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2bac7da1349c75e5cf89612ab9619a1920d5d974
platform-external-webrtc/modules/rtp_rtcp/source/rtp_sender.cc
Steve Anton 2bac7da134 Optimize sending the MID and (R)RID header extensions 
These RTP header extensions are used for Unified Plan SDP / BUNDLE and
replace SSRC signaling.

Previously, the RTPSender would attach these header extensions to every
packet when configured. Now, the header extensions will be attached to
every packet until the an RTCP RR is received on that SSRC which
indicates the receiver knows what MID/RID the SSRC is associated with.

This should reduce overhead by 2-4 bytes per packet when the MID header
extension is used and by 4-8 bytes when both header extensions are used.

Bug: webrtc:10078
Change-Id: I5fa3ce28a75224adf11d2792bf4ff8dc76e46d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146480
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28685}
2019-07-25 19:23:14 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include <algorithm>
#include <limits>
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/array_view.h"
#include "api/transport/field_trial_based_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/source/byte_io.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 "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/time_utils.h"
namespace webrtc {
namespace {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
constexpr size_t kMaxPaddingLength = 224;
constexpr size_t kMinAudioPaddingLength = 50;
constexpr int kSendSideDelayWindowMs = 1000;
constexpr size_t kRtpHeaderLength = 12;
constexpr uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1.
constexpr uint32_t kTimestampTicksPerMs = 90;
constexpr int kBitrateStatisticsWindowMs = 1000;
constexpr size_t kMinFlexfecPacketsToStoreForPacing = 50;
// Min size needed to get payload padding from packet history.
constexpr int kMinPayloadPaddingBytes = 50;
template <typename Extension>
constexpr RtpExtensionSize CreateExtensionSize() {
return {Extension::kId, Extension::kValueSizeBytes};
}
template <typename Extension>
constexpr RtpExtensionSize CreateMaxExtensionSize() {
return {Extension::kId, Extension::kMaxValueSizeBytes};
}
// Size info for header extensions that might be used in padding or FEC packets.
constexpr RtpExtensionSize kFecOrPaddingExtensionSizes[] = {
CreateExtensionSize<AbsoluteSendTime>(),
CreateExtensionSize<TransmissionOffset>(),
CreateExtensionSize<TransportSequenceNumber>(),
CreateExtensionSize<PlayoutDelayLimits>(),
CreateMaxExtensionSize<RtpMid>(),
};
// Size info for header extensions that might be used in video packets.
constexpr RtpExtensionSize kVideoExtensionSizes[] = {
CreateExtensionSize<AbsoluteSendTime>(),
CreateExtensionSize<AbsoluteCaptureTimeExtension>(),
CreateExtensionSize<TransmissionOffset>(),
CreateExtensionSize<TransportSequenceNumber>(),
CreateExtensionSize<PlayoutDelayLimits>(),
CreateExtensionSize<VideoOrientation>(),
CreateExtensionSize<VideoContentTypeExtension>(),
CreateExtensionSize<VideoTimingExtension>(),
CreateMaxExtensionSize<RtpStreamId>(),
CreateMaxExtensionSize<RepairedRtpStreamId>(),
CreateMaxExtensionSize<RtpMid>(),
{RtpGenericFrameDescriptorExtension00::kId,
RtpGenericFrameDescriptorExtension00::kMaxSizeBytes},
{RtpGenericFrameDescriptorExtension01::kId,
RtpGenericFrameDescriptorExtension01::kMaxSizeBytes},
};
// TODO(bugs.webrtc.org/10633): Remove when downstream code stops using
// priority. At the time of writing, the priority can be directly mapped to a
// packet type. This is only for a transition period.
RtpPacketToSend::Type PacketPriorityToType(RtpPacketSender::Priority priority) {
switch (priority) {
case RtpPacketSender::Priority::kLowPriority:
return RtpPacketToSend::Type::kVideo;
case RtpPacketSender::Priority::kNormalPriority:
return RtpPacketToSend::Type::kRetransmission;
case RtpPacketSender::Priority::kHighPriority:
return RtpPacketToSend::Type::kAudio;
default:
RTC_NOTREACHED() << "Unexpected priority: " << priority;
return RtpPacketToSend::Type::kVideo;
}
}
// TODO(bugs.webrtc.org/10633): Remove when packets are always owned by pacer.
RtpPacketSender::Priority PacketTypeToPriority(RtpPacketToSend::Type type) {
switch (type) {
case RtpPacketToSend::Type::kAudio:
return RtpPacketSender::Priority::kHighPriority;
case RtpPacketToSend::Type::kVideo:
return RtpPacketSender::Priority::kLowPriority;
case RtpPacketToSend::Type::kRetransmission:
return RtpPacketSender::Priority::kNormalPriority;
case RtpPacketToSend::Type::kForwardErrorCorrection:
return RtpPacketSender::Priority::kLowPriority;
break;
case RtpPacketToSend::Type::kPadding:
RTC_NOTREACHED() << "Unexpected type for legacy path: kPadding";
break;
}
return RtpPacketSender::Priority::kLowPriority;
}
bool IsEnabled(absl::string_view name,
const WebRtcKeyValueConfig* field_trials) {
FieldTrialBasedConfig default_trials;
auto& trials = field_trials ? *field_trials : default_trials;
return trials.Lookup(name).find("Enabled") == 0;
}
bool HasBweExtension(const RtpHeaderExtensionMap& extensions_map) {
return extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber) ||
extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber02) ||
extensions_map.IsRegistered(kRtpExtensionAbsoluteSendTime) ||
extensions_map.IsRegistered(kRtpExtensionTransmissionTimeOffset);
}
} // namespace
RTPSender::RTPSender(const RtpRtcp::Configuration& config)
: clock_(config.clock),
random_(clock_->TimeInMicroseconds()),
audio_configured_(config.audio),
flexfec_ssrc_(config.flexfec_sender
? absl::make_optional(config.flexfec_sender->ssrc())
: absl::nullopt),
paced_sender_(config.paced_sender),
transport_sequence_number_allocator_(
config.transport_sequence_number_allocator),
transport_feedback_observer_(config.transport_feedback_callback),
transport_(config.outgoing_transport),
sending_media_(true), // Default to sending media.
force_part_of_allocation_(false),
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
last_payload_type_(-1),
rtp_header_extension_map_(config.extmap_allow_mixed),
packet_history_(clock_),
flexfec_packet_history_(clock_),
// Statistics
send_delays_(),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
rtp_stats_callback_(nullptr),
total_bitrate_sent_(kBitrateStatisticsWindowMs,
RateStatistics::kBpsScale),
nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
send_side_delay_observer_(config.send_side_delay_observer),
event_log_(config.event_log),
send_packet_observer_(config.send_packet_observer),
bitrate_callback_(config.send_bitrate_observer),
// RTP variables
sequence_number_forced_(false),
ssrc_(config.media_send_ssrc),
ssrc_has_acked_(false),
rtx_ssrc_has_acked_(false),
last_rtp_timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
media_has_been_sent_(false),
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
ssrc_rtx_(config.rtx_send_ssrc),
rtp_overhead_bytes_per_packet_(0),
supports_bwe_extension_(false),
retransmission_rate_limiter_(config.retransmission_rate_limiter),
overhead_observer_(config.overhead_observer),
populate_network2_timestamp_(config.populate_network2_timestamp),
send_side_bwe_with_overhead_(
IsEnabled("WebRTC-SendSideBwe-WithOverhead", config.field_trials)),
pacer_legacy_packet_referencing_(
IsEnabled("WebRTC-Pacer-LegacyPacketReferencing",
config.field_trials)) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
// Store FlexFEC packets in the packet history data structure, so they can
// be found when paced.
if (flexfec_ssrc_) {
flexfec_packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull,
kMinFlexfecPacketsToStoreForPacing);
}
}
RTPSender::RTPSender(
bool audio,
Clock* clock,
Transport* transport,
RtpPacketSender* paced_sender,
absl::optional<uint32_t> flexfec_ssrc,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_observer,
BitrateStatisticsObserver* bitrate_callback,
SendSideDelayObserver* send_side_delay_observer,
RtcEventLog* event_log,
SendPacketObserver* send_packet_observer,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
bool populate_network2_timestamp,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
bool extmap_allow_mixed,
const WebRtcKeyValueConfig& field_trials)
: clock_(clock),
random_(clock_->TimeInMicroseconds()),
audio_configured_(audio),
flexfec_ssrc_(flexfec_ssrc),
paced_sender_(paced_sender),
transport_sequence_number_allocator_(sequence_number_allocator),
transport_feedback_observer_(transport_feedback_observer),
transport_(transport),
sending_media_(true), // Default to sending media.
force_part_of_allocation_(false),
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
last_payload_type_(-1),
rtp_header_extension_map_(extmap_allow_mixed),
packet_history_(clock),
flexfec_packet_history_(clock),
// Statistics
send_delays_(),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
rtp_stats_callback_(nullptr),
total_bitrate_sent_(kBitrateStatisticsWindowMs,
RateStatistics::kBpsScale),
nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
send_side_delay_observer_(send_side_delay_observer),
event_log_(event_log),
send_packet_observer_(send_packet_observer),
bitrate_callback_(bitrate_callback),
// RTP variables
sequence_number_forced_(false),
ssrc_has_acked_(false),
rtx_ssrc_has_acked_(false),
last_rtp_timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
media_has_been_sent_(false),
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
rtp_overhead_bytes_per_packet_(0),
supports_bwe_extension_(false),
retransmission_rate_limiter_(retransmission_rate_limiter),
overhead_observer_(overhead_observer),
populate_network2_timestamp_(populate_network2_timestamp),
send_side_bwe_with_overhead_(
field_trials.Lookup("WebRTC-SendSideBwe-WithOverhead")
.find("Enabled") == 0),
pacer_legacy_packet_referencing_(
field_trials.Lookup("WebRTC-Pacer-LegacyPacketReferencing")
.find("Enabled") == 0) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
// Store FlexFEC packets in the packet history data structure, so they can
// be found when paced.
if (flexfec_ssrc_) {
flexfec_packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStoreAndCull,
kMinFlexfecPacketsToStoreForPacing);
}
}
RTPSender::~RTPSender() {
// TODO(tommi): Use a thread checker to ensure the object is created and
// deleted on the same thread. At the moment this isn't possible due to
// voe::ChannelOwner in voice engine. To reproduce, run:
// voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
// TODO(tommi,holmer): We don't grab locks in the dtor before accessing member
// variables but we grab them in all other methods. (what's the design?)
// Start documenting what thread we're on in what method so that it's easier
// to understand performance attributes and possibly remove locks.
}
rtc::ArrayView<const RtpExtensionSize> RTPSender::FecExtensionSizes() {
return rtc::MakeArrayView(kFecOrPaddingExtensionSizes,
arraysize(kFecOrPaddingExtensionSizes));
}
rtc::ArrayView<const RtpExtensionSize> RTPSender::VideoExtensionSizes() {
return rtc::MakeArrayView(kVideoExtensionSizes,
arraysize(kVideoExtensionSizes));
}
uint16_t RTPSender::ActualSendBitrateKbit() const {
rtc::CritScope cs(&statistics_crit_);
return static_cast<uint16_t>(
total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0) /
1000);
}
uint32_t RTPSender::NackOverheadRate() const {
rtc::CritScope cs(&statistics_crit_);
return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
void RTPSender::SetExtmapAllowMixed(bool extmap_allow_mixed) {
rtc::CritScope lock(&send_critsect_);
rtp_header_extension_map_.SetExtmapAllowMixed(extmap_allow_mixed);
}
int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
rtc::CritScope lock(&send_critsect_);
bool registered = rtp_header_extension_map_.RegisterByType(id, type);
supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
return registered ? 0 : -1;
}
bool RTPSender::RegisterRtpHeaderExtension(const std::string& uri, int id) {
rtc::CritScope lock(&send_critsect_);
bool registered = rtp_header_extension_map_.RegisterByUri(id, uri);
supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
return registered;
}
bool RTPSender::IsRtpHeaderExtensionRegistered(RTPExtensionType type) const {
rtc::CritScope lock(&send_critsect_);
return rtp_header_extension_map_.IsRegistered(type);
}
int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
rtc::CritScope lock(&send_critsect_);
int32_t deregistered = rtp_header_extension_map_.Deregister(type);
supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
return deregistered;
}
void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
RTC_DCHECK_GE(max_packet_size, 100);
RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
rtc::CritScope lock(&send_critsect_);
max_packet_size_ = max_packet_size;
}
size_t RTPSender::MaxRtpPacketSize() const {
return max_packet_size_;
}
void RTPSender::SetRtxStatus(int mode) {
rtc::CritScope lock(&send_critsect_);
rtx_ = mode;
}
int RTPSender::RtxStatus() const {
rtc::CritScope lock(&send_critsect_);
return rtx_;
}
void RTPSender::SetRtxSsrc(uint32_t ssrc) {
rtc::CritScope lock(&send_critsect_);
ssrc_rtx_.emplace(ssrc);
}
uint32_t RTPSender::RtxSsrc() const {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK(ssrc_rtx_);
return *ssrc_rtx_;
}
void RTPSender::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK_LE(payload_type, 127);
RTC_DCHECK_LE(associated_payload_type, 127);
if (payload_type < 0) {
RTC_LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type << ".";
return;
}
rtx_payload_type_map_[associated_payload_type] = payload_type;
}
size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
const PacedPacketInfo& pacing_info) {
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return 0;
if ((rtx_ & kRtxRedundantPayloads) == 0)
return 0;
}
int bytes_left = static_cast<int>(bytes_to_send);
while (bytes_left >= kMinPayloadPaddingBytes) {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPayloadPaddingPacket();
if (!packet)
break;
size_t payload_size = packet->payload_size();
if (!PrepareAndSendPacket(std::move(packet), true, false, pacing_info))
break;
bytes_left -= payload_size;
}
return bytes_to_send - bytes_left;
}
size_t RTPSender::SendPadData(size_t bytes,
const PacedPacketInfo& pacing_info) {
size_t padding_bytes_in_packet;
size_t max_payload_size = max_packet_size_ - RtpHeaderLength();
if (audio_configured_) {
// Allow smaller padding packets for audio.
padding_bytes_in_packet =
rtc::SafeClamp(bytes, kMinAudioPaddingLength,
rtc::SafeMin(max_payload_size, kMaxPaddingLength));
} else {
// Always send full padding packets. This is accounted for by the
// RtpPacketSender, which will make sure we don't send too much padding even
// if a single packet is larger than requested.
// We do this to avoid frequently sending small packets on higher bitrates.
padding_bytes_in_packet = rtc::SafeMin(max_payload_size, kMaxPaddingLength);
}
size_t bytes_sent = 0;
while (bytes_sent < bytes) {
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t ssrc;
uint32_t timestamp;
int64_t capture_time_ms;
uint16_t sequence_number;
int payload_type;
bool over_rtx;
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
break;
timestamp = last_rtp_timestamp_;
capture_time_ms = capture_time_ms_;
if (rtx_ == kRtxOff) {
if (last_payload_type_ == -1)
break;
// Without RTX we can't send padding in the middle of frames.
// For audio marker bits doesn't mark the end of a frame and frames
// are usually a single packet, so for now we don't apply this rule
// for audio.
if (!audio_configured_ && !last_packet_marker_bit_) {
break;
}
if (!ssrc_) {
RTC_LOG(LS_ERROR) << "SSRC unset.";
return 0;
}
RTC_DCHECK(ssrc_);
ssrc = *ssrc_;
sequence_number = sequence_number_;
++sequence_number_;
payload_type = last_payload_type_;
over_rtx = false;
} else {
// Without abs-send-time or transport sequence number a media packet
// must be sent before padding so that the timestamps used for
// estimation are correct.
if (!media_has_been_sent_ &&
!(rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId) ||
(rtp_header_extension_map_.IsRegistered(
TransportSequenceNumber::kId) &&
transport_sequence_number_allocator_))) {
break;
}
// Only change change the timestamp of padding packets sent over RTX.
// Padding only packets over RTP has to be sent as part of a media
// frame (and therefore the same timestamp).
if (last_timestamp_time_ms_ > 0) {
timestamp +=
(now_ms - last_timestamp_time_ms_) * kTimestampTicksPerMs;
capture_time_ms += (now_ms - last_timestamp_time_ms_);
}
if (!ssrc_rtx_) {
RTC_LOG(LS_ERROR) << "RTX SSRC unset.";
return 0;
}
RTC_DCHECK(ssrc_rtx_);
ssrc = *ssrc_rtx_;
sequence_number = sequence_number_rtx_;
++sequence_number_rtx_;
payload_type = rtx_payload_type_map_.begin()->second;
over_rtx = true;
}
}
RtpPacketToSend padding_packet(&rtp_header_extension_map_);
padding_packet.SetPayloadType(payload_type);
padding_packet.SetMarker(false);
padding_packet.SetSequenceNumber(sequence_number);
padding_packet.SetTimestamp(timestamp);
padding_packet.SetSsrc(ssrc);
if (capture_time_ms > 0) {
padding_packet.SetExtension<TransmissionOffset>(
(now_ms - capture_time_ms) * kTimestampTicksPerMs);
}
padding_packet.SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
PacketOptions options;
// Padding packets are never retransmissions.
options.is_retransmit = false;
bool has_transport_seq_num;
{
rtc::CritScope lock(&send_critsect_);
has_transport_seq_num =
UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
options.included_in_allocation =
has_transport_seq_num || force_part_of_allocation_;
options.included_in_feedback = has_transport_seq_num;
}
padding_packet.SetPadding(padding_bytes_in_packet);
if (has_transport_seq_num) {
AddPacketToTransportFeedback(options.packet_id, padding_packet,
pacing_info);
}
if (!SendPacketToNetwork(padding_packet, options, pacing_info))
break;
bytes_sent += padding_bytes_in_packet;
UpdateRtpStats(padding_packet, over_rtx, false);
}
return bytes_sent;
}
void RTPSender::SetStorePacketsStatus(bool enable, uint16_t number_to_store) {
packet_history_.SetStorePacketsStatus(
enable ? RtpPacketHistory::StorageMode::kStoreAndCull
: RtpPacketHistory::StorageMode::kDisabled,
number_to_store);
}
bool RTPSender::StorePackets() const {
return packet_history_.GetStorageMode() !=
RtpPacketHistory::StorageMode::kDisabled;
}
int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
// Try to find packet in RTP packet history. Also verify RTT here, so that we
// don't retransmit too often.
absl::optional<RtpPacketHistory::PacketState> stored_packet =
packet_history_.GetPacketState(packet_id);
if (!stored_packet || stored_packet->pending_transmission) {
// Packet not found or already queued for retransmission, ignore.
return 0;
}
const int32_t packet_size = static_cast<int32_t>(stored_packet->packet_size);
const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
if (paced_sender_) {
if (pacer_legacy_packet_referencing_) {
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure reasons.
if (retransmission_rate_limiter_ &&
!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return -1;
}
// Mark packet as being in pacer queue again, to prevent duplicates.
if (!packet_history_.SetPendingTransmission(packet_id)) {
// Packet has already been removed from history, return early.
return 0;
}
paced_sender_->InsertPacket(
RtpPacketSender::kNormalPriority, stored_packet->ssrc,
stored_packet->rtp_sequence_number, stored_packet->capture_time_ms,
stored_packet->packet_size, true);
} else {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPacketAndMarkAsPending(
packet_id, [&](const RtpPacketToSend& stored_packet) {
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure
// reasons.
std::unique_ptr<RtpPacketToSend> retransmit_packet;
if (retransmission_rate_limiter_ &&
!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return retransmit_packet;
}
if (rtx) {
retransmit_packet = BuildRtxPacket(stored_packet);
} else {
retransmit_packet =
absl::make_unique<RtpPacketToSend>(stored_packet);
}
retransmit_packet->set_retransmitted_sequence_number(
stored_packet.SequenceNumber());
return retransmit_packet;
});
if (!packet) {
return -1;
}
packet->set_packet_type(RtpPacketToSend::Type::kRetransmission);
paced_sender_->EnqueuePacket(std::move(packet));
}
return packet_size;
}
// TODO(sprang): Replace this whole code-path with a pass-through pacer.
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure reasons.
if (retransmission_rate_limiter_ &&
!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return -1;
}
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPacketAndSetSendTime(packet_id);
if (!packet) {
// Packet could theoretically time out between the first check and this one.
return 0;
}
if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
return -1;
return packet_size;
}
void RTPSender::OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number) {
rtc::CritScope lock(&send_critsect_);
ssrc_has_acked_ = true;
}
void RTPSender::OnReceivedAckOnRtxSsrc(
int64_t extended_highest_sequence_number) {
rtc::CritScope lock(&send_critsect_);
rtx_ssrc_has_acked_ = true;
}
bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info) {
int bytes_sent = -1;
if (transport_) {
UpdateRtpOverhead(packet);
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->Log(absl::make_unique<RtcEventRtpPacketOutgoing>(
packet, pacing_info.probe_cluster_id));
}
}
// TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
if (bytes_sent <= 0) {
RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
return false;
}
return true;
}
void RTPSender::OnReceivedNack(
const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt) {
packet_history_.SetRtt(5 + avg_rtt);
for (uint16_t seq_no : nack_sequence_numbers) {
const int32_t bytes_sent = ReSendPacket(seq_no);
if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
RTC_LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
<< ", Discard rest of packets.";
break;
}
}
}
// Called from pacer when we can send the packet.
RtpPacketSendResult RTPSender::TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) {
if (!SendingMedia()) {
return RtpPacketSendResult::kPacketNotFound;
}
std::unique_ptr<RtpPacketToSend> packet;
if (ssrc == SSRC()) {
packet = packet_history_.GetPacketAndSetSendTime(sequence_number);
} else if (ssrc == FlexfecSsrc()) {
packet = flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number);
}
if (!packet) {
// Packet cannot be found or was resent too recently.
return RtpPacketSendResult::kPacketNotFound;
}
return PrepareAndSendPacket(
std::move(packet),
retransmission && (RtxStatus() & kRtxRetransmitted) > 0,
retransmission, pacing_info)
? RtpPacketSendResult::kSuccess
: RtpPacketSendResult::kTransportUnavailable;
}
// Called from pacer when we can send the packet.
bool RTPSender::TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(packet);
const uint32_t packet_ssrc = packet->Ssrc();
const auto packet_type = packet->packet_type();
RTC_DCHECK(packet_type.has_value());
PacketOptions options;
bool is_media = false;
bool is_rtx = false;
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_) {
return false;
}
switch (*packet_type) {
case RtpPacketToSend::Type::kAudio:
case RtpPacketToSend::Type::kVideo:
if (packet_ssrc != ssrc_) {
return false;
}
is_media = true;
break;
case RtpPacketToSend::Type::kRetransmission:
case RtpPacketToSend::Type::kPadding:
// Both padding and retransmission must be on either the media or the
// RTX stream.
if (packet_ssrc == ssrc_rtx_) {
is_rtx = true;
} else if (packet_ssrc != ssrc_) {
return false;
}
break;
case RtpPacketToSend::Type::kForwardErrorCorrection:
// FlexFEC is on separate SSRC, ULPFEC uses media SSRC.
if (packet_ssrc != ssrc_ && packet_ssrc != flexfec_ssrc_) {
return false;
}
break;
}
options.included_in_allocation = force_part_of_allocation_;
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
// the pacer, these modifications of the header below are happening after the
// FEC protection packets are calculated. This will corrupt recovered packets
// at the same place. It's not an issue for extensions, which are present in
// all the packets (their content just may be incorrect on recovered packets).
// In case of VideoTimingExtension, since it's present not in every packet,
// data after rtp header may be corrupted if these packets are protected by
// the FEC.
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t diff_ms = now_ms - packet->capture_time_ms();
if (packet->IsExtensionReserved<TransmissionOffset>()) {
packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff_ms);
}
if (packet->IsExtensionReserved<AbsoluteSendTime>()) {
packet->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
}
if (packet->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet->set_network2_time_ms(now_ms);
} else {
packet->set_pacer_exit_time_ms(now_ms);
}
}
// Downstream code actually uses this flag to distinguish between media and
// everything else.
options.is_retransmit = !is_media;
if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) {
options.packet_id = *packet_id;
options.included_in_feedback = true;
options.included_in_allocation = true;
AddPacketToTransportFeedback(*packet_id, *packet, pacing_info);
}
options.application_data.assign(packet->application_data().begin(),
packet->application_data().end());
if (packet->packet_type() != RtpPacketToSend::Type::kPadding &&
packet->packet_type() != RtpPacketToSend::Type::kRetransmission) {
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet_ssrc);
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet_ssrc);
}
const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
// Put packet in retransmission history or update pending status even if
// actual sending fails.
if (is_media && packet->allow_retransmission()) {
packet_history_.PutRtpPacket(absl::make_unique<RtpPacketToSend>(*packet),
StorageType::kAllowRetransmission, now_ms);
} else if (packet->retransmitted_sequence_number()) {
packet_history_.MarkPacketAsSent(*packet->retransmitted_sequence_number());
}
if (send_success) {
UpdateRtpStats(*packet, is_rtx,
packet_type == RtpPacketToSend::Type::kRetransmission);
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
// Return true even if transport failed (will be handled by retransmissions
// instead in that case), so that PacketRouter does not have to iterate over
// all other RTP modules and fail to send there too.
return true;
}
bool RTPSender::SupportsPadding() const {
rtc::CritScope lock(&send_critsect_);
return sending_media_ && supports_bwe_extension_;
}
bool RTPSender::SupportsRtxPayloadPadding() const {
rtc::CritScope lock(&send_critsect_);
return sending_media_ && supports_bwe_extension_ &&
(rtx_ & kRtxRedundantPayloads);
}
bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
bool send_over_rtx,
bool is_retransmit,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(packet);
int64_t capture_time_ms = packet->capture_time_ms();
RtpPacketToSend* packet_to_send = packet.get();
std::unique_ptr<RtpPacketToSend> packet_rtx;
if (send_over_rtx) {
packet_rtx = BuildRtxPacket(*packet);
if (!packet_rtx)
return false;
packet_to_send = packet_rtx.get();
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
// the pacer, these modifications of the header below are happening after the
// FEC protection packets are calculated. This will corrupt recovered packets
// at the same place. It's not an issue for extensions, which are present in
// all the packets (their content just may be incorrect on recovered packets).
// In case of VideoTimingExtension, since it's present not in every packet,
// data after rtp header may be corrupted if these packets are protected by
// the FEC.
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t diff_ms = now_ms - capture_time_ms;
packet_to_send->SetExtension<TransmissionOffset>(kTimestampTicksPerMs *
diff_ms);
packet_to_send->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
if (packet_to_send->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet_to_send->set_network2_time_ms(now_ms);
} else {
packet_to_send->set_pacer_exit_time_ms(now_ms);
}
}
PacketOptions options;
// If we are sending over RTX, it also means this is a retransmission.
// E.g. RTPSender::TrySendRedundantPayloads calls PrepareAndSendPacket with
// send_over_rtx = true but is_retransmit = false.
options.is_retransmit = is_retransmit || send_over_rtx;
bool has_transport_seq_num;
{
rtc::CritScope lock(&send_critsect_);
has_transport_seq_num =
UpdateTransportSequenceNumber(packet_to_send, &options.packet_id);
options.included_in_allocation =
has_transport_seq_num || force_part_of_allocation_;
options.included_in_feedback = has_transport_seq_num;
}
if (has_transport_seq_num) {
AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
pacing_info);
}
options.application_data.assign(packet_to_send->application_data().begin(),
packet_to_send->application_data().end());
if (!is_retransmit && !send_over_rtx) {
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet->Ssrc());
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet->Ssrc());
}
if (!SendPacketToNetwork(*packet_to_send, options, pacing_info))
return false;
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(*packet_to_send, send_over_rtx, is_retransmit);
return true;
}
void RTPSender::UpdateRtpStats(const RtpPacketToSend& packet,
bool is_rtx,
bool is_retransmit) {
int64_t now_ms = clock_->TimeInMilliseconds();
rtc::CritScope lock(&statistics_crit_);
StreamDataCounters* counters = is_rtx ? &rtx_rtp_stats_ : &rtp_stats_;
total_bitrate_sent_.Update(packet.size(), now_ms);
if (counters->first_packet_time_ms == -1)
counters->first_packet_time_ms = now_ms;
if (packet.packet_type() == RtpPacketToSend::Type::kForwardErrorCorrection) {
counters->fec.AddPacket(packet);
}
if (is_retransmit) {
counters->retransmitted.AddPacket(packet);
nack_bitrate_sent_.Update(packet.size(), now_ms);
}
counters->transmitted.AddPacket(packet);
if (rtp_stats_callback_)
rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
}
size_t RTPSender::TimeToSendPadding(size_t bytes,
const PacedPacketInfo& pacing_info) {
if (bytes == 0)
return 0;
size_t bytes_sent = TrySendRedundantPayloads(bytes, pacing_info);
if (bytes_sent < bytes)
bytes_sent += SendPadData(bytes - bytes_sent, pacing_info);
return bytes_sent;
}
std::vector<std::unique_ptr<RtpPacketToSend>> RTPSender::GeneratePadding(
size_t target_size_bytes) {
// This method does not actually send packets, it just generates
// them and puts them in the pacer queue. Since this should incur
// low overhead, keep the lock for the scope of the method in order
// to make the code more readable.
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
size_t bytes_left = target_size_bytes;
if (SupportsRtxPayloadPadding()) {
while (bytes_left >= kMinPayloadPaddingBytes) {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPayloadPaddingPacket(
[&](const RtpPacketToSend& packet)
-> std::unique_ptr<RtpPacketToSend> {
return BuildRtxPacket(packet);
});
if (!packet) {
break;
}
bytes_left -= std::min(bytes_left, packet->payload_size());
packet->set_packet_type(RtpPacketToSend::Type::kPadding);
padding_packets.push_back(std::move(packet));
}
}
rtc::CritScope lock(&send_critsect_);
if (!sending_media_) {
return {};
}
size_t padding_bytes_in_packet;
const size_t max_payload_size = max_packet_size_ - RtpHeaderLength();
if (audio_configured_) {
// Allow smaller padding packets for audio.
padding_bytes_in_packet = rtc::SafeClamp<size_t>(
bytes_left, kMinAudioPaddingLength,
rtc::SafeMin(max_payload_size, kMaxPaddingLength));
} else {
// Always send full padding packets. This is accounted for by the
// RtpPacketSender, which will make sure we don't send too much padding even
// if a single packet is larger than requested.
// We do this to avoid frequently sending small packets on higher bitrates.
padding_bytes_in_packet = rtc::SafeMin(max_payload_size, kMaxPaddingLength);
}
while (bytes_left > 0) {
auto padding_packet =
absl::make_unique<RtpPacketToSend>(&rtp_header_extension_map_);
padding_packet->set_packet_type(RtpPacketToSend::Type::kPadding);
padding_packet->SetMarker(false);
padding_packet->SetTimestamp(last_rtp_timestamp_);
padding_packet->set_capture_time_ms(capture_time_ms_);
if (rtx_ == kRtxOff) {
if (last_payload_type_ == -1) {
break;
}
// Without RTX we can't send padding in the middle of frames.
// For audio marker bits doesn't mark the end of a frame and frames
// are usually a single packet, so for now we don't apply this rule
// for audio.
if (!audio_configured_ && !last_packet_marker_bit_) {
break;
}
RTC_DCHECK(ssrc_);
padding_packet->SetSsrc(*ssrc_);
padding_packet->SetPayloadType(last_payload_type_);
padding_packet->SetSequenceNumber(sequence_number_++);
} else {
// Without abs-send-time or transport sequence number a media packet
// must be sent before padding so that the timestamps used for
// estimation are correct.
if (!media_has_been_sent_ &&
!(rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId) ||
rtp_header_extension_map_.IsRegistered(
TransportSequenceNumber::kId))) {
break;
}
// Only change the timestamp of padding packets sent over RTX.
// Padding only packets over RTP has to be sent as part of a media
// frame (and therefore the same timestamp).
int64_t now_ms = clock_->TimeInMilliseconds();
if (last_timestamp_time_ms_ > 0) {
padding_packet->SetTimestamp(padding_packet->Timestamp() +
(now_ms - last_timestamp_time_ms_) *
kTimestampTicksPerMs);
padding_packet->set_capture_time_ms(padding_packet->capture_time_ms() +
(now_ms - last_timestamp_time_ms_));
}
RTC_DCHECK(ssrc_rtx_);
padding_packet->SetSsrc(*ssrc_rtx_);
padding_packet->SetSequenceNumber(sequence_number_rtx_++);
padding_packet->SetPayloadType(rtx_payload_type_map_.begin()->second);
}
if (rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId)) {
padding_packet->ReserveExtension<TransportSequenceNumber>();
}
if (rtp_header_extension_map_.IsRegistered(TransmissionOffset::kId)) {
padding_packet->ReserveExtension<TransmissionOffset>();
}
if (rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId)) {
padding_packet->ReserveExtension<AbsoluteSendTime>();
}
padding_packet->SetPadding(padding_bytes_in_packet);
bytes_left -= std::min(bytes_left, padding_bytes_in_packet);
padding_packets.push_back(std::move(padding_packet));
}
return padding_packets;
}
bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage) {
RTC_DCHECK(packet);
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t ssrc = packet->Ssrc();
if (paced_sender_) {
uint16_t seq_no = packet->SequenceNumber();
int64_t capture_time_ms = packet->capture_time_ms();
size_t packet_size =
send_side_bwe_with_overhead_ ? packet->size() : packet->payload_size();
auto packet_type = packet->packet_type();
RTC_CHECK(packet_type) << "Packet type must be set before sending.";
if (packet->capture_time_ms() <= 0) {
packet->set_capture_time_ms(now_ms);
}
if (pacer_legacy_packet_referencing_) {
// If |pacer_reference_packets_| then pacer needs to find the packet in
// the history when it is time to send, so move packet there.
if (ssrc == FlexfecSsrc()) {
// Store FlexFEC packets in a separate history since they are on a
// separate SSRC.
flexfec_packet_history_.PutRtpPacket(std::move(packet), storage,
absl::nullopt);
} else {
packet_history_.PutRtpPacket(std::move(packet), storage, absl::nullopt);
}
paced_sender_->InsertPacket(PacketTypeToPriority(*packet_type), ssrc,
seq_no, capture_time_ms, packet_size, false);
} else {
packet->set_allow_retransmission(storage ==
StorageType::kAllowRetransmission);
paced_sender_->EnqueuePacket(std::move(packet));
}
return true;
}
PacketOptions options;
options.is_retransmit = false;
// |capture_time_ms| <= 0 is considered invalid.
// TODO(holmer): This should be changed all over Video Engine so that negative
// time is consider invalid, while 0 is considered a valid time.
if (packet->capture_time_ms() > 0) {
packet->SetExtension<TransmissionOffset>(
kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
if (populate_network2_timestamp_ &&
packet->HasExtension<VideoTimingExtension>()) {
packet->set_network2_time_ms(now_ms);
}
}
packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
bool has_transport_seq_num;
{
rtc::CritScope lock(&send_critsect_);
has_transport_seq_num =
UpdateTransportSequenceNumber(packet.get(), &options.packet_id);
options.included_in_allocation =
has_transport_seq_num || force_part_of_allocation_;
options.included_in_feedback = has_transport_seq_num;
}
if (has_transport_seq_num) {
AddPacketToTransportFeedback(options.packet_id, *packet.get(),
PacedPacketInfo());
}
options.application_data.assign(packet->application_data().begin(),
packet->application_data().end());
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet->Ssrc());
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet->Ssrc());
bool sent = SendPacketToNetwork(*packet, options, PacedPacketInfo());
if (sent) {
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(*packet, false, false);
}
// To support retransmissions, we store the media packet as sent in the
// packet history (even if send failed).
if (storage == kAllowRetransmission) {
RTC_DCHECK_EQ(ssrc, SSRC());
packet_history_.PutRtpPacket(std::move(packet), storage, now_ms);
}
return sent;
}
bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage,
RtpPacketSender::Priority priority) {
packet->set_packet_type(PacketPriorityToType(priority));
return SendToNetwork(std::move(packet), storage);
}
void RTPSender::RecomputeMaxSendDelay() {
max_delay_it_ = send_delays_.begin();
for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
if (it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
}
}
void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms,
int64_t now_ms,
uint32_t ssrc) {
if (!send_side_delay_observer_ || capture_time_ms <= 0)
return;
int avg_delay_ms = 0;
int max_delay_ms = 0;
uint64_t total_packet_send_delay_ms = 0;
{
rtc::CritScope cs(&statistics_crit_);
// Compute the max and average of the recent capture-to-send delays.
// The time complexity of the current approach depends on the distribution
// of the delay values. This could be done more efficiently.
// Remove elements older than kSendSideDelayWindowMs.
auto lower_bound =
send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs);
for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
if (max_delay_it_ == it) {
max_delay_it_ = send_delays_.end();
}
sum_delays_ms_ -= it->second;
}
send_delays_.erase(send_delays_.begin(), lower_bound);
if (max_delay_it_ == send_delays_.end()) {
// Removed the previous max. Need to recompute.
RecomputeMaxSendDelay();
}
// Add the new element.
RTC_DCHECK_GE(now_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(now_ms, std::numeric_limits<int64_t>::max() / 2);
RTC_DCHECK_GE(capture_time_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(capture_time_ms, std::numeric_limits<int64_t>::max() / 2);
int64_t diff_ms = now_ms - capture_time_ms;
RTC_DCHECK_GE(diff_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(diff_ms,
static_cast<int64_t>(std::numeric_limits<int>::max()));
int new_send_delay = rtc::dchecked_cast<int>(now_ms - capture_time_ms);
SendDelayMap::iterator it;
bool inserted;
std::tie(it, inserted) =
send_delays_.insert(std::make_pair(now_ms, new_send_delay));
if (!inserted) {
// TODO(terelius): If we have multiple delay measurements during the same
// millisecond then we keep the most recent one. It is not clear that this
// is the right decision, but it preserves an earlier behavior.
int previous_send_delay = it->second;
sum_delays_ms_ -= previous_send_delay;
it->second = new_send_delay;
if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
RecomputeMaxSendDelay();
}
}
if (max_delay_it_ == send_delays_.end() ||
it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
sum_delays_ms_ += new_send_delay;
total_packet_send_delay_ms_ += new_send_delay;
total_packet_send_delay_ms = total_packet_send_delay_ms_;
size_t num_delays = send_delays_.size();
RTC_DCHECK(max_delay_it_ != send_delays_.end());
max_delay_ms = rtc::dchecked_cast<int>(max_delay_it_->second);
int64_t avg_ms = (sum_delays_ms_ + num_delays / 2) / num_delays;
RTC_DCHECK_GE(avg_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(avg_ms,
static_cast<int64_t>(std::numeric_limits<int>::max()));
avg_delay_ms =
rtc::dchecked_cast<int>((sum_delays_ms_ + num_delays / 2) / num_delays);
}
send_side_delay_observer_->SendSideDelayUpdated(
avg_delay_ms, max_delay_ms, total_packet_send_delay_ms, ssrc);
}
void RTPSender::UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc) {
if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1)
return;
send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
}
void RTPSender::ProcessBitrate() {
if (!bitrate_callback_)
return;
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t ssrc;
{
rtc::CritScope lock(&send_critsect_);
if (!ssrc_)
return;
ssrc = *ssrc_;
}
rtc::CritScope lock(&statistics_crit_);
bitrate_callback_->Notify(total_bitrate_sent_.Rate(now_ms).value_or(0),
nack_bitrate_sent_.Rate(now_ms).value_or(0), ssrc);
}
size_t RTPSender::RtpHeaderLength() const {
rtc::CritScope lock(&send_critsect_);
size_t rtp_header_length = kRtpHeaderLength;
rtp_header_length += sizeof(uint32_t) * csrcs_.size();
rtp_header_length += RtpHeaderExtensionSize(kFecOrPaddingExtensionSizes,
rtp_header_extension_map_);
return rtp_header_length;
}
uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
rtc::CritScope lock(&send_critsect_);
uint16_t first_allocated_sequence_number = sequence_number_;
sequence_number_ += packets_to_send;
return first_allocated_sequence_number;
}
void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
rtc::CritScope lock(&statistics_crit_);
*rtp_stats = rtp_stats_;
*rtx_stats = rtx_rtp_stats_;
}
std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
rtc::CritScope lock(&send_critsect_);
// TODO(danilchap): Find better motivator and value for extra capacity.
// RtpPacketizer might slightly miscalulate needed size,
// SRTP may benefit from extra space in the buffer and do encryption in place
// saving reallocation.
// While sending slightly oversized packet increase chance of dropped packet,
// it is better than crash on drop packet without trying to send it.
static constexpr int kExtraCapacity = 16;
auto packet = absl::make_unique<RtpPacketToSend>(
&rtp_header_extension_map_, max_packet_size_ + kExtraCapacity);
RTC_DCHECK(ssrc_);
packet->SetSsrc(*ssrc_);
packet->SetCsrcs(csrcs_);
// Reserve extensions, if registered, RtpSender set in SendToNetwork.
packet->ReserveExtension<AbsoluteSendTime>();
packet->ReserveExtension<TransmissionOffset>();
packet->ReserveExtension<TransportSequenceNumber>();
// BUNDLE requires that the receiver "bind" the received SSRC to the values
// in the MID and/or (R)RID header extensions if present. Therefore, the
// sender can reduce overhead by omitting these header extensions once it
// knows that the receiver has "bound" the SSRC.
//
// The algorithm here is fairly simple: Always attach a MID and/or RID (if
// configured) to the outgoing packets until an RTCP receiver report comes
// back for this SSRC. That feedback indicates the receiver must have
// received a packet with the SSRC and header extension(s), so the sender
// then stops attaching the MID and RID.
if (!ssrc_has_acked_) {
// These are no-ops if the corresponding header extension is not registered.
if (!mid_.empty()) {
packet->SetExtension<RtpMid>(mid_);
}
if (!rid_.empty()) {
packet->SetExtension<RtpStreamId>(rid_);
}
}
return packet;
}
bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return false;
RTC_DCHECK(packet->Ssrc() == ssrc_);
packet->SetSequenceNumber(sequence_number_++);
// Remember marker bit to determine if padding can be inserted with
// sequence number following |packet|.
last_packet_marker_bit_ = packet->Marker();
// Remember payload type to use in the padding packet if rtx is disabled.
last_payload_type_ = packet->PayloadType();
// Save timestamps to generate timestamp field and extensions for the padding.
last_rtp_timestamp_ = packet->Timestamp();
last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
capture_time_ms_ = packet->capture_time_ms();
return true;
}
bool RTPSender::UpdateTransportSequenceNumber(RtpPacketToSend* packet,
int* packet_id) {
RTC_DCHECK(packet);
RTC_DCHECK(packet_id);
if (!rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId))
return false;
if (!transport_sequence_number_allocator_)
return false;
*packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
if (!packet->SetExtension<TransportSequenceNumber>(*packet_id))
return false;
return true;
}
void RTPSender::SetSendingMediaStatus(bool enabled) {
rtc::CritScope lock(&send_critsect_);
sending_media_ = enabled;
}
bool RTPSender::SendingMedia() const {
rtc::CritScope lock(&send_critsect_);
return sending_media_;
}
void RTPSender::SetAsPartOfAllocation(bool part_of_allocation) {
rtc::CritScope lock(&send_critsect_);
force_part_of_allocation_ = part_of_allocation;
}
void RTPSender::SetTimestampOffset(uint32_t timestamp) {
rtc::CritScope lock(&send_critsect_);
timestamp_offset_ = timestamp;
}
uint32_t RTPSender::TimestampOffset() const {
rtc::CritScope lock(&send_critsect_);
return timestamp_offset_;
}
void RTPSender::SetSSRC(uint32_t ssrc) {
{
rtc::CritScope lock(&send_critsect_);
if (ssrc_ == ssrc) {
return; // Since it's the same SSRC, don't reset anything.
}
ssrc_.emplace(ssrc);
if (!sequence_number_forced_) {
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
// Clear RTP packet history, since any packets there belong to the old SSRC
// and they may conflict with packets from the new one.
packet_history_.Clear();
}
uint32_t RTPSender::SSRC() const {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK(ssrc_);
return *ssrc_;
}
void RTPSender::SetRid(const std::string& rid) {
// RID is used in simulcast scenario when multiple layers share the same mid.
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK_LE(rid.length(), RtpStreamId::kMaxValueSizeBytes);
rid_ = rid;
}
void RTPSender::SetMid(const std::string& mid) {
// This is configured via the API.
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK_LE(mid.length(), RtpMid::kMaxValueSizeBytes);
mid_ = mid;
}
absl::optional<uint32_t> RTPSender::FlexfecSsrc() const {
return flexfec_ssrc_;
}
void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
rtc::CritScope lock(&send_critsect_);
csrcs_ = csrcs;
}
void RTPSender::SetSequenceNumber(uint16_t seq) {
bool updated_sequence_number = false;
{
rtc::CritScope lock(&send_critsect_);
sequence_number_forced_ = true;
if (sequence_number_ != seq) {
updated_sequence_number = true;
}
sequence_number_ = seq;
}
if (updated_sequence_number) {
// Sequence number series has been reset to a new value, clear RTP packet
// history, since any packets there may conflict with new ones.
packet_history_.Clear();
}
}
uint16_t RTPSender::SequenceNumber() const {
rtc::CritScope lock(&send_critsect_);
return sequence_number_;
}
static void CopyHeaderAndExtensionsToRtxPacket(const RtpPacketToSend& packet,
RtpPacketToSend* rtx_packet) {
// Set the relevant fixed packet headers. The following are not set:
// * Payload type - it is replaced in rtx packets.
// * Sequence number - RTX has a separate sequence numbering.
// * SSRC - RTX stream has its own SSRC.
rtx_packet->SetMarker(packet.Marker());
rtx_packet->SetTimestamp(packet.Timestamp());
// Set the variable fields in the packet header:
// * CSRCs - must be set before header extensions.
// * Header extensions - replace Rid header with RepairedRid header.
const std::vector<uint32_t> csrcs = packet.Csrcs();
rtx_packet->SetCsrcs(csrcs);
for (int extension_num = kRtpExtensionNone + 1;
extension_num < kRtpExtensionNumberOfExtensions; ++extension_num) {
auto extension = static_cast<RTPExtensionType>(extension_num);
// Stream ID header extensions (MID, RSID) are sent per-SSRC. Since RTX
// operates on a different SSRC, the presence and values of these header
// extensions should be determined separately and not blindly copied.
if (extension == kRtpExtensionMid ||
extension == kRtpExtensionRtpStreamId) {
continue;
}
// Empty extensions should be supported, so not checking |source.empty()|.
if (!packet.HasExtension(extension)) {
continue;
}
rtc::ArrayView<const uint8_t> source = packet.FindExtension(extension);
rtc::ArrayView<uint8_t> destination =
rtx_packet->AllocateExtension(extension, source.size());
// Could happen if any:
// 1. Extension has 0 length.
// 2. Extension is not registered in destination.
// 3. Allocating extension in destination failed.
if (destination.empty() || source.size() != destination.size()) {
continue;
}
std::memcpy(destination.begin(), source.begin(), destination.size());
}
}
std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
const RtpPacketToSend& packet) {
std::unique_ptr<RtpPacketToSend> rtx_packet;
// Add original RTP header.
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return nullptr;
RTC_DCHECK(ssrc_rtx_);
// Replace payload type.
auto kv = rtx_payload_type_map_.find(packet.PayloadType());
if (kv == rtx_payload_type_map_.end())
return nullptr;
rtx_packet = absl::make_unique<RtpPacketToSend>(&rtp_header_extension_map_,
max_packet_size_);
rtx_packet->SetPayloadType(kv->second);
// Replace sequence number.
rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
// Replace SSRC.
rtx_packet->SetSsrc(*ssrc_rtx_);
CopyHeaderAndExtensionsToRtxPacket(packet, rtx_packet.get());
// RTX packets are sent on an SSRC different from the main media, so the
// decision to attach MID and/or RRID header extensions is completely
// separate from that of the main media SSRC.
//
// Note that RTX packets must used the RepairedRtpStreamId (RRID) header
// extension instead of the RtpStreamId (RID) header extension even though
// the payload is identical.
if (!rtx_ssrc_has_acked_) {
// These are no-ops if the corresponding header extension is not
// registered.
if (!mid_.empty()) {
rtx_packet->SetExtension<RtpMid>(mid_);
}
if (!rid_.empty()) {
rtx_packet->SetExtension<RepairedRtpStreamId>(rid_);
}
}
}
RTC_DCHECK(rtx_packet);
uint8_t* rtx_payload =
rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
if (rtx_payload == nullptr)
return nullptr;
// Add OSN (original sequence number).
ByteWriter<uint16_t>::WriteBigEndian(rtx_payload, packet.SequenceNumber());
// Add original payload data.
auto payload = packet.payload();
memcpy(rtx_payload + kRtxHeaderSize, payload.data(), payload.size());
// Add original application data.
rtx_packet->set_application_data(packet.application_data());
// Copy capture time so e.g. TransmissionOffset is correctly set.
rtx_packet->set_capture_time_ms(packet.capture_time_ms());
return rtx_packet;
}
void RTPSender::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
rtc::CritScope cs(&statistics_crit_);
rtp_stats_callback_ = callback;
}
StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
rtc::CritScope cs(&statistics_crit_);
return rtp_stats_callback_;
}
uint32_t RTPSender::BitrateSent() const {
rtc::CritScope cs(&statistics_crit_);
return total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
void RTPSender::SetRtpState(const RtpState& rtp_state) {
rtc::CritScope lock(&send_critsect_);
sequence_number_ = rtp_state.sequence_number;
sequence_number_forced_ = true;
timestamp_offset_ = rtp_state.start_timestamp;
last_rtp_timestamp_ = rtp_state.timestamp;
capture_time_ms_ = rtp_state.capture_time_ms;
last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms;
media_has_been_sent_ = rtp_state.media_has_been_sent;
ssrc_has_acked_ = rtp_state.ssrc_has_acked;
}
RtpState RTPSender::GetRtpState() const {
rtc::CritScope lock(&send_critsect_);
RtpState state;
state.sequence_number = sequence_number_;
state.start_timestamp = timestamp_offset_;
state.timestamp = last_rtp_timestamp_;
state.capture_time_ms = capture_time_ms_;
state.last_timestamp_time_ms = last_timestamp_time_ms_;
state.media_has_been_sent = media_has_been_sent_;
state.ssrc_has_acked = ssrc_has_acked_;
return state;
}
void RTPSender::SetRtxRtpState(const RtpState& rtp_state) {
rtc::CritScope lock(&send_critsect_);
sequence_number_rtx_ = rtp_state.sequence_number;
rtx_ssrc_has_acked_ = rtp_state.ssrc_has_acked;
}
RtpState RTPSender::GetRtxRtpState() const {
rtc::CritScope lock(&send_critsect_);
RtpState state;
state.sequence_number = sequence_number_rtx_;
state.start_timestamp = timestamp_offset_;
state.ssrc_has_acked = rtx_ssrc_has_acked_;
return state;
}
void RTPSender::AddPacketToTransportFeedback(
uint16_t packet_id,
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info) {
if (transport_feedback_observer_) {
size_t packet_size = packet.payload_size() + packet.padding_size();
if (send_side_bwe_with_overhead_) {
packet_size = packet.size();
}
RtpPacketSendInfo packet_info;
packet_info.ssrc = SSRC();
packet_info.transport_sequence_number = packet_id;
packet_info.has_rtp_sequence_number = true;
packet_info.rtp_sequence_number = packet.SequenceNumber();
packet_info.length = packet_size;
packet_info.pacing_info = pacing_info;
transport_feedback_observer_->OnAddPacket(packet_info);
}
}
void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
if (!overhead_observer_)
return;
size_t overhead_bytes_per_packet;
{
rtc::CritScope lock(&send_critsect_);
if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
return;
}
rtp_overhead_bytes_per_packet_ = packet.headers_size();
overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
}
overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
}
int64_t RTPSender::LastTimestampTimeMs() const {
rtc::CritScope lock(&send_critsect_);
return last_timestamp_time_ms_;
}
void RTPSender::SetRtt(int64_t rtt_ms) {
packet_history_.SetRtt(rtt_ms);
flexfec_packet_history_.SetRtt(rtt_ms);
}
void RTPSender::OnPacketsAcknowledged(
rtc::ArrayView<const uint16_t> sequence_numbers) {
packet_history_.CullAcknowledgedPackets(sequence_numbers);
}
} // namespace webrtc
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