blob: 51441f6ab8d431d4510092bdcf2f852f44edab47 [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 "modules/rtp_rtcp/source/rtp_sender_video.h"
#include <stdlib.h>
#include <string.h>
#include <limits>
#include <memory>
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.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_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
constexpr size_t kRedForFecHeaderLength = 1;
constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
constexpr int64_t kMaxUnretransmittableFrameIntervalMs = 33 * 4;
// This is experimental field trial to exclude transport sequence number from
// FEC packets and should only be used in conjunction with datagram transport.
// Datagram transport removes transport sequence numbers from RTP packets and
// uses datagram feedback loop to re-generate RTCP feedback packets, but FEC
// contorol packets are calculated before sequence number is removed and as a
// result recovered packets will be corrupt unless we also remove transport
// sequence number during FEC calculation.
//
// TODO(sukhanov): We need to find find better way to implement FEC with
// datagram transport, probably moving FEC to datagram integration layter. We
// should also remove special field trial once we switch datagram path from
// RTCConfiguration flags to field trial and use the same field trial for FEC
// workaround.
const char kExcludeTransportSequenceNumberFromFecFieldTrial[] =
"WebRTC-ExcludeTransportSequenceNumberFromFec";
void BuildRedPayload(const RtpPacketToSend& media_packet,
RtpPacketToSend* red_packet) {
uint8_t* red_payload = red_packet->AllocatePayload(
kRedForFecHeaderLength + media_packet.payload_size());
RTC_DCHECK(red_payload);
red_payload[0] = media_packet.PayloadType();
auto media_payload = media_packet.payload();
memcpy(&red_payload[kRedForFecHeaderLength], media_payload.data(),
media_payload.size());
}
void AddRtpHeaderExtensions(const RTPVideoHeader& video_header,
const absl::optional<PlayoutDelay>& playout_delay,
VideoFrameType frame_type,
bool set_video_rotation,
bool set_color_space,
bool set_frame_marking,
bool first_packet,
bool last_packet,
RtpPacketToSend* packet) {
// Color space requires two-byte header extensions if HDR metadata is
// included. Therefore, it's best to add this extension first so that the
// other extensions in the same packet are written as two-byte headers at
// once.
if (last_packet && set_color_space && video_header.color_space)
packet->SetExtension<ColorSpaceExtension>(video_header.color_space.value());
if (last_packet && set_video_rotation)
packet->SetExtension<VideoOrientation>(video_header.rotation);
// Report content type only for key frames.
if (last_packet && frame_type == VideoFrameType::kVideoFrameKey &&
video_header.content_type != VideoContentType::UNSPECIFIED)
packet->SetExtension<VideoContentTypeExtension>(video_header.content_type);
if (last_packet &&
video_header.video_timing.flags != VideoSendTiming::kInvalid)
packet->SetExtension<VideoTimingExtension>(video_header.video_timing);
// If transmitted, add to all packets; ack logic depends on this.
if (playout_delay) {
packet->SetExtension<PlayoutDelayLimits>(*playout_delay);
}
if (set_frame_marking) {
FrameMarking frame_marking = video_header.frame_marking;
frame_marking.start_of_frame = first_packet;
frame_marking.end_of_frame = last_packet;
packet->SetExtension<FrameMarkingExtension>(frame_marking);
}
if (video_header.generic) {
RtpGenericFrameDescriptor generic_descriptor;
generic_descriptor.SetFirstPacketInSubFrame(first_packet);
generic_descriptor.SetLastPacketInSubFrame(last_packet);
generic_descriptor.SetDiscardable(video_header.generic->discardable);
if (first_packet) {
generic_descriptor.SetFrameId(
static_cast<uint16_t>(video_header.generic->frame_id));
for (int64_t dep : video_header.generic->dependencies) {
generic_descriptor.AddFrameDependencyDiff(
video_header.generic->frame_id - dep);
}
uint8_t spatial_bimask = 1 << video_header.generic->spatial_index;
for (int layer : video_header.generic->higher_spatial_layers) {
RTC_DCHECK_GT(layer, video_header.generic->spatial_index);
RTC_DCHECK_LT(layer, 8);
spatial_bimask |= 1 << layer;
}
generic_descriptor.SetSpatialLayersBitmask(spatial_bimask);
generic_descriptor.SetTemporalLayer(video_header.generic->temporal_index);
if (frame_type == VideoFrameType::kVideoFrameKey) {
generic_descriptor.SetResolution(video_header.width,
video_header.height);
}
}
if (!packet->SetExtension<RtpGenericFrameDescriptorExtension01>(
generic_descriptor)) {
packet->SetExtension<RtpGenericFrameDescriptorExtension00>(
generic_descriptor);
}
}
}
bool MinimizeDescriptor(const RTPVideoHeader& full, RTPVideoHeader* minimized) {
if (full.codec == VideoCodecType::kVideoCodecVP8) {
minimized->codec = VideoCodecType::kVideoCodecVP8;
const auto& vp8 = absl::get<RTPVideoHeaderVP8>(full.video_type_header);
// Set minimum fields the RtpPacketizer is using to create vp8 packets.
auto& min_vp8 = minimized->video_type_header.emplace<RTPVideoHeaderVP8>();
min_vp8.InitRTPVideoHeaderVP8();
min_vp8.nonReference = vp8.nonReference;
return true;
}
// TODO(danilchap): Reduce vp9 codec specific descriptor too.
return false;
}
bool IsBaseLayer(const RTPVideoHeader& video_header) {
switch (video_header.codec) {
case kVideoCodecVP8: {
const auto& vp8 =
absl::get<RTPVideoHeaderVP8>(video_header.video_type_header);
return (vp8.temporalIdx == 0 || vp8.temporalIdx == kNoTemporalIdx);
}
case kVideoCodecVP9: {
const auto& vp9 =
absl::get<RTPVideoHeaderVP9>(video_header.video_type_header);
return (vp9.temporal_idx == 0 || vp9.temporal_idx == kNoTemporalIdx);
}
case kVideoCodecH264:
// TODO(kron): Implement logic for H264 once WebRTC supports temporal
// layers for H264.
break;
default:
break;
}
return true;
}
const char* FrameTypeToString(VideoFrameType frame_type) {
switch (frame_type) {
case VideoFrameType::kEmptyFrame:
return "empty";
case VideoFrameType::kVideoFrameKey:
return "video_key";
case VideoFrameType::kVideoFrameDelta:
return "video_delta";
default:
RTC_NOTREACHED();
return "";
}
}
} // namespace
RTPSenderVideo::RTPSenderVideo(Clock* clock,
RTPSender* rtp_sender,
FlexfecSender* flexfec_sender,
PlayoutDelayOracle* playout_delay_oracle,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
bool need_rtp_packet_infos,
const WebRtcKeyValueConfig& field_trials)
: rtp_sender_(rtp_sender),
clock_(clock),
retransmission_settings_(kRetransmitBaseLayer |
kConditionallyRetransmitHigherLayers),
last_rotation_(kVideoRotation_0),
transmit_color_space_next_frame_(false),
playout_delay_oracle_(playout_delay_oracle),
rtp_sequence_number_map_(need_rtp_packet_infos
? absl::make_unique<RtpSequenceNumberMap>(
kRtpSequenceNumberMapMaxEntries)
: nullptr),
red_payload_type_(-1),
ulpfec_payload_type_(-1),
flexfec_sender_(flexfec_sender),
delta_fec_params_{0, 1, kFecMaskRandom},
key_fec_params_{0, 1, kFecMaskRandom},
fec_bitrate_(1000, RateStatistics::kBpsScale),
video_bitrate_(1000, RateStatistics::kBpsScale),
packetization_overhead_bitrate_(1000, RateStatistics::kBpsScale),
frame_encryptor_(frame_encryptor),
require_frame_encryption_(require_frame_encryption),
generic_descriptor_auth_experiment_(
field_trials.Lookup("WebRTC-GenericDescriptorAuth").find("Enabled") ==
0),
exclude_transport_sequence_number_from_fec_experiment_(
field_trials.Lookup(kExcludeTransportSequenceNumberFromFecFieldTrial)
.find("Enabled") == 0) {
RTC_DCHECK(playout_delay_oracle_);
}
RTPSenderVideo::~RTPSenderVideo() {}
void RTPSenderVideo::RegisterPayloadType(int8_t payload_type,
absl::string_view payload_name,
bool raw_payload) {
absl::optional<VideoCodecType> video_type;
if (!raw_payload) {
if (absl::EqualsIgnoreCase(payload_name, "VP8")) {
video_type = kVideoCodecVP8;
} else if (absl::EqualsIgnoreCase(payload_name, "VP9")) {
video_type = kVideoCodecVP9;
} else if (absl::EqualsIgnoreCase(payload_name, "H264")) {
video_type = kVideoCodecH264;
} else {
video_type = kVideoCodecGeneric;
}
}
{
rtc::CritScope cs(&payload_type_crit_);
payload_type_map_[payload_type] = video_type;
}
// Backward compatibility for older receivers without temporal layer logic
if (absl::EqualsIgnoreCase(payload_name, "H264")) {
rtc::CritScope cs(&crit_);
retransmission_settings_ = kRetransmitBaseLayer | kRetransmitHigherLayers;
}
}
void RTPSenderVideo::SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage) {
// Remember some values about the packet before sending it away.
size_t packet_size = packet->size();
uint16_t seq_num = packet->SequenceNumber();
packet->set_packet_type(RtpPacketToSend::Type::kVideo);
if (!LogAndSendToNetwork(std::move(packet), storage)) {
RTC_LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
return;
}
rtc::CritScope cs(&stats_crit_);
video_bitrate_.Update(packet_size, clock_->TimeInMilliseconds());
}
void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec(
std::unique_ptr<RtpPacketToSend> media_packet,
StorageType media_packet_storage,
bool protect_media_packet) {
uint16_t media_seq_num = media_packet->SequenceNumber();
std::unique_ptr<RtpPacketToSend> red_packet(
new RtpPacketToSend(*media_packet));
BuildRedPayload(*media_packet, red_packet.get());
std::vector<std::unique_ptr<RedPacket>> fec_packets;
{
// Only protect while creating RED and FEC packets, not when sending.
rtc::CritScope cs(&crit_);
red_packet->SetPayloadType(red_payload_type_);
if (ulpfec_enabled()) {
if (protect_media_packet) {
if (exclude_transport_sequence_number_from_fec_experiment_) {
// See comments at the top of the file why experiment
// "WebRTC-kExcludeTransportSequenceNumberFromFec" is needed in
// conjunction with datagram transport.
// TODO(sukhanov): We may also need to implement it for flexfec_sender
// if we decide to keep this approach in the future.
uint16_t transport_senquence_number;
if (media_packet->GetExtension<webrtc::TransportSequenceNumber>(
&transport_senquence_number)) {
if (!media_packet->RemoveExtension(
webrtc::TransportSequenceNumber::kId)) {
RTC_NOTREACHED()
<< "Failed to remove transport sequence number, packet="
<< media_packet->ToString();
}
}
}
ulpfec_generator_.AddRtpPacketAndGenerateFec(
media_packet->data(), media_packet->payload_size(),
media_packet->headers_size());
}
uint16_t num_fec_packets = ulpfec_generator_.NumAvailableFecPackets();
if (num_fec_packets > 0) {
uint16_t first_fec_sequence_number =
rtp_sender_->AllocateSequenceNumber(num_fec_packets);
fec_packets = ulpfec_generator_.GetUlpfecPacketsAsRed(
red_payload_type_, ulpfec_payload_type_, first_fec_sequence_number);
RTC_DCHECK_EQ(num_fec_packets, fec_packets.size());
}
}
}
// Send |red_packet| instead of |packet| for allocated sequence number.
size_t red_packet_size = red_packet->size();
red_packet->set_packet_type(RtpPacketToSend::Type::kVideo);
if (LogAndSendToNetwork(std::move(red_packet), media_packet_storage)) {
rtc::CritScope cs(&stats_crit_);
video_bitrate_.Update(red_packet_size, clock_->TimeInMilliseconds());
} else {
RTC_LOG(LS_WARNING) << "Failed to send RED packet " << media_seq_num;
}
for (const auto& fec_packet : fec_packets) {
// TODO(danilchap): Make ulpfec_generator_ generate RtpPacketToSend to avoid
// reparsing them.
std::unique_ptr<RtpPacketToSend> rtp_packet(
new RtpPacketToSend(*media_packet));
RTC_CHECK(rtp_packet->Parse(fec_packet->data(), fec_packet->length()));
rtp_packet->set_capture_time_ms(media_packet->capture_time_ms());
rtp_packet->set_packet_type(RtpPacketToSend::Type::kForwardErrorCorrection);
uint16_t fec_sequence_number = rtp_packet->SequenceNumber();
if (LogAndSendToNetwork(std::move(rtp_packet), kDontRetransmit)) {
rtc::CritScope cs(&stats_crit_);
fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds());
} else {
RTC_LOG(LS_WARNING) << "Failed to send ULPFEC packet "
<< fec_sequence_number;
}
}
}
void RTPSenderVideo::SendVideoPacketWithFlexfec(
std::unique_ptr<RtpPacketToSend> media_packet,
StorageType media_packet_storage,
bool protect_media_packet) {
RTC_DCHECK(flexfec_sender_);
if (protect_media_packet)
flexfec_sender_->AddRtpPacketAndGenerateFec(*media_packet);
SendVideoPacket(std::move(media_packet), media_packet_storage);
if (flexfec_sender_->FecAvailable()) {
std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
flexfec_sender_->GetFecPackets();
for (auto& fec_packet : fec_packets) {
size_t packet_length = fec_packet->size();
uint16_t seq_num = fec_packet->SequenceNumber();
fec_packet->set_packet_type(
RtpPacketToSend::Type::kForwardErrorCorrection);
if (LogAndSendToNetwork(std::move(fec_packet), kDontRetransmit)) {
rtc::CritScope cs(&stats_crit_);
fec_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());
} else {
RTC_LOG(LS_WARNING) << "Failed to send FlexFEC packet " << seq_num;
}
}
}
}
bool RTPSenderVideo::LogAndSendToNetwork(
std::unique_ptr<RtpPacketToSend> packet,
StorageType storage) {
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
int64_t now_ms = clock_->TimeInMilliseconds();
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
rtp_sender_->ActualSendBitrateKbit(),
packet->Ssrc());
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
FecOverheadRate() / 1000, packet->Ssrc());
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
rtp_sender_->NackOverheadRate() / 1000,
packet->Ssrc());
#endif
return rtp_sender_->SendToNetwork(std::move(packet), storage);
}
void RTPSenderVideo::SetUlpfecConfig(int red_payload_type,
int ulpfec_payload_type) {
// Sanity check. Per the definition of UlpfecConfig (see config.h),
// a payload type of -1 means that the corresponding feature is
// turned off.
RTC_DCHECK_GE(red_payload_type, -1);
RTC_DCHECK_LE(red_payload_type, 127);
RTC_DCHECK_GE(ulpfec_payload_type, -1);
RTC_DCHECK_LE(ulpfec_payload_type, 127);
rtc::CritScope cs(&crit_);
red_payload_type_ = red_payload_type;
ulpfec_payload_type_ = ulpfec_payload_type;
// Must not enable ULPFEC without RED.
RTC_DCHECK(!(red_enabled() ^ ulpfec_enabled()));
// Reset FEC parameters.
delta_fec_params_ = FecProtectionParams{0, 1, kFecMaskRandom};
key_fec_params_ = FecProtectionParams{0, 1, kFecMaskRandom};
}
size_t RTPSenderVideo::CalculateFecPacketOverhead() const {
if (flexfec_enabled())
return flexfec_sender_->MaxPacketOverhead();
size_t overhead = 0;
if (red_enabled()) {
// The RED overhead is due to a small header.
overhead += kRedForFecHeaderLength;
}
if (ulpfec_enabled()) {
// For ULPFEC, the overhead is the FEC headers plus RED for FEC header
// (see above) plus anything in RTP header beyond the 12 bytes base header
// (CSRC list, extensions...)
// This reason for the header extensions to be included here is that
// from an FEC viewpoint, they are part of the payload to be protected.
// (The base RTP header is already protected by the FEC header.)
overhead += ulpfec_generator_.MaxPacketOverhead() +
(rtp_sender_->RtpHeaderLength() - kRtpHeaderSize);
}
return overhead;
}
void RTPSenderVideo::SetFecParameters(const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) {
rtc::CritScope cs(&crit_);
delta_fec_params_ = delta_params;
key_fec_params_ = key_params;
}
absl::optional<uint32_t> RTPSenderVideo::FlexfecSsrc() const {
if (flexfec_sender_) {
return flexfec_sender_->ssrc();
}
return absl::nullopt;
}
bool RTPSenderVideo::SendVideo(
VideoFrameType frame_type,
int8_t payload_type,
uint32_t rtp_timestamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* video_header,
absl::optional<int64_t> expected_retransmission_time_ms) {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
FrameTypeToString(frame_type));
if (frame_type == VideoFrameType::kEmptyFrame)
return true;
if (payload_size == 0)
return false;
RTC_CHECK(video_header);
size_t fec_packet_overhead;
bool red_enabled;
int32_t retransmission_settings;
bool set_video_rotation;
bool set_color_space = false;
bool set_frame_marking =
video_header->codec == kVideoCodecH264 &&
video_header->frame_marking.temporal_id != kNoTemporalIdx;
const absl::optional<PlayoutDelay> playout_delay =
playout_delay_oracle_->PlayoutDelayToSend(video_header->playout_delay);
{
rtc::CritScope cs(&crit_);
// According to
// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
// ts_126114v120700p.pdf Section 7.4.5:
// The MTSI client shall add the payload bytes as defined in this clause
// onto the last RTP packet in each group of packets which make up a key
// frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
// (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
// packet in each group of packets which make up another type of frame
// (e.g. a P-Frame) only if the current value is different from the previous
// value sent.
// Set rotation when key frame or when changed (to follow standard).
// Or when different from 0 (to follow current receiver implementation).
set_video_rotation = frame_type == VideoFrameType::kVideoFrameKey ||
video_header->rotation != last_rotation_ ||
video_header->rotation != kVideoRotation_0;
last_rotation_ = video_header->rotation;
// Send color space when changed or if the frame is a key frame. Keep
// sending color space information until the first base layer frame to
// guarantee that the information is retrieved by the receiver.
if (video_header->color_space != last_color_space_) {
last_color_space_ = video_header->color_space;
set_color_space = true;
transmit_color_space_next_frame_ = !IsBaseLayer(*video_header);
} else {
set_color_space = frame_type == VideoFrameType::kVideoFrameKey ||
transmit_color_space_next_frame_;
transmit_color_space_next_frame_ = transmit_color_space_next_frame_
? !IsBaseLayer(*video_header)
: false;
}
// FEC settings.
const FecProtectionParams& fec_params =
frame_type == VideoFrameType::kVideoFrameKey ? key_fec_params_
: delta_fec_params_;
if (flexfec_enabled())
flexfec_sender_->SetFecParameters(fec_params);
if (ulpfec_enabled())
ulpfec_generator_.SetFecParameters(fec_params);
fec_packet_overhead = CalculateFecPacketOverhead();
red_enabled = this->red_enabled();
retransmission_settings = retransmission_settings_;
}
// Maximum size of packet including rtp headers.
// Extra space left in case packet will be resent using fec or rtx.
int packet_capacity = rtp_sender_->MaxRtpPacketSize() - fec_packet_overhead -
(rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0);
std::unique_ptr<RtpPacketToSend> single_packet =
rtp_sender_->AllocatePacket();
RTC_DCHECK_LE(packet_capacity, single_packet->capacity());
single_packet->SetPayloadType(payload_type);
single_packet->SetTimestamp(rtp_timestamp);
single_packet->set_capture_time_ms(capture_time_ms);
auto first_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
auto middle_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
auto last_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
// Simplest way to estimate how much extensions would occupy is to set them.
AddRtpHeaderExtensions(*video_header, playout_delay, frame_type,
set_video_rotation, set_color_space, set_frame_marking,
/*first=*/true, /*last=*/true, single_packet.get());
AddRtpHeaderExtensions(*video_header, playout_delay, frame_type,
set_video_rotation, set_color_space, set_frame_marking,
/*first=*/true, /*last=*/false, first_packet.get());
AddRtpHeaderExtensions(*video_header, playout_delay, frame_type,
set_video_rotation, set_color_space, set_frame_marking,
/*first=*/false, /*last=*/false, middle_packet.get());
AddRtpHeaderExtensions(*video_header, playout_delay, frame_type,
set_video_rotation, set_color_space, set_frame_marking,
/*first=*/false, /*last=*/true, last_packet.get());
RTC_DCHECK_GT(packet_capacity, single_packet->headers_size());
RTC_DCHECK_GT(packet_capacity, first_packet->headers_size());
RTC_DCHECK_GT(packet_capacity, middle_packet->headers_size());
RTC_DCHECK_GT(packet_capacity, last_packet->headers_size());
RtpPacketizer::PayloadSizeLimits limits;
limits.max_payload_len = packet_capacity - middle_packet->headers_size();
RTC_DCHECK_GE(single_packet->headers_size(), middle_packet->headers_size());
limits.single_packet_reduction_len =
single_packet->headers_size() - middle_packet->headers_size();
RTC_DCHECK_GE(first_packet->headers_size(), middle_packet->headers_size());
limits.first_packet_reduction_len =
first_packet->headers_size() - middle_packet->headers_size();
RTC_DCHECK_GE(last_packet->headers_size(), middle_packet->headers_size());
limits.last_packet_reduction_len =
last_packet->headers_size() - middle_packet->headers_size();
RTPVideoHeader minimized_video_header;
const RTPVideoHeader* packetize_video_header = video_header;
rtc::ArrayView<const uint8_t> generic_descriptor_raw_00 =
first_packet->GetRawExtension<RtpGenericFrameDescriptorExtension00>();
rtc::ArrayView<const uint8_t> generic_descriptor_raw_01 =
first_packet->GetRawExtension<RtpGenericFrameDescriptorExtension01>();
if (!generic_descriptor_raw_00.empty() &&
!generic_descriptor_raw_01.empty()) {
RTC_LOG(LS_WARNING) << "Two versions of GFD extension used.";
return false;
}
rtc::ArrayView<const uint8_t> generic_descriptor_raw =
!generic_descriptor_raw_01.empty() ? generic_descriptor_raw_01
: generic_descriptor_raw_00;
if (!generic_descriptor_raw.empty()) {
if (MinimizeDescriptor(*video_header, &minimized_video_header)) {
packetize_video_header = &minimized_video_header;
}
}
// TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
rtc::Buffer encrypted_video_payload;
if (frame_encryptor_ != nullptr) {
if (generic_descriptor_raw.empty()) {
return false;
}
const size_t max_ciphertext_size =
frame_encryptor_->GetMaxCiphertextByteSize(cricket::MEDIA_TYPE_VIDEO,
payload_size);
encrypted_video_payload.SetSize(max_ciphertext_size);
size_t bytes_written = 0;
// Only enable header authentication if the field trial is enabled.
rtc::ArrayView<const uint8_t> additional_data;
if (generic_descriptor_auth_experiment_) {
additional_data = generic_descriptor_raw;
}
if (frame_encryptor_->Encrypt(
cricket::MEDIA_TYPE_VIDEO, first_packet->Ssrc(), additional_data,
rtc::MakeArrayView(payload_data, payload_size),
encrypted_video_payload, &bytes_written) != 0) {
return false;
}
encrypted_video_payload.SetSize(bytes_written);
payload_data = encrypted_video_payload.data();
payload_size = encrypted_video_payload.size();
} else if (require_frame_encryption_) {
RTC_LOG(LS_WARNING)
<< "No FrameEncryptor is attached to this video sending stream but "
<< "one is required since require_frame_encryptor is set";
}
absl::optional<VideoCodecType> type;
{
rtc::CritScope cs(&payload_type_crit_);
const auto it = payload_type_map_.find(payload_type);
if (it == payload_type_map_.end()) {
RTC_LOG(LS_ERROR) << "Payload type " << static_cast<int>(payload_type)
<< " not registered.";
return false;
}
type = it->second;
}
std::unique_ptr<RtpPacketizer> packetizer = RtpPacketizer::Create(
type, rtc::MakeArrayView(payload_data, payload_size), limits,
*packetize_video_header, frame_type, fragmentation);
const uint8_t temporal_id = GetTemporalId(*video_header);
// TODO(bugs.webrtc.org/10714): retransmission_settings_ should generally be
// replaced by expected_retransmission_time_ms.has_value(). For now, though,
// only VP8 with an injected frame buffer controller actually controls it.
const StorageType storage =
expected_retransmission_time_ms.has_value()
? GetStorageType(temporal_id, retransmission_settings,
expected_retransmission_time_ms.value())
: StorageType::kDontRetransmit;
const size_t num_packets = packetizer->NumPackets();
size_t unpacketized_payload_size;
if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
unpacketized_payload_size = 0;
for (uint16_t i = 0; i < fragmentation->fragmentationVectorSize; ++i) {
unpacketized_payload_size += fragmentation->fragmentationLength[i];
}
} else {
unpacketized_payload_size = payload_size;
}
size_t packetized_payload_size = 0;
if (num_packets == 0)
return false;
uint16_t first_sequence_number;
bool first_frame = first_frame_sent_();
for (size_t i = 0; i < num_packets; ++i) {
std::unique_ptr<RtpPacketToSend> packet;
int expected_payload_capacity;
// Choose right packet template:
if (num_packets == 1) {
packet = std::move(single_packet);
expected_payload_capacity =
limits.max_payload_len - limits.single_packet_reduction_len;
} else if (i == 0) {
packet = std::move(first_packet);
expected_payload_capacity =
limits.max_payload_len - limits.first_packet_reduction_len;
} else if (i == num_packets - 1) {
packet = std::move(last_packet);
expected_payload_capacity =
limits.max_payload_len - limits.last_packet_reduction_len;
} else {
packet = absl::make_unique<RtpPacketToSend>(*middle_packet);
expected_payload_capacity = limits.max_payload_len;
}
if (!packetizer->NextPacket(packet.get()))
return false;
RTC_DCHECK_LE(packet->payload_size(), expected_payload_capacity);
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
return false;
packetized_payload_size += packet->payload_size();
if (rtp_sequence_number_map_ && i == 0) {
first_sequence_number = packet->SequenceNumber();
}
if (i == 0) {
playout_delay_oracle_->OnSentPacket(packet->SequenceNumber(),
playout_delay);
}
// No FEC protection for upper temporal layers, if used.
bool protect_packet = temporal_id == 0 || temporal_id == kNoTemporalIdx;
// Put packetization finish timestamp into extension.
if (packet->HasExtension<VideoTimingExtension>()) {
packet->set_packetization_finish_time_ms(clock_->TimeInMilliseconds());
// TODO(webrtc:10750): wait a couple of months and remove the statement
// below. For now we can't use packets with VideoTimingFrame extensions in
// Fec because the extension is modified after FEC is calculated by pacer
// and network. This may cause corruptions in video payload and header.
// The fix in receive code is implemented, but until all the receivers
// are updated, senders can't send potentially breaking packets.
protect_packet = false;
}
if (flexfec_enabled()) {
// TODO(brandtr): Remove the FlexFEC code path when FlexfecSender
// is wired up to PacedSender instead.
SendVideoPacketWithFlexfec(std::move(packet), storage, protect_packet);
} else if (red_enabled) {
SendVideoPacketAsRedMaybeWithUlpfec(std::move(packet), storage,
protect_packet);
} else {
SendVideoPacket(std::move(packet), storage);
}
if (first_frame) {
if (i == 0) {
RTC_LOG(LS_INFO)
<< "Sent first RTP packet of the first video frame (pre-pacer)";
}
if (i == num_packets - 1) {
RTC_LOG(LS_INFO)
<< "Sent last RTP packet of the first video frame (pre-pacer)";
}
}
}
if (rtp_sequence_number_map_) {
const uint32_t timestamp = rtp_timestamp - rtp_sender_->TimestampOffset();
rtc::CritScope cs(&crit_);
rtp_sequence_number_map_->InsertFrame(first_sequence_number, num_packets,
timestamp);
}
rtc::CritScope cs(&stats_crit_);
RTC_DCHECK_GE(packetized_payload_size, unpacketized_payload_size);
packetization_overhead_bitrate_.Update(
packetized_payload_size - unpacketized_payload_size,
clock_->TimeInMilliseconds());
TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
rtp_timestamp);
return true;
}
uint32_t RTPSenderVideo::VideoBitrateSent() const {
rtc::CritScope cs(&stats_crit_);
return video_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
uint32_t RTPSenderVideo::FecOverheadRate() const {
rtc::CritScope cs(&stats_crit_);
return fec_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
uint32_t RTPSenderVideo::PacketizationOverheadBps() const {
rtc::CritScope cs(&stats_crit_);
return packetization_overhead_bitrate_.Rate(clock_->TimeInMilliseconds())
.value_or(0);
}
std::vector<RtpSequenceNumberMap::Info> RTPSenderVideo::GetSentRtpPacketInfos(
rtc::ArrayView<const uint16_t> sequence_numbers) const {
RTC_DCHECK(!sequence_numbers.empty());
std::vector<RtpSequenceNumberMap::Info> results;
if (!rtp_sequence_number_map_) {
return results;
}
results.reserve(sequence_numbers.size());
{
rtc::CritScope cs(&crit_);
for (uint16_t sequence_number : sequence_numbers) {
const absl::optional<RtpSequenceNumberMap::Info> info =
rtp_sequence_number_map_->Get(sequence_number);
if (!info) {
// The empty vector will be returned. We can delay the clearing
// of the vector until after we exit the critical section.
break;
}
results.push_back(*info);
}
}
if (results.size() != sequence_numbers.size()) {
results.clear(); // Some sequence number was not found.
}
return results;
}
StorageType RTPSenderVideo::GetStorageType(
uint8_t temporal_id,
int32_t retransmission_settings,
int64_t expected_retransmission_time_ms) {
if (retransmission_settings == kRetransmitOff)
return StorageType::kDontRetransmit;
rtc::CritScope cs(&stats_crit_);
// Media packet storage.
if ((retransmission_settings & kConditionallyRetransmitHigherLayers) &&
UpdateConditionalRetransmit(temporal_id,
expected_retransmission_time_ms)) {
retransmission_settings |= kRetransmitHigherLayers;
}
if (temporal_id == kNoTemporalIdx)
return kAllowRetransmission;
if ((retransmission_settings & kRetransmitBaseLayer) && temporal_id == 0)
return kAllowRetransmission;
if ((retransmission_settings & kRetransmitHigherLayers) && temporal_id > 0)
return kAllowRetransmission;
return kDontRetransmit;
}
uint8_t RTPSenderVideo::GetTemporalId(const RTPVideoHeader& header) {
struct TemporalIdGetter {
uint8_t operator()(const RTPVideoHeaderVP8& vp8) { return vp8.temporalIdx; }
uint8_t operator()(const RTPVideoHeaderVP9& vp9) {
return vp9.temporal_idx;
}
uint8_t operator()(const RTPVideoHeaderH264&) { return kNoTemporalIdx; }
uint8_t operator()(const absl::monostate&) { return kNoTemporalIdx; }
};
switch (header.codec) {
case kVideoCodecH264:
return header.frame_marking.temporal_id;
default:
return absl::visit(TemporalIdGetter(), header.video_type_header);
}
}
bool RTPSenderVideo::UpdateConditionalRetransmit(
uint8_t temporal_id,
int64_t expected_retransmission_time_ms) {
int64_t now_ms = clock_->TimeInMilliseconds();
// Update stats for any temporal layer.
TemporalLayerStats* current_layer_stats =
&frame_stats_by_temporal_layer_[temporal_id];
current_layer_stats->frame_rate_fp1000s.Update(1, now_ms);
int64_t tl_frame_interval = now_ms - current_layer_stats->last_frame_time_ms;
current_layer_stats->last_frame_time_ms = now_ms;
// Conditional retransmit only applies to upper layers.
if (temporal_id != kNoTemporalIdx && temporal_id > 0) {
if (tl_frame_interval >= kMaxUnretransmittableFrameIntervalMs) {
// Too long since a retransmittable frame in this layer, enable NACK
// protection.
return true;
} else {
// Estimate when the next frame of any lower layer will be sent.
const int64_t kUndefined = std::numeric_limits<int64_t>::max();
int64_t expected_next_frame_time = kUndefined;
for (int i = temporal_id - 1; i >= 0; --i) {
TemporalLayerStats* stats = &frame_stats_by_temporal_layer_[i];
absl::optional<uint32_t> rate = stats->frame_rate_fp1000s.Rate(now_ms);
if (rate) {
int64_t tl_next = stats->last_frame_time_ms + 1000000 / *rate;
if (tl_next - now_ms > -expected_retransmission_time_ms &&
tl_next < expected_next_frame_time) {
expected_next_frame_time = tl_next;
}
}
}
if (expected_next_frame_time == kUndefined ||
expected_next_frame_time - now_ms > expected_retransmission_time_ms) {
// The next frame in a lower layer is expected at a later time (or
// unable to tell due to lack of data) than a retransmission is
// estimated to be able to arrive, so allow this packet to be nacked.
return true;
}
}
}
return false;
}
} // namespace webrtc