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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 "modules/rtp_rtcp/source/rtp_sender_video.h"
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#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 "api/transport/rtp/dependency_descriptor.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/absolute_capture_time_sender.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.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 "modules/rtp_rtcp/source/time_util.h"
#include "modules/rtp_rtcp/source/transformable_encoded_frame.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 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 a 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());
}
bool MinimizeDescriptor(RTPVideoHeader* video_header) {
if (auto* vp8 =
absl::get_if<RTPVideoHeaderVP8>(&video_header->video_type_header)) {
// Set minimum fields the RtpPacketizer is using to create vp8 packets.
// nonReference is the only field that doesn't require extra space.
bool non_reference = vp8->nonReference;
vp8->InitRTPVideoHeaderVP8();
vp8->nonReference = non_reference;
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;
}
#if RTC_TRACE_EVENTS_ENABLED
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 "";
}
}
#endif
bool IsNoopDelay(const PlayoutDelay& delay) {
return delay.min_ms == -1 && delay.max_ms == -1;
}
} // namespace
RTPSenderVideo::RTPSenderVideo(const Config& config)
: rtp_sender_(config.rtp_sender),
clock_(config.clock),
retransmission_settings_(
config.enable_retransmit_all_layers
? kRetransmitAllLayers
: (kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers)),
last_rotation_(kVideoRotation_0),
transmit_color_space_next_frame_(false),
current_playout_delay_{-1, -1},
playout_delay_pending_(false),
red_payload_type_(config.red_payload_type),
fec_generator_(config.fec_generator),
video_bitrate_(1000, RateStatistics::kBpsScale),
packetization_overhead_bitrate_(1000, RateStatistics::kBpsScale),
frame_encryptor_(config.frame_encryptor),
require_frame_encryption_(config.require_frame_encryption),
generic_descriptor_auth_experiment_(
config.field_trials->Lookup("WebRTC-GenericDescriptorAuth")
.find("Disabled") != 0),
exclude_transport_sequence_number_from_fec_experiment_(
config.field_trials
->Lookup(kExcludeTransportSequenceNumberFromFecFieldTrial)
.find("Enabled") == 0),
absolute_capture_time_sender_(config.clock),
frame_transformer_delegate_(
config.frame_transformer
? new rtc::RefCountedObject<
RTPSenderVideoFrameTransformerDelegate>(
this,
std::move(config.frame_transformer))
: nullptr) {
if (frame_transformer_delegate_)
frame_transformer_delegate_->Init();
}
RTPSenderVideo::~RTPSenderVideo() {
if (frame_transformer_delegate_)
frame_transformer_delegate_->Reset();
}
void RTPSenderVideo::LogAndSendToNetwork(
std::vector<std::unique_ptr<RtpPacketToSend>> packets,
size_t unpacketized_payload_size) {
int64_t now_ms = clock_->TimeInMilliseconds();
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
for (const auto& packet : packets) {
if (packet->packet_type() == RtpPacketMediaType::kForwardErrorCorrection) {
const uint32_t ssrc = packet->Ssrc();
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
FecOverheadRate() / 1000, ssrc);
}
}
#endif
{
rtc::CritScope cs(&stats_crit_);
size_t packetized_payload_size = 0;
for (const auto& packet : packets) {
if (*packet->packet_type() == RtpPacketMediaType::kVideo) {
video_bitrate_.Update(packet->size(), now_ms);
packetized_payload_size += packet->payload_size();
}
}
// AV1 packetizer may produce less packetized bytes than unpacketized.
if (packetized_payload_size >= unpacketized_payload_size) {
packetization_overhead_bitrate_.Update(
packetized_payload_size - unpacketized_payload_size,
clock_->TimeInMilliseconds());
}
}
rtp_sender_->EnqueuePackets(std::move(packets));
}
size_t RTPSenderVideo::FecPacketOverhead() const {
size_t overhead = fec_generator_ ? fec_generator_->MaxPacketOverhead() : 0u;
if (red_enabled()) {
// The RED overhead is due to a small header.
overhead += kRedForFecHeaderLength;
// TODO(bugs.webrtc.org/11340): Move this into UlpfecGenerator.
if (fec_generator_ &&
fec_generator_->GetFecType() == VideoFecGenerator::FecType::kUlpFec) {
// 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 += rtp_sender_->RtpHeaderLength() - kRtpHeaderSize;
}
}
return overhead;
}
void RTPSenderVideo::SetFecParameters(const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) {
if (fec_generator_) {
fec_generator_->SetProtectionParameters(delta_params, key_params);
}
}
void RTPSenderVideo::SetVideoStructure(
const FrameDependencyStructure* video_structure) {
if (frame_transformer_delegate_) {
frame_transformer_delegate_->SetVideoStructureUnderLock(video_structure);
return;
}
// Lock is being held by SetVideoStructure() caller.
SetVideoStructureUnderLock(video_structure);
}
void RTPSenderVideo::SetVideoStructureUnderLock(
const FrameDependencyStructure* video_structure) {
RTC_DCHECK_RUNS_SERIALIZED(&send_checker_);
if (video_structure == nullptr) {
video_structure_ = nullptr;
return;
}
// Simple sanity checks video structure is set up.
RTC_DCHECK_GT(video_structure->num_decode_targets, 0);
RTC_DCHECK_GT(video_structure->templates.size(), 0);
int structure_id = 0;
if (video_structure_) {
if (*video_structure_ == *video_structure) {
// Same structure (just a new key frame), no update required.
return;
}
// When setting different video structure make sure structure_id is updated
// so that templates from different structures do not collide.
static constexpr int kMaxTemplates = 64;
structure_id =
(video_structure_->structure_id + video_structure_->templates.size()) %
kMaxTemplates;
}
video_structure_ =
std::make_unique<FrameDependencyStructure>(*video_structure);
video_structure_->structure_id = structure_id;
// TODO(bugs.webrtc.org/10342): Support chains.
video_structure_->num_chains = 0;
}
void RTPSenderVideo::AddRtpHeaderExtensions(
const RTPVideoHeader& video_header,
const absl::optional<AbsoluteCaptureTime>& absolute_capture_time,
bool first_packet,
bool last_packet,
RtpPacketToSend* packet) const {
// 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.
bool set_color_space =
video_header.color_space != last_color_space_ ||
video_header.frame_type == VideoFrameType::kVideoFrameKey ||
transmit_color_space_next_frame_;
// 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());
// 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).
bool set_video_rotation =
video_header.frame_type == VideoFrameType::kVideoFrameKey ||
video_header.rotation != last_rotation_ ||
video_header.rotation != kVideoRotation_0;
if (last_packet && set_video_rotation)
packet->SetExtension<VideoOrientation>(video_header.rotation);
// Report content type only for key frames.
if (last_packet &&
video_header.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_pending_) {
packet->SetExtension<PlayoutDelayLimits>(current_playout_delay_);
}
if (first_packet && absolute_capture_time) {
packet->SetExtension<AbsoluteCaptureTimeExtension>(*absolute_capture_time);
}
if (video_header.codec == kVideoCodecH264 &&
video_header.frame_marking.temporal_id != kNoTemporalIdx) {
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) {
bool extension_is_set = false;
if (video_structure_ != nullptr) {
DependencyDescriptor descriptor;
descriptor.first_packet_in_frame = first_packet;
descriptor.last_packet_in_frame = last_packet;
descriptor.frame_number = video_header.generic->frame_id & 0xFFFF;
descriptor.frame_dependencies.spatial_id =
video_header.generic->spatial_index;
descriptor.frame_dependencies.temporal_id =
video_header.generic->temporal_index;
for (int64_t dep : video_header.generic->dependencies) {
descriptor.frame_dependencies.frame_diffs.push_back(
video_header.generic->frame_id - dep);
}
descriptor.frame_dependencies.decode_target_indications =
video_header.generic->decode_target_indications;
RTC_DCHECK_EQ(
descriptor.frame_dependencies.decode_target_indications.size(),
video_structure_->num_decode_targets);
// To avoid extra structure copy, temporary share ownership of the
// video_structure with the dependency descriptor.
if (video_header.frame_type == VideoFrameType::kVideoFrameKey &&
first_packet) {
descriptor.attached_structure =
absl::WrapUnique(video_structure_.get());
}
extension_is_set = packet->SetExtension<RtpDependencyDescriptorExtension>(
*video_structure_, descriptor);
// Remove the temporary shared ownership.
descriptor.attached_structure.release();
}
// Do not use v0/v1 generic frame descriptor when v2 is stored.
if (!extension_is_set) {
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;
generic_descriptor.SetSpatialLayersBitmask(spatial_bimask);
generic_descriptor.SetTemporalLayer(
video_header.generic->temporal_index);
if (video_header.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 RTPSenderVideo::SendVideo(
int payload_type,
absl::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
int64_t capture_time_ms,
rtc::ArrayView<const uint8_t> payload,
const RTPFragmentationHeader* fragmentation,
RTPVideoHeader video_header,
absl::optional<int64_t> expected_retransmission_time_ms) {
#if RTC_TRACE_EVENTS_ENABLED
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
FrameTypeToString(video_header.frame_type));
#endif
RTC_CHECK_RUNS_SERIALIZED(&send_checker_);
if (video_header.frame_type == VideoFrameType::kEmptyFrame)
return true;
if (payload.empty())
return false;
int32_t retransmission_settings = retransmission_settings_;
if (codec_type == VideoCodecType::kVideoCodecH264) {
// Backward compatibility for older receivers without temporal layer logic.
retransmission_settings = kRetransmitBaseLayer | kRetransmitHigherLayers;
}
MaybeUpdateCurrentPlayoutDelay(video_header);
if (video_header.frame_type == VideoFrameType::kVideoFrameKey &&
!IsNoopDelay(current_playout_delay_)) {
// Force playout delay on key-frames, if set.
playout_delay_pending_ = true;
}
// 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() - FecPacketOverhead() -
(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);
const absl::optional<AbsoluteCaptureTime> absolute_capture_time =
absolute_capture_time_sender_.OnSendPacket(
AbsoluteCaptureTimeSender::GetSource(single_packet->Ssrc(),
single_packet->Csrcs()),
single_packet->Timestamp(), kVideoPayloadTypeFrequency,
Int64MsToUQ32x32(single_packet->capture_time_ms() + NtpOffsetMs()),
/*estimated_capture_clock_offset=*/absl::nullopt);
auto first_packet = std::make_unique<RtpPacketToSend>(*single_packet);
auto middle_packet = std::make_unique<RtpPacketToSend>(*single_packet);
auto last_packet = std::make_unique<RtpPacketToSend>(*single_packet);
// Simplest way to estimate how much extensions would occupy is to set them.
AddRtpHeaderExtensions(video_header, absolute_capture_time,
/*first_packet=*/true, /*last_packet=*/true,
single_packet.get());
AddRtpHeaderExtensions(video_header, absolute_capture_time,
/*first_packet=*/true, /*last_packet=*/false,
first_packet.get());
AddRtpHeaderExtensions(video_header, absolute_capture_time,
/*first_packet=*/false, /*last_packet=*/false,
middle_packet.get());
AddRtpHeaderExtensions(video_header, absolute_capture_time,
/*first_packet=*/false, /*last_packet=*/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();
bool has_generic_descriptor_00 =
first_packet->HasExtension<RtpGenericFrameDescriptorExtension00>();
bool has_generic_descriptor_01 =
first_packet->HasExtension<RtpGenericFrameDescriptorExtension01>();
bool has_dependency_descriptor =
first_packet->HasExtension<RtpDependencyDescriptorExtension>();
if (has_generic_descriptor_00 && has_generic_descriptor_01) {
RTC_LOG(LS_WARNING) << "Two versions of GFD extension used.";
return false;
}
// Minimization of the vp8 descriptor may erase temporal_id, so save it.
const uint8_t temporal_id = GetTemporalId(video_header);
bool has_generic_descriptor = has_generic_descriptor_00 ||
has_generic_descriptor_01 ||
has_dependency_descriptor;
if (has_generic_descriptor) {
MinimizeDescriptor(&video_header);
}
// TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
rtc::Buffer encrypted_video_payload;
if (frame_encryptor_ != nullptr) {
if (!has_generic_descriptor) {
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;
// Enable header authentication if the field trial isn't disabled.
std::vector<uint8_t> additional_data;
if (generic_descriptor_auth_experiment_) {
additional_data = RtpDescriptorAuthentication(video_header);
}
if (frame_encryptor_->Encrypt(
cricket::MEDIA_TYPE_VIDEO, first_packet->Ssrc(), additional_data,
payload, encrypted_video_payload, &bytes_written) != 0) {
return false;
}
encrypted_video_payload.SetSize(bytes_written);
payload = encrypted_video_payload;
} 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";
}
std::unique_ptr<RtpPacketizer> packetizer = RtpPacketizer::Create(
codec_type, payload, limits, video_header, fragmentation);
// 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 bool allow_retransmission =
expected_retransmission_time_ms.has_value()
? AllowRetransmission(temporal_id, retransmission_settings,
expected_retransmission_time_ms.value())
: false;
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();
}
if (num_packets == 0)
return false;
bool first_frame = first_frame_sent_();
std::vector<std::unique_ptr<RtpPacketToSend>> rtp_packets;
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 = std::make_unique<RtpPacketToSend>(*middle_packet);
expected_payload_capacity = limits.max_payload_len;
}
packet->set_first_packet_of_frame(i == 0);
if (!packetizer->NextPacket(packet.get()))
return false;
RTC_DCHECK_LE(packet->payload_size(), expected_payload_capacity);
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
return false;
// No FEC protection for upper temporal layers, if used.
bool protect_packet = temporal_id == 0 || temporal_id == kNoTemporalIdx;
packet->set_allow_retransmission(allow_retransmission);
// Put packetization finish timestamp into extension.
if (packet->HasExtension<VideoTimingExtension>()) {
packet->set_packetization_finish_time_ms(clock_->TimeInMilliseconds());
}
if (protect_packet && fec_generator_) {
if (red_enabled() &&
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 (packet->GetExtension<webrtc::TransportSequenceNumber>(
&transport_senquence_number)) {
if (!packet->RemoveExtension(webrtc::TransportSequenceNumber::kId)) {
RTC_NOTREACHED()
<< "Failed to remove transport sequence number, packet="
<< packet->ToString();
}
}
}
fec_generator_->AddPacketAndGenerateFec(*packet);
}
if (red_enabled()) {
std::unique_ptr<RtpPacketToSend> red_packet(new RtpPacketToSend(*packet));
BuildRedPayload(*packet, red_packet.get());
red_packet->SetPayloadType(*red_payload_type_);
// Send |red_packet| instead of |packet| for allocated sequence number.
red_packet->set_packet_type(RtpPacketMediaType::kVideo);
red_packet->set_allow_retransmission(packet->allow_retransmission());
rtp_packets.emplace_back(std::move(red_packet));
} else {
packet->set_packet_type(RtpPacketMediaType::kVideo);
rtp_packets.emplace_back(std::move(packet));
}
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 (fec_generator_) {
// Fetch any FEC packets generated from the media frame and add them to
// the list of packets to send.
auto fec_packets = fec_generator_->GetFecPackets();
// TODO(bugs.webrtc.org/11340): Move sequence number assignment into
// UlpfecGenerator.
const bool generate_sequence_numbers = !fec_generator_->FecSsrc();
for (auto& fec_packet : fec_packets) {
if (generate_sequence_numbers) {
rtp_sender_->AssignSequenceNumber(fec_packet.get());
}
rtp_packets.emplace_back(std::move(fec_packet));
}
}
LogAndSendToNetwork(std::move(rtp_packets), unpacketized_payload_size);
// Update details about the last sent frame.
last_rotation_ = video_header.rotation;
if (video_header.color_space != last_color_space_) {
last_color_space_ = video_header.color_space;
transmit_color_space_next_frame_ = !IsBaseLayer(video_header);
} else {
transmit_color_space_next_frame_ =
transmit_color_space_next_frame_ ? !IsBaseLayer(video_header) : false;
}
if (video_header.frame_type == VideoFrameType::kVideoFrameKey ||
(IsBaseLayer(video_header) &&
!(video_header.generic.has_value() ? video_header.generic->discardable
: false))) {
// This frame has guaranteed delivery, no need to populate playout
// delay extensions until it changes again.
playout_delay_pending_ = false;
}
TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
rtp_timestamp);
return true;
}
bool RTPSenderVideo::SendEncodedImage(
int payload_type,
absl::optional<VideoCodecType> codec_type,
uint32_t rtp_timestamp,
const EncodedImage& encoded_image,
const RTPFragmentationHeader* fragmentation,
RTPVideoHeader video_header,
absl::optional<int64_t> expected_retransmission_time_ms) {
if (frame_transformer_delegate_) {
// The frame will be sent async once transformed.
return frame_transformer_delegate_->TransformFrame(
payload_type, codec_type, rtp_timestamp, encoded_image, fragmentation,
video_header, expected_retransmission_time_ms, rtp_sender_->SSRC());
}
return SendVideo(payload_type, codec_type, rtp_timestamp,
encoded_image.capture_time_ms_, encoded_image, fragmentation,
video_header, expected_retransmission_time_ms);
}
uint32_t RTPSenderVideo::VideoBitrateSent() const {
rtc::CritScope cs(&stats_crit_);
return video_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
uint32_t RTPSenderVideo::FecOverheadRate() const {
return fec_generator_ ? fec_generator_->CurrentFecRate().bps<uint32_t>() : 0u;
}
uint32_t RTPSenderVideo::PacketizationOverheadBps() const {
rtc::CritScope cs(&stats_crit_);
return packetization_overhead_bitrate_.Rate(clock_->TimeInMilliseconds())
.value_or(0);
}
bool RTPSenderVideo::AllowRetransmission(
uint8_t temporal_id,
int32_t retransmission_settings,
int64_t expected_retransmission_time_ms) {
if (retransmission_settings == kRetransmitOff)
return false;
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 true;
if ((retransmission_settings & kRetransmitBaseLayer) && temporal_id == 0)
return true;
if ((retransmission_settings & kRetransmitHigherLayers) && temporal_id > 0)
return true;
return false;
}
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 RTPVideoHeaderLegacyGeneric&) {
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;
}
void RTPSenderVideo::MaybeUpdateCurrentPlayoutDelay(
const RTPVideoHeader& header) {
if (IsNoopDelay(header.playout_delay)) {
return;
}
PlayoutDelay requested_delay = header.playout_delay;
if (requested_delay.min_ms > PlayoutDelayLimits::kMaxMs ||
requested_delay.max_ms > PlayoutDelayLimits::kMaxMs) {
RTC_DLOG(LS_ERROR)
<< "Requested playout delay values out of range, ignored";
return;
}
if (requested_delay.max_ms != -1 &&
requested_delay.min_ms > requested_delay.max_ms) {
RTC_DLOG(LS_ERROR) << "Requested playout delay values out of order";
return;
}
if (!playout_delay_pending_) {
current_playout_delay_ = requested_delay;
playout_delay_pending_ = true;
return;
}
if ((requested_delay.min_ms == -1 ||
requested_delay.min_ms == current_playout_delay_.min_ms) &&
(requested_delay.max_ms == -1 ||
requested_delay.max_ms == current_playout_delay_.max_ms)) {
// No change, ignore.
return;
}
if (requested_delay.min_ms == -1) {
RTC_DCHECK_GE(requested_delay.max_ms, 0);
requested_delay.min_ms =
std::min(current_playout_delay_.min_ms, requested_delay.max_ms);
}
if (requested_delay.max_ms == -1) {
requested_delay.max_ms =
std::max(current_playout_delay_.max_ms, requested_delay.min_ms);
}
current_playout_delay_ = requested_delay;
playout_delay_pending_ = true;
}
} // namespace webrtc