Google Git
Sign in
webrtc / src / ba91dbcb3e55e336a410b7a3438c325d833c6171 / . / video / rtp_video_stream_receiver2.cc
blob: 63d8c3835d08f98c28d0c80738ac41f2f76bc0a6 [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_receiver2.h"
#include <algorithm>
#include <limits>
#include <memory>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/base/macros.h"
#include "absl/memory/memory.h"
#include "absl/types/optional.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/ulpfec_receiver.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/video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
#include "modules/utility/include/process_thread.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_module2.h"
#include "modules/video_coding/packet_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/ntp_time.h"
#include "video/receive_statistics_proxy2.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;
int PacketBufferMaxSize() {
// 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;
}
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) {
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;
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp =
ModuleRtpRtcpImpl2::Create(configuration);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
return rtp_rtcp;
}
std::unique_ptr<NackModule2> MaybeConstructNackModule(
TaskQueueBase* current_queue,
const VideoReceiveStream::Config& config,
Clock* clock,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender) {
if (config.rtp.nack.rtp_history_ms == 0)
return nullptr;
return std::make_unique<NackModule2>(current_queue, clock, nack_sender,
keyframe_request_sender);
}
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_);
}
void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RequestKeyFrame() {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
request_key_frame_ = true;
}
void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendNack(
const std::vector<uint16_t>& sequence_numbers,
bool buffering_allowed) {
RTC_DCHECK_RUN_ON(&worker_task_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(&worker_task_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(&worker_task_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);
}
}
RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
TaskQueueBase* current_queue,
Clock* clock,
Transport* transport,
RtcpRttStats* rtt_stats,
PacketRouter* packet_router,
const VideoReceiveStream::Config* config,
ReceiveStatistics* rtp_receive_statistics,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
ProcessThread* process_thread,
NackSender* nack_sender,
KeyFrameRequestSender* keyframe_request_sender,
video_coding::OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
: clock_(clock),
config_(*config),
packet_router_(packet_router),
process_thread_(process_thread),
ntp_estimator_(clock),
rtp_header_extensions_(config_.rtp.extensions),
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_(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,
rtcp_packet_type_counter_observer,
rtcp_cname_callback,
config_.rtp.rtcp_xr.receiver_reference_time_report,
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),
nack_module_(MaybeConstructNackModule(current_queue,
config_,
clock_,
&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_)),
packet_buffer_(clock_, kPacketBufferStartSize, PacketBufferMaxSize()),
has_received_frame_(false),
frames_decryptable_(false),
absolute_capture_time_receiver_(clock) {
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);
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);
}
ParseFieldTrial(
{&forced_playout_delay_max_ms_, &forced_playout_delay_min_ms_},
field_trial::FindFullName("WebRTC-ForcePlayoutDelay"));
process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
if (config_.rtp.lntf.enabled) {
loss_notification_controller_ =
std::make_unique<LossNotificationController>(&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_);
}
reference_finder_ =
std::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_ =
std::make_unique<BufferedFrameDecryptor>(this, this);
if (frame_decryptor != nullptr) {
buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
}
}
if (frame_transformer) {
frame_transformer_delegate_ = new rtc::RefCountedObject<
RtpVideoStreamReceiverFrameTransformerDelegate>(
this, std::move(frame_transformer), rtc::Thread::Current(),
config_.rtp.remote_ssrc);
frame_transformer_delegate_->Init();
}
}
RtpVideoStreamReceiver2::~RtpVideoStreamReceiver2() {
RTC_DCHECK(secondary_sinks_.empty());
process_thread_->DeRegisterModule(rtp_rtcp_.get());
if (packet_router_)
packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
UpdateHistograms();
if (frame_transformer_delegate_)
frame_transformer_delegate_->Reset();
}
void RtpVideoStreamReceiver2::AddReceiveCodec(
uint8_t payload_type,
const VideoCodec& video_codec,
const std::map<std::string, std::string>& codec_params,
bool raw_payload) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
if (codec_params.count(cricket::kH264FmtpSpsPpsIdrInKeyframe) ||
field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
packet_buffer_.ForceSpsPpsIdrIsH264Keyframe();
}
payload_type_map_.emplace(
payload_type, raw_payload
? std::make_unique<VideoRtpDepacketizerRaw>()
: CreateVideoRtpDepacketizer(video_codec.codecType));
pt_codec_params_.emplace(payload_type, codec_params);
}
absl::optional<Syncable::Info> RtpVideoStreamReceiver2::GetSyncInfo() const {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
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;
}
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;
}
RtpVideoStreamReceiver2::ParseGenericDependenciesResult
RtpVideoStreamReceiver2::ParseGenericDependenciesExtension(
const RtpPacketReceived& rtp_packet,
RTPVideoHeader* video_header) {
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.
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(&worker_task_checker_);
auto packet = std::make_unique<video_coding::PacketBuffer::Packet>(
rtp_packet, video, ntp_estimator_.Estimate(rtp_packet.Timestamp()),
clock_->TimeInMilliseconds());
// Try to extrapolate absolute capture time if it is missing.
packet->packet_info.set_absolute_capture_time(
absolute_capture_time_receiver_.OnReceivePacket(
AbsoluteCaptureTimeReceiver::GetSource(packet->packet_info.ssrc(),
packet->packet_info.csrcs()),
packet->packet_info.rtp_timestamp(),
// Assume frequency is the same one for all video frames.
kVideoPayloadTypeFrequency,
packet->packet_info.absolute_capture_time()));
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();
if (const auto* vp9_header =
absl::get_if<RTPVideoHeaderVP9>(&video_header.video_type_header)) {
video_header.is_last_packet_in_frame |= vp9_header->end_of_frame;
video_header.is_first_packet_in_frame |= vp9_header->beginning_of_frame;
}
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 (generic_descriptor_state == kDropPacket)
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_;
}
}
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();
ABSL_FALLTHROUGH_INTENDED;
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 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 RtpVideoStreamReceiver2::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();
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 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_sender_) {
keyframe_request_sender_->RequestKeyFrame();
} else {
rtp_rtcp_->SendPictureLossIndication();
}
}
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::IsUlpfecEnabled() const {
return config_.rtp.ulpfec_payload_type != -1;
}
bool RtpVideoStreamReceiver2::IsRetransmissionsEnabled() const {
return config_.rtp.nack.rtp_history_ms > 0;
}
void RtpVideoStreamReceiver2::RequestPacketRetransmit(
const std::vector<uint16_t>& sequence_numbers) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
rtp_rtcp_->SendNack(sequence_numbers);
}
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(&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());
if (packet->is_first_packet_in_frame()) {
first_packet = packet.get();
max_nack_count = packet->times_nacked;
min_recv_time = packet->packet_info.receive_time_ms();
max_recv_time = packet->packet_info.receive_time_ms();
payloads.clear();
packet_infos.clear();
} else {
max_nack_count = std::max(max_nack_count, packet->times_nacked);
min_recv_time =
std::min(min_recv_time, packet->packet_info.receive_time_ms());
max_recv_time =
std::max(max_recv_time, packet->packet_info.receive_time_ms());
}
payloads.emplace_back(packet->video_payload);
packet_infos.push_back(packet->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<video_coding::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, //
first_packet->ntp_time_ms, //
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)));
}
}
RTC_DCHECK(frame_boundary);
if (result.buffer_cleared) {
RequestKeyFrame();
}
}
void RtpVideoStreamReceiver2::OnAssembledFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&worker_task_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<video_coding::RtpFrameReferenceFinder>(
this, 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 {
reference_finder_->ManageFrame(std::move(frame));
}
}
void RtpVideoStreamReceiver2::OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
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();
last_completed_picture_id_ =
std::max(last_completed_picture_id_, frame->id.picture_id);
complete_frame_callback_->OnCompleteFrame(std::move(frame));
}
void RtpVideoStreamReceiver2::OnDecryptedFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
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) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
if (buffered_frame_decryptor_ == nullptr) {
buffered_frame_decryptor_ =
std::make_unique<BufferedFrameDecryptor>(this, this);
}
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_ =
new rtc::RefCountedObject<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(&worker_task_checker_);
if (nack_module_)
nack_module_->UpdateRtt(max_rtt_ms);
}
absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedPacketMs() const {
return packet_buffer_.LastReceivedPacketMs();
}
absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
const {
return packet_buffer_.LastReceivedKeyframePacketMs();
}
void RtpVideoStreamReceiver2::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 RtpVideoStreamReceiver2::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 RtpVideoStreamReceiver2::ManageFrame(
std::unique_ptr<video_coding::RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
reference_finder_->ManageFrame(std::move(frame));
}
void RtpVideoStreamReceiver2::ReceivePacket(const RtpPacketReceived& packet) {
RTC_DCHECK_RUN_ON(&worker_task_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() == 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;
}
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(&worker_task_checker_);
if (packet.PayloadType() == config_.rtp.red_payload_type &&
packet.payload_size() > 0) {
if (packet.payload()[0] == config_.rtp.ulpfec_payload_type) {
// Notify video_receiver about received FEC packets to avoid NACKing these
// packets.
NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
}
if (!ulpfec_receiver_->AddReceivedRedPacket(
packet, config_.rtp.ulpfec_payload_type)) {
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 RtpVideoStreamReceiver2::NotifyReceiverOfEmptyPacket(uint16_t seq_num) {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
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(&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);
absl::optional<int64_t> remote_to_local_clock_offset_ms =
ntp_estimator_.EstimateRemoteToLocalClockOffsetMs();
if (remote_to_local_clock_offset_ms.has_value()) {
absolute_capture_time_receiver_.SetRemoteToLocalClockOffset(
Int64MsToQ32x32(*remote_to_local_clock_offset_ms));
}
}
return true;
}
void RtpVideoStreamReceiver2::FrameContinuous(int64_t picture_id) {
RTC_DCHECK_RUN_ON(&worker_task_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(&worker_task_checker_);
// Running on the decoder thread.
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) {
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(&worker_task_checker_);
receiving_ = true;
}
void RtpVideoStreamReceiver2::StopReceive() {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
receiving_ = false;
}
void RtpVideoStreamReceiver2::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));
}
if (config_.rtp.ulpfec_payload_type != -1) {
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.FecBitrateReceivedInKbps",
static_cast<int>(counter.num_bytes * 8 / elapsed_sec / 1000));
}
}
void RtpVideoStreamReceiver2::InsertSpsPpsIntoTracker(uint8_t payload_type) {
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());
}
} // namespace webrtc