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const FieldTrialsView& field_trials,
RtcEventLog* event_log)
: field_trials_(field_trials),
worker_queue_(current_queue),
clock_(clock),
config_(*config),
packet_router_(packet_router),
ntp_estimator_(clock),
forced_playout_delay_max_ms_("max_ms", absl::nullopt),
forced_playout_delay_min_ms_("min_ms", absl::nullopt),
rtp_receive_statistics_(rtp_receive_statistics),
ulpfec_receiver_(
MaybeConstructUlpfecReceiver(config->rtp.remote_ssrc,
config->rtp.red_payload_type,
config->rtp.ulpfec_payload_type,
this,
clock_)),
red_payload_type_(config_.rtp.red_payload_type),
packet_sink_(config->rtp.packet_sink_),
receiving_(false),
last_packet_log_ms_(-1),
rtp_rtcp_(CreateRtpRtcpModule(
clock,
rtp_receive_statistics_,
transport,
rtt_stats,
rtcp_packet_type_counter_observer,
rtcp_cname_callback,
config_.rtp.rtcp_xr.receiver_reference_time_report,
config_.rtp.local_ssrc,
event_log)),
nack_periodic_processor_(nack_periodic_processor),
complete_frame_callback_(complete_frame_callback),
keyframe_request_method_(config_.rtp.keyframe_method),
// TODO(bugs.webrtc.org/10336): Let `rtcp_feedback_buffer_` communicate
// directly with `rtp_rtcp_`.
rtcp_feedback_buffer_(this, this, this),
nack_module_(MaybeConstructNackModule(current_queue,
nack_periodic_processor,
config_.rtp.nack,
clock_,
&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_,
field_trials_)),
packet_buffer_(kPacketBufferStartSize,
PacketBufferMaxSize(field_trials_)),
reference_finder_(std::make_unique<RtpFrameReferenceFinder>()),
has_received_frame_(false),
frames_decryptable_(false),
absolute_capture_time_interpolator_(clock) {
packet_sequence_checker_.Detach();
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.";
// 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);
if (config_.rtp.nack.rtp_history_ms > 0) {
rtp_receive_statistics_->SetMaxReorderingThreshold(config_.rtp.remote_ssrc,
kMaxPacketAgeToNack);
}
ParseFieldTrial(
{&forced_playout_delay_max_ms_, &forced_playout_delay_min_ms_},
field_trials_.Lookup("WebRTC-ForcePlayoutDelay"));
if (config_.rtp.lntf.enabled) {
loss_notification_controller_ =
std::make_unique<LossNotificationController>(&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_);
}
// Only construct the encrypted receiver if frame encryption is enabled.
if (config_.crypto_options.sframe.require_frame_encryption) {
buffered_frame_decryptor_ =
std::make_unique<BufferedFrameDecryptor>(this, this, field_trials_);
if (frame_decryptor != nullptr) {
buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
}
}
if (frame_transformer) {
frame_transformer_delegate_ =
rtc::make_ref_counted<RtpVideoStreamReceiverFrameTransformerDelegate>(
this, std::move(frame_transformer), rtc::Thread::Current(),
config_.rtp.remote_ssrc);
frame_transformer_delegate_->Init();
}
}
RtpVideoStreamReceiver2::~RtpVideoStreamReceiver2() {
if (packet_router_)
packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
ulpfec_receiver_.reset();
if (frame_transformer_delegate_)
frame_transformer_delegate_->Reset();
}
void RtpVideoStreamReceiver2::AddReceiveCodec(
uint8_t payload_type,
VideoCodecType video_codec,
const std::map<std::string, std::string>& codec_params,
bool raw_payload) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (codec_params.count(cricket::kH264FmtpSpsPpsIdrInKeyframe) > 0 ||
field_trials_.IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
packet_buffer_.ForceSpsPpsIdrIsH264Keyframe();
}
payload_type_map_.emplace(
payload_type, raw_payload ? std::make_unique<VideoRtpDepacketizerRaw>()
: CreateVideoRtpDepacketizer(video_codec));
pt_codec_params_.emplace(payload_type, codec_params);
}
void RtpVideoStreamReceiver2::RemoveReceiveCodec(uint8_t payload_type) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
auto codec_params_it = pt_codec_params_.find(payload_type);
if (codec_params_it == pt_codec_params_.end())
return;
const bool sps_pps_idr_in_key_frame =
codec_params_it->second.count(cricket::kH264FmtpSpsPpsIdrInKeyframe) > 0;
pt_codec_params_.erase(codec_params_it);
payload_type_map_.erase(payload_type);
if (sps_pps_idr_in_key_frame) {
bool reset_setting = true;
for (auto& [unused, codec_params] : pt_codec_params_) {
if (codec_params.count(cricket::kH264FmtpSpsPpsIdrInKeyframe) > 0) {
reset_setting = false;
break;
}
}
if (reset_setting) {
packet_buffer_.ResetSpsPpsIdrIsH264Keyframe();
}
}
}
void RtpVideoStreamReceiver2::RemoveReceiveCodecs() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
pt_codec_params_.clear();
payload_type_map_.clear();
packet_buffer_.ResetSpsPpsIdrIsH264Keyframe();
}
absl::optional<Syncable::Info> RtpVideoStreamReceiver2::GetSyncInfo() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
Syncable::Info info;
if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs,
&info.capture_time_ntp_frac,
/*rtcp_arrival_time_secs=*/nullptr,
/*rtcp_arrival_time_frac=*/nullptr,
&info.capture_time_source_clock) != 0) {
return absl::nullopt;
}
if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_) {
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;
}
RtpVideoStreamReceiver2::ParseGenericDependenciesResult
RtpVideoStreamReceiver2::ParseGenericDependenciesExtension(
const RtpPacketReceived& rtp_packet,
RTPVideoHeader* video_header) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (rtp_packet.HasExtension<RtpDependencyDescriptorExtension>()) {
webrtc::DependencyDescriptor dependency_descriptor;
if (!rtp_packet.GetExtension<RtpDependencyDescriptorExtension>(
video_structure_.get(), &dependency_descriptor)) {
// Descriptor is there, but failed to parse. Either it is invalid,
// or too old packet (after relevant video_structure_ changed),
// or too new packet (before relevant video_structure_ arrived).
// Drop such packet to be on the safe side.
// TODO(bugs.webrtc.org/10342): Stash too new packet.
Timestamp now = clock_->CurrentTime();
if (now - last_logged_failed_to_parse_dd_ > TimeDelta::Seconds(1)) {
last_logged_failed_to_parse_dd_ = now;
RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc()
<< " Failed to parse dependency descriptor.";
}
return kDropPacket;
}
if (dependency_descriptor.attached_structure != nullptr &&
!dependency_descriptor.first_packet_in_frame) {
RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc()
<< "Invalid dependency descriptor: structure "
"attached to non first packet of a frame.";
return kDropPacket;
}
video_header->is_first_packet_in_frame =
dependency_descriptor.first_packet_in_frame;
video_header->is_last_packet_in_frame =
dependency_descriptor.last_packet_in_frame;
int64_t frame_id =
frame_id_unwrapper_.Unwrap(dependency_descriptor.frame_number);
auto& generic_descriptor_info = video_header->generic.emplace();
generic_descriptor_info.frame_id = frame_id;
generic_descriptor_info.spatial_index =
dependency_descriptor.frame_dependencies.spatial_id;
generic_descriptor_info.temporal_index =
dependency_descriptor.frame_dependencies.temporal_id;
for (int fdiff : dependency_descriptor.frame_dependencies.frame_diffs) {
generic_descriptor_info.dependencies.push_back(frame_id - fdiff);
}
generic_descriptor_info.decode_target_indications =
dependency_descriptor.frame_dependencies.decode_target_indications;
if (dependency_descriptor.resolution) {
video_header->width = dependency_descriptor.resolution->Width();
video_header->height = dependency_descriptor.resolution->Height();
}
// FrameDependencyStructure is sent in dependency descriptor of the first
// packet of a key frame and required for parsed dependency descriptor in
// all the following packets until next key frame.
// Save it if there is a (potentially) new structure.
if (dependency_descriptor.attached_structure) {
RTC_DCHECK(dependency_descriptor.first_packet_in_frame);
if (video_structure_frame_id_ > frame_id) {
RTC_LOG(LS_WARNING)
<< "Arrived key frame with id " << frame_id << " and structure id "
<< dependency_descriptor.attached_structure->structure_id
<< " is older than the latest received key frame with id "
<< *video_structure_frame_id_ << " and structure id "
<< video_structure_->structure_id;
return kDropPacket;
}
video_structure_ = std::move(dependency_descriptor.attached_structure);
video_structure_frame_id_ = frame_id;
video_header->frame_type = VideoFrameType::kVideoFrameKey;
} else {
video_header->frame_type = VideoFrameType::kVideoFrameDelta;
}
return kHasGenericDescriptor;
}
RtpGenericFrameDescriptor generic_frame_descriptor;
if (!rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>(
&generic_frame_descriptor)) {
return kNoGenericDescriptor;
}
video_header->is_first_packet_in_frame =
generic_frame_descriptor.FirstPacketInSubFrame();
video_header->is_last_packet_in_frame =
generic_frame_descriptor.LastPacketInSubFrame();
if (generic_frame_descriptor.FirstPacketInSubFrame()) {
video_header->frame_type =
generic_frame_descriptor.FrameDependenciesDiffs().empty()
? VideoFrameType::kVideoFrameKey
: VideoFrameType::kVideoFrameDelta;
auto& generic_descriptor_info = video_header->generic.emplace();
int64_t frame_id =
frame_id_unwrapper_.Unwrap(generic_frame_descriptor.FrameId());
generic_descriptor_info.frame_id = frame_id;
generic_descriptor_info.spatial_index =
generic_frame_descriptor.SpatialLayer();
generic_descriptor_info.temporal_index =
generic_frame_descriptor.TemporalLayer();
for (uint16_t fdiff : generic_frame_descriptor.FrameDependenciesDiffs()) {
generic_descriptor_info.dependencies.push_back(frame_id - fdiff);
}
}
video_header->width = generic_frame_descriptor.Width();
video_header->height = generic_frame_descriptor.Height();
return kHasGenericDescriptor;
}
void RtpVideoStreamReceiver2::OnReceivedPayloadData(
rtc::CopyOnWriteBuffer codec_payload,
const RtpPacketReceived& rtp_packet,
const RTPVideoHeader& video) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
auto packet =
std::make_unique<video_coding::PacketBuffer::Packet>(rtp_packet, video);
int64_t unwrapped_rtp_seq_num =
rtp_seq_num_unwrapper_.Unwrap(rtp_packet.SequenceNumber());
RtpPacketInfo& packet_info =
packet_infos_
.emplace(unwrapped_rtp_seq_num,
RtpPacketInfo(rtp_packet.Ssrc(), rtp_packet.Csrcs(),
rtp_packet.Timestamp(),
/*receive_time_ms=*/clock_->CurrentTime()))
.first->second;
// Try to extrapolate absolute capture time if it is missing.
packet_info.set_absolute_capture_time(
absolute_capture_time_interpolator_.OnReceivePacket(
AbsoluteCaptureTimeInterpolator::GetSource(packet_info.ssrc(),
packet_info.csrcs()),
packet_info.rtp_timestamp(),
// Assume frequency is the same one for all video frames.
kVideoPayloadTypeFrequency,
rtp_packet.GetExtension<AbsoluteCaptureTimeExtension>()));
RTPVideoHeader& video_header = packet->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_packet.Marker();
rtp_packet.GetExtension<VideoOrientation>(&video_header.rotation);
rtp_packet.GetExtension<VideoContentTypeExtension>(
&video_header.content_type);
rtp_packet.GetExtension<VideoTimingExtension>(&video_header.video_timing);
if (forced_playout_delay_max_ms_ && forced_playout_delay_min_ms_) {
video_header.playout_delay.max_ms = *forced_playout_delay_max_ms_;
video_header.playout_delay.min_ms = *forced_playout_delay_min_ms_;
} else {
rtp_packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay);
}
ParseGenericDependenciesResult generic_descriptor_state =
ParseGenericDependenciesExtension(rtp_packet, &video_header);
if (!rtp_packet.recovered()) {
UpdatePacketReceiveTimestamps(
rtp_packet, video_header.frame_type == VideoFrameType::kVideoFrameKey);
}
if (generic_descriptor_state == kDropPacket) {
Timestamp now = clock_->CurrentTime();
if (video_structure_ == nullptr &&
next_keyframe_request_for_missing_video_structure_ < now) {
// No video structure received yet, most likely part of the initial
// keyframe was lost.
RequestKeyFrame();
next_keyframe_request_for_missing_video_structure_ =
now + TimeDelta::Seconds(1);
}
return;
}
// 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 = rtp_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_;
}
}
video_header.video_frame_tracking_id =
rtp_packet.GetExtension<VideoFrameTrackingIdExtension>();
if (loss_notification_controller_) {
if (rtp_packet.recovered()) {
// TODO(bugs.webrtc.org/10336): Implement support for reordering.
RTC_LOG(LS_INFO)
<< "LossNotificationController does not support reordering.";
} else if (generic_descriptor_state == kNoGenericDescriptor) {
RTC_LOG(LS_WARNING) << "LossNotificationController requires generic "
"frame descriptor, but it is missing.";
} else {
if (video_header.is_first_packet_in_frame) {
RTC_DCHECK(video_header.generic);
LossNotificationController::FrameDetails frame;
frame.is_keyframe =
video_header.frame_type == VideoFrameType::kVideoFrameKey;
frame.frame_id = video_header.generic->frame_id;
frame.frame_dependencies = video_header.generic->dependencies;
loss_notification_controller_->OnReceivedPacket(
rtp_packet.SequenceNumber(), &frame);
} else {
loss_notification_controller_->OnReceivedPacket(
rtp_packet.SequenceNumber(), nullptr);
}
}
}
if (nack_module_) {
const bool is_keyframe =
video_header.is_first_packet_in_frame &&
video_header.frame_type == VideoFrameType::kVideoFrameKey;
packet->times_nacked = nack_module_->OnReceivedPacket(
rtp_packet.SequenceNumber(), is_keyframe, rtp_packet.recovered());
} else {
packet->times_nacked = -1;
}
if (codec_payload.size() == 0) {
NotifyReceiverOfEmptyPacket(packet->seq_num);
rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
return;
}
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->payload_type != last_payload_type_) {
last_payload_type_ = packet->payload_type;
InsertSpsPpsIntoTracker(packet->payload_type);
}
video_coding::H264SpsPpsTracker::FixedBitstream fixed =
tracker_.CopyAndFixBitstream(
rtc::MakeArrayView(codec_payload.cdata(), codec_payload.size()),
&packet->video_header);
switch (fixed.action) {
case video_coding::H264SpsPpsTracker::kRequestKeyframe:
rtcp_feedback_buffer_.RequestKeyFrame();
rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
[[fallthrough]];
case video_coding::H264SpsPpsTracker::kDrop:
return;
case video_coding::H264SpsPpsTracker::kInsert:
packet->video_payload = std::move(fixed.bitstream);
break;
}
} else {
packet->video_payload = std::move(codec_payload);
}
rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
frame_counter_.Add(packet->timestamp);
OnInsertedPacket(packet_buffer_.InsertPacket(std::move(packet)));
}
void RtpVideoStreamReceiver2::OnRecoveredPacket(
const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (packet.PayloadType() == red_payload_type_) {
RTC_LOG(LS_WARNING) << "Discarding recovered packet with RED encapsulation";
return;
}
ReceivePacket(packet);
}
// This method handles both regular RTP packets and packets recovered
// via FlexFEC.
void RtpVideoStreamReceiver2::OnRtpPacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (!receiving_)
return;
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);
}
if (packet_sink_) {
packet_sink_->OnRtpPacket(packet);
}
}
void RtpVideoStreamReceiver2::RequestKeyFrame() {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
// 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_method_ == KeyFrameReqMethod::kPliRtcp) {
rtp_rtcp_->SendPictureLossIndication();
} else if (keyframe_request_method_ == KeyFrameReqMethod::kFirRtcp) {
rtp_rtcp_->SendFullIntraRequest();
}
}
void RtpVideoStreamReceiver2::SendNack(
const std::vector<uint16_t>& sequence_numbers,
bool /*buffering_allowed*/) {
rtp_rtcp_->SendNack(sequence_numbers);
}
void RtpVideoStreamReceiver2::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 RtpVideoStreamReceiver2::IsDecryptable() const {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
return frames_decryptable_;
}
void RtpVideoStreamReceiver2::OnInsertedPacket(
video_coding::PacketBuffer::InsertResult result) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_RUN_ON(&worker_task_checker_);
video_coding::PacketBuffer::Packet* first_packet = nullptr;
int max_nack_count;
int64_t min_recv_time;
int64_t max_recv_time;
std::vector<rtc::ArrayView<const uint8_t>> payloads;
RtpPacketInfos::vector_type packet_infos;
bool frame_boundary = true;
for (auto& packet : result.packets) {
// PacketBuffer promisses frame boundaries are correctly set on each
// packet. Document that assumption with the DCHECKs.
RTC_DCHECK_EQ(frame_boundary, packet->is_first_packet_in_frame());
int64_t unwrapped_rtp_seq_num =
rtp_seq_num_unwrapper_.Unwrap(packet->seq_num);
RTC_DCHECK_GT(packet_infos_.count(unwrapped_rtp_seq_num), 0);
RtpPacketInfo& packet_info = packet_infos_[unwrapped_rtp_seq_num];
if (packet->is_first_packet_in_frame()) {
first_packet = packet.get();
max_nack_count = packet->times_nacked;
min_recv_time = packet_info.receive_time().ms();
max_recv_time = packet_info.receive_time().ms();
} else {
max_nack_count = std::max(max_nack_count, packet->times_nacked);
min_recv_time = std::min(min_recv_time, packet_info.receive_time().ms());
max_recv_time = std::max(max_recv_time, packet_info.receive_time().ms());
}
payloads.emplace_back(packet->video_payload);
packet_infos.push_back(packet_info);
frame_boundary = packet->is_last_packet_in_frame();
if (packet->is_last_packet_in_frame()) {
auto depacketizer_it = payload_type_map_.find(first_packet->payload_type);
RTC_CHECK(depacketizer_it != payload_type_map_.end());
rtc::scoped_refptr<EncodedImageBuffer> bitstream =
depacketizer_it->second->AssembleFrame(payloads);
if (!bitstream) {
// Failed to assemble a frame. Discard and continue.
continue;
}
const video_coding::PacketBuffer::Packet& last_packet = *packet;
OnAssembledFrame(std::make_unique<RtpFrameObject>(
first_packet->seq_num, //
last_packet.seq_num, //
last_packet.marker_bit, //
max_nack_count, //
min_recv_time, //
max_recv_time, //
first_packet->timestamp, //
ntp_estimator_.Estimate(first_packet->timestamp), //
last_packet.video_header.video_timing, //
first_packet->payload_type, //
first_packet->codec(), //
last_packet.video_header.rotation, //
last_packet.video_header.content_type, //
first_packet->video_header, //
last_packet.video_header.color_space, //
RtpPacketInfos(std::move(packet_infos)), //
std::move(bitstream)));
payloads.clear();
packet_infos.clear();
}
}
RTC_DCHECK(frame_boundary);
if (result.buffer_cleared) {
last_received_rtp_system_time_.reset();
last_received_keyframe_rtp_system_time_.reset();
last_received_keyframe_rtp_timestamp_.reset();
packet_infos_.clear();
RequestKeyFrame();
}
}
void RtpVideoStreamReceiver2::OnAssembledFrame(
std::unique_ptr<RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(frame);
const absl::optional<RTPVideoHeader::GenericDescriptorInfo>& descriptor =
frame->GetRtpVideoHeader().generic;
if (loss_notification_controller_ && descriptor) {
loss_notification_controller_->OnAssembledFrame(
frame->first_seq_num(), descriptor->frame_id,
absl::c_linear_search(descriptor->decode_target_indications,
DecodeTargetIndication::kDiscardable),
descriptor->dependencies);
}
// 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;
}
// Reset `reference_finder_` if `frame` is new and the codec have changed.
if (current_codec_) {
bool frame_is_newer =
AheadOf(frame->Timestamp(), last_assembled_frame_rtp_timestamp_);
if (frame->codec_type() != current_codec_) {
if (frame_is_newer) {
// When we reset the `reference_finder_` we don't want new picture ids
// to overlap with old picture ids. To ensure that doesn't happen we
// start from the `last_completed_picture_id_` and add an offset in case
// of reordering.
reference_finder_ = std::make_unique<RtpFrameReferenceFinder>(
last_completed_picture_id_ + std::numeric_limits<uint16_t>::max());
current_codec_ = frame->codec_type();
} else {
// Old frame from before the codec switch, discard it.
return;
}
}
if (frame_is_newer) {
last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
}
} else {
current_codec_ = frame->codec_type();
last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
}
if (buffered_frame_decryptor_ != nullptr) {
buffered_frame_decryptor_->ManageEncryptedFrame(std::move(frame));
} else if (frame_transformer_delegate_) {
frame_transformer_delegate_->TransformFrame(std::move(frame));
} else {
OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame)));
}
}
void RtpVideoStreamReceiver2::OnCompleteFrames(
RtpFrameReferenceFinder::ReturnVector frames) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
for (auto& frame : frames) {
last_seq_num_for_pic_id_[frame->Id()] = frame->last_seq_num();
last_completed_picture_id_ =
std::max(last_completed_picture_id_, frame->Id());
complete_frame_callback_->OnCompleteFrame(std::move(frame));
}
}
void RtpVideoStreamReceiver2::OnDecryptedFrame(
std::unique_ptr<RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame)));
}
void RtpVideoStreamReceiver2::OnDecryptionStatusChange(
FrameDecryptorInterface::Status status) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
// Called from BufferedFrameDecryptor::DecryptFrame.
frames_decryptable_ =
(status == FrameDecryptorInterface::Status::kOk) ||
(status == FrameDecryptorInterface::Status::kRecoverable);
}
void RtpVideoStreamReceiver2::SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
// TODO(bugs.webrtc.org/11993): Update callers or post the operation over to
// the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (buffered_frame_decryptor_ == nullptr) {
buffered_frame_decryptor_ =
std::make_unique<BufferedFrameDecryptor>(this, this, field_trials_);
}
buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
}
void RtpVideoStreamReceiver2::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
frame_transformer_delegate_ =
rtc::make_ref_counted<RtpVideoStreamReceiverFrameTransformerDelegate>(
this, std::move(frame_transformer), rtc::Thread::Current(),
config_.rtp.remote_ssrc);
frame_transformer_delegate_->Init();
}
void RtpVideoStreamReceiver2::UpdateRtt(int64_t max_rtt_ms) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (nack_module_)
nack_module_->UpdateRtt(max_rtt_ms);
}
void RtpVideoStreamReceiver2::OnLocalSsrcChange(uint32_t local_ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_rtcp_->SetLocalSsrc(local_ssrc);
}
void RtpVideoStreamReceiver2::SetRtcpMode(RtcpMode mode) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_rtcp_->SetRTCPStatus(mode);
}
void RtpVideoStreamReceiver2::SetReferenceTimeReport(bool enabled) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_rtcp_->SetNonSenderRttMeasurement(enabled);
}
void RtpVideoStreamReceiver2::SetPacketSink(
RtpPacketSinkInterface* packet_sink) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
packet_sink_ = packet_sink;
}
void RtpVideoStreamReceiver2::SetLossNotificationEnabled(bool enabled) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (enabled && !loss_notification_controller_) {
loss_notification_controller_ =
std::make_unique<LossNotificationController>(&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_);
} else if (!enabled && loss_notification_controller_) {
loss_notification_controller_.reset();
rtcp_feedback_buffer_.ClearLossNotificationState();
}
}
void RtpVideoStreamReceiver2::SetNackHistory(TimeDelta history) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (history.ms() == 0) {
nack_module_.reset();
} else if (!nack_module_) {
nack_module_ = std::make_unique<NackRequester>(
worker_queue_, nack_periodic_processor_, clock_, &rtcp_feedback_buffer_,
&rtcp_feedback_buffer_, field_trials_);
}
rtp_receive_statistics_->SetMaxReorderingThreshold(
config_.rtp.remote_ssrc,
history.ms() > 0 ? kMaxPacketAgeToNack : kDefaultMaxReorderingThreshold);
}
int RtpVideoStreamReceiver2::ulpfec_payload_type() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return ulpfec_receiver_ ? ulpfec_receiver_->ulpfec_payload_type() : -1;
}
int RtpVideoStreamReceiver2::red_payload_type() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return red_payload_type_;
}
void RtpVideoStreamReceiver2::SetProtectionPayloadTypes(
int red_payload_type,
int ulpfec_payload_type) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(red_payload_type >= -1 && red_payload_type < 0x80);
RTC_DCHECK(ulpfec_payload_type >= -1 && ulpfec_payload_type < 0x80);
red_payload_type_ = red_payload_type;
ulpfec_receiver_ =
MaybeConstructUlpfecReceiver(config_.rtp.remote_ssrc, red_payload_type,
ulpfec_payload_type, this, clock_);
}
absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedPacketMs() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (last_received_rtp_system_time_) {
return absl::optional<int64_t>(last_received_rtp_system_time_->ms());
}
return absl::nullopt;
}
absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (last_received_keyframe_rtp_system_time_) {
return absl::optional<int64_t>(
last_received_keyframe_rtp_system_time_->ms());
}
return absl::nullopt;
}
void RtpVideoStreamReceiver2::ManageFrame(
std::unique_ptr<RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame)));
}
void RtpVideoStreamReceiver2::ReceivePacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
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() == red_payload_type_) {
ParseAndHandleEncapsulatingHeader(packet);
return;
}
const auto type_it = payload_type_map_.find(packet.PayloadType());
if (type_it == payload_type_map_.end()) {
return;
}
absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
type_it->second->Parse(packet.PayloadBuffer());
if (parsed_payload == absl::nullopt) {
RTC_LOG(LS_WARNING) << "Failed parsing payload.";
return;
}
OnReceivedPayloadData(std::move(parsed_payload->video_payload), packet,
parsed_payload->video_header);
}
void RtpVideoStreamReceiver2::ParseAndHandleEncapsulatingHeader(
const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_EQ(packet.PayloadType(), red_payload_type_);
if (!ulpfec_receiver_ || packet.payload_size() == 0U)
return;
if (packet.payload()[0] == ulpfec_receiver_->ulpfec_payload_type()) {
// Notify video_receiver about received FEC packets to avoid NACKing these
// packets.
NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
}
if (ulpfec_receiver_->AddReceivedRedPacket(packet)) {
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 RtpVideoStreamReceiver2::NotifyReceiverOfEmptyPacket(uint16_t seq_num) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_RUN_ON(&worker_task_checker_);
OnCompleteFrames(reference_finder_->PaddingReceived(seq_num));
OnInsertedPacket(packet_buffer_.InsertPadding(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 RtpVideoStreamReceiver2::DeliverRtcp(const uint8_t* rtcp_packet,
size_t rtcp_packet_length) {
RTC_DCHECK_RUN_ON(&packet_sequence_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 received_ntp_secs = 0;
uint32_t received_ntp_frac = 0;
if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &received_ntp_secs,
&received_ntp_frac, &rtp_timestamp) != 0) {
// Waiting for RTCP.
return true;
}
NtpTime received_ntp(received_ntp_secs, received_ntp_frac);
int64_t time_since_received =
clock_->CurrentNtpInMilliseconds() - received_ntp.ToMs();
// Don't use old SRs to estimate time.
if (time_since_received <= 1) {
ntp_estimator_.UpdateRtcpTimestamp(
TimeDelta::Millis(rtt), NtpTime(ntp_secs, ntp_frac), rtp_timestamp);
absl::optional<int64_t> remote_to_local_clock_offset =
ntp_estimator_.EstimateRemoteToLocalClockOffset();
if (remote_to_local_clock_offset.has_value()) {
capture_clock_offset_updater_.SetRemoteToLocalClockOffset(
*remote_to_local_clock_offset);
}
}
return true;
}
void RtpVideoStreamReceiver2::FrameContinuous(int64_t picture_id) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (!nack_module_)
return;
int seq_num = -1;
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 RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
int seq_num = -1;
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) {
int64_t unwrapped_rtp_seq_num = rtp_seq_num_unwrapper_.Unwrap(seq_num);
packet_infos_.erase(packet_infos_.begin(),
packet_infos_.upper_bound(unwrapped_rtp_seq_num));
packet_buffer_.ClearTo(seq_num);
reference_finder_->ClearTo(seq_num);
}
}
void RtpVideoStreamReceiver2::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
: RtcpMode::kOff);
}
void RtpVideoStreamReceiver2::StartReceive() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
receiving_ = true;
}
void RtpVideoStreamReceiver2::StopReceive() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
receiving_ = false;
}
void RtpVideoStreamReceiver2::InsertSpsPpsIntoTracker(uint8_t payload_type) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_RUN_ON(&worker_task_checker_);
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());
}
void RtpVideoStreamReceiver2::UpdatePacketReceiveTimestamps(
const RtpPacketReceived& packet,
bool is_keyframe) {
Timestamp now = clock_->CurrentTime();
if (is_keyframe ||
last_received_keyframe_rtp_timestamp_ == packet.Timestamp()) {
last_received_keyframe_rtp_timestamp_ = packet.Timestamp();
last_received_keyframe_rtp_system_time_ = now;
}
last_received_rtp_system_time_ = now;
last_received_rtp_timestamp_ = packet.Timestamp();
// 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: " << ToString(packet.arrival_time());
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();
}
}
} // namespace webrtc
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