Moves RtpSequenceNumberMap from RtpSenderVideo to RtpSenderEgress.
Bug: webrtc:11340
Change-Id: Icd9032e3589324cb9ee7b699b38a35e733081e55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168192
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30481}
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 3ae0794..956ac0c 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -177,6 +177,8 @@
configuration.rtx_send_ssrc = rtp_config.rtx.ssrcs[i];
}
+ configuration.need_rtp_packet_infos = rtp_config.lntf.enabled;
+
auto rtp_rtcp = RtpRtcp::Create(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
@@ -192,7 +194,6 @@
video_config.frame_encryptor = frame_encryptor;
video_config.require_frame_encryption =
crypto_options.sframe.require_frame_encryption;
- video_config.need_rtp_packet_infos = rtp_config.lntf.enabled;
video_config.enable_retransmit_all_layers = false;
video_config.field_trials = &field_trial_config;
const bool should_disable_red_and_ulpfec =
@@ -770,7 +771,7 @@
rtc::ArrayView<const uint16_t> sequence_numbers) const {
for (const auto& rtp_stream : rtp_streams_) {
if (ssrc == rtp_stream.rtp_rtcp->SSRC()) {
- return rtp_stream.sender_video->GetSentRtpPacketInfos(sequence_numbers);
+ return rtp_stream.rtp_rtcp->GetSentRtpPacketInfos(sequence_numbers);
}
}
return std::vector<RtpSequenceNumberMap::Info>();
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index fbb3bb3..b256f38 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -28,6 +28,7 @@
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/deprecation.h"
@@ -125,6 +126,8 @@
uint32_t local_media_ssrc = 0;
absl::optional<uint32_t> rtx_send_ssrc;
+ bool need_rtp_packet_infos = false;
+
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
};
@@ -284,6 +287,9 @@
virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes) = 0;
+ virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+ rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
+
// **************************************************************************
// RTCP
// **************************************************************************
diff --git a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 83bc7cc..8864df0 100644
--- a/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -94,6 +94,9 @@
MOCK_METHOD1(
GeneratePadding,
std::vector<std::unique_ptr<RtpPacketToSend>>(size_t target_size_bytes));
+ MOCK_CONST_METHOD1(GetSentRtpPacketInfos,
+ std::vector<RtpSequenceNumberMap::Info>(
+ rtc::ArrayView<const uint16_t> sequence_numbers));
MOCK_METHOD2(RegisterRtcpObservers,
void(RtcpIntraFrameObserver* intra_frame_callback,
RtcpBandwidthObserver* bandwidth_callback));
diff --git a/modules/rtp_rtcp/source/rtp_packet_to_send.h b/modules/rtp_rtcp/source/rtp_packet_to_send.h
index d30bbbf..57493e3 100644
--- a/modules/rtp_rtcp/source/rtp_packet_to_send.h
+++ b/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -98,12 +98,19 @@
VideoTimingExtension::kNetwork2TimestampDeltaOffset);
}
+ void set_first_packet_of_frame(bool is_first_packet) {
+ is_first_packet_of_frame_ = is_first_packet;
+ }
+
+ bool is_first_packet_of_frame() const { return is_first_packet_of_frame_; }
+
private:
int64_t capture_time_ms_ = 0;
absl::optional<RtpPacketMediaType> packet_type_;
bool allow_retransmission_ = false;
absl::optional<uint16_t> retransmitted_sequence_number_;
std::vector<uint8_t> application_data_;
+ bool is_first_packet_of_frame_ = false;
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index dfbac29..ff30143 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -224,6 +224,7 @@
void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) {
rtcp_sender_.SetTimestampOffset(timestamp);
rtp_sender_->packet_generator.SetTimestampOffset(timestamp);
+ rtp_sender_->packet_sender.SetTimestampOffset(timestamp);
}
uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
@@ -393,6 +394,13 @@
target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent());
}
+std::vector<RtpSequenceNumberMap::Info>
+ModuleRtpRtcpImpl::GetSentRtpPacketInfos(
+ rtc::ArrayView<const uint16_t> sequence_numbers) const {
+ RTC_DCHECK(rtp_sender_);
+ return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers);
+}
+
size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.MaxRtpPacketSize();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index c03683f..80488a8 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -143,6 +143,9 @@
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes) override;
+ std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+ rtc::ArrayView<const uint16_t> sequence_numbers) const override;
+
// RTCP part.
// Get RTCP status.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 5e4cce9..e259566 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -152,6 +152,7 @@
config.rtt_stats = &rtt_stats_;
config.rtcp_report_interval_ms = rtcp_report_interval_ms_;
config.local_media_ssrc = is_sender_ ? kSenderSsrc : kReceiverSsrc;
+ config.need_rtp_packet_infos = true;
impl_.reset(new ModuleRtpRtcpImpl(config));
impl_->SetRemoteSSRC(is_sender_ ? kReceiverSsrc : kSenderSsrc);
@@ -569,4 +570,61 @@
EXPECT_EQ(sender_.transport_.NumRtcpSent(), 2u);
}
+TEST_F(RtpRtcpImplTest, StoresPacketInfoForSentPackets) {
+ const uint32_t kStartTimestamp = 1u;
+ SetUp();
+ sender_.impl_->SetStartTimestamp(kStartTimestamp);
+
+ PacedPacketInfo pacing_info;
+ RtpPacketToSend packet(nullptr);
+ packet.set_packet_type(RtpPacketToSend::Type::kVideo);
+ packet.SetSsrc(kSenderSsrc);
+
+ // Single-packet frame.
+ packet.SetTimestamp(1);
+ packet.SetSequenceNumber(1);
+ packet.set_first_packet_of_frame(true);
+ packet.SetMarker(true);
+ sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+ std::vector<RtpSequenceNumberMap::Info> seqno_info =
+ sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{1});
+
+ EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info(
+ /*timestamp=*/1 - kStartTimestamp,
+ /*is_first=*/1,
+ /*is_last=*/1)));
+
+ // Three-packet frame.
+ packet.SetTimestamp(2);
+ packet.SetSequenceNumber(2);
+ packet.set_first_packet_of_frame(true);
+ packet.SetMarker(false);
+ sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+ packet.SetSequenceNumber(3);
+ packet.set_first_packet_of_frame(false);
+ sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+ packet.SetSequenceNumber(4);
+ packet.SetMarker(true);
+ sender_.impl_->TrySendPacket(&packet, pacing_info);
+
+ seqno_info =
+ sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{2, 3, 4});
+
+ EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info(
+ /*timestamp=*/2 - kStartTimestamp,
+ /*is_first=*/1,
+ /*is_last=*/0),
+ RtpSequenceNumberMap::Info(
+ /*timestamp=*/2 - kStartTimestamp,
+ /*is_first=*/0,
+ /*is_last=*/0),
+ RtpSequenceNumberMap::Info(
+ /*timestamp=*/2 - kStartTimestamp,
+ /*is_first=*/0,
+ /*is_last=*/1)));
+}
+
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index dc888f9..fafd3ca 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -25,6 +25,7 @@
constexpr uint32_t kTimestampTicksPerMs = 90;
constexpr int kSendSideDelayWindowMs = 1000;
constexpr int kBitrateStatisticsWindowMs = 1000;
+constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
bool IsEnabled(absl::string_view name,
const WebRtcKeyValueConfig* field_trials) {
@@ -67,6 +68,7 @@
transport_(config.outgoing_transport),
event_log_(config.event_log),
is_audio_(config.audio),
+ need_rtp_packet_infos_(config.need_rtp_packet_infos),
transport_feedback_observer_(config.transport_feedback_callback),
send_side_delay_observer_(config.send_side_delay_observer),
send_packet_observer_(config.send_packet_observer),
@@ -75,14 +77,18 @@
bitrate_callback_(config.send_bitrate_observer),
media_has_been_sent_(false),
force_part_of_allocation_(false),
+ timestamp_offset_(0),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
rtp_overhead_bytes_per_packet_(0),
total_bitrate_sent_(kBitrateStatisticsWindowMs,
RateStatistics::kBpsScale),
- nack_bitrate_sent_(kBitrateStatisticsWindowMs,
- RateStatistics::kBpsScale) {}
+ nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
+ rtp_sequence_number_map_(need_rtp_packet_infos_
+ ? std::make_unique<RtpSequenceNumberMap>(
+ kRtpSequenceNumberMapMaxEntries)
+ : nullptr) {}
void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) {
@@ -113,6 +119,20 @@
{
rtc::CritScope lock(&lock_);
options.included_in_allocation = force_part_of_allocation_;
+
+ if (need_rtp_packet_infos_ &&
+ packet->packet_type() == RtpPacketToSend::Type::kVideo) {
+ RTC_DCHECK(rtp_sequence_number_map_);
+ // Last packet of a frame, add it to sequence number info map.
+ const uint32_t timestamp = packet->Timestamp() - timestamp_offset_;
+ bool is_first_packet_of_frame = packet->is_first_packet_of_frame();
+ bool is_last_packet_of_frame = packet->Marker();
+
+ rtp_sequence_number_map_->InsertPacket(
+ packet->SequenceNumber(),
+ RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
+ is_last_packet_of_frame));
+ }
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
@@ -226,6 +246,35 @@
media_has_been_sent_ = media_sent;
}
+void RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
+ rtc::CritScope lock(&lock_);
+ timestamp_offset_ = timestamp;
+}
+
+std::vector<RtpSequenceNumberMap::Info> RtpSenderEgress::GetSentRtpPacketInfos(
+ rtc::ArrayView<const uint16_t> sequence_numbers) const {
+ RTC_DCHECK(!sequence_numbers.empty());
+ if (!need_rtp_packet_infos_) {
+ return std::vector<RtpSequenceNumberMap::Info>();
+ }
+
+ std::vector<RtpSequenceNumberMap::Info> results;
+ results.reserve(sequence_numbers.size());
+
+ rtc::CritScope cs(&lock_);
+ for (uint16_t sequence_number : sequence_numbers) {
+ const auto& 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.
+ return std::vector<RtpSequenceNumberMap::Info>();
+ }
+ results.push_back(*info);
+ }
+
+ return results;
+}
+
bool RtpSenderEgress::HasCorrectSsrc(const RtpPacketToSend& packet) const {
switch (*packet.packet_type()) {
case RtpPacketMediaType::kAudio:
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index e72a0cf..3d4999f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -23,6 +23,7 @@
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/rate_statistics.h"
#include "rtc_base/thread_annotations.h"
@@ -64,6 +65,15 @@
void ForceIncludeSendPacketsInAllocation(bool part_of_allocation);
bool MediaHasBeenSent() const;
void SetMediaHasBeenSent(bool media_sent);
+ void SetTimestampOffset(uint32_t timestamp);
+
+ // For each sequence number in |sequence_number|, recall the last RTP packet
+ // which bore it - its timestamp and whether it was the first and/or last
+ // packet in that frame. If all of the given sequence numbers could be
+ // recalled, return a vector with all of them (in corresponding order).
+ // If any could not be recalled, return an empty vector.
+ std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
+ rtc::ArrayView<const uint16_t> sequence_numbers) const;
private:
// Maps capture time in milliseconds to send-side delay in milliseconds.
@@ -100,6 +110,7 @@
Transport* const transport_;
RtcEventLog* const event_log_;
const bool is_audio_;
+ const bool need_rtp_packet_infos_;
TransportFeedbackObserver* const transport_feedback_observer_;
SendSideDelayObserver* const send_side_delay_observer_;
@@ -111,6 +122,7 @@
rtc::CriticalSection lock_;
bool media_has_been_sent_ RTC_GUARDED_BY(lock_);
bool force_part_of_allocation_ RTC_GUARDED_BY(lock_);
+ uint32_t timestamp_offset_ RTC_GUARDED_BY(lock_);
SendDelayMap send_delays_ RTC_GUARDED_BY(lock_);
SendDelayMap::const_iterator max_delay_it_ RTC_GUARDED_BY(lock_);
@@ -122,6 +134,13 @@
StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_);
RateStatistics total_bitrate_sent_ RTC_GUARDED_BY(lock_);
RateStatistics nack_bitrate_sent_ RTC_GUARDED_BY(lock_);
+
+ // Maps sent packets' sequence numbers to a tuple consisting of:
+ // 1. The timestamp, without the randomizing offset mandated by the RFC.
+ // 2. Whether the packet was the first in its frame.
+ // 3. Whether the packet was the last in its frame.
+ const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
+ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 99fb822..532e069 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -41,7 +41,6 @@
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
@@ -253,7 +252,6 @@
PlayoutDelayOracle* playout_delay_oracle,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
- bool need_rtp_packet_infos,
bool enable_retransmit_all_layers,
const WebRtcKeyValueConfig& field_trials)
: RTPSenderVideo([&] {
@@ -263,7 +261,6 @@
config.flexfec_sender = flexfec_sender;
config.frame_encryptor = frame_encryptor;
config.require_frame_encryption = require_frame_encryption;
- config.need_rtp_packet_infos = need_rtp_packet_infos;
config.enable_retransmit_all_layers = enable_retransmit_all_layers;
config.field_trials = &field_trials;
return config;
@@ -280,10 +277,6 @@
transmit_color_space_next_frame_(false),
current_playout_delay_{-1, -1},
playout_delay_pending_(false),
- rtp_sequence_number_map_(config.need_rtp_packet_infos
- ? std::make_unique<RtpSequenceNumberMap>(
- kRtpSequenceNumberMapMaxEntries)
- : nullptr),
red_payload_type_(config.red_payload_type),
ulpfec_payload_type_(config.ulpfec_payload_type),
flexfec_sender_(config.flexfec_sender),
@@ -729,7 +722,6 @@
if (num_packets == 0)
return false;
- uint16_t first_sequence_number;
bool first_frame = first_frame_sent_();
std::vector<std::unique_ptr<RtpPacketToSend>> rtp_packets;
for (size_t i = 0; i < num_packets; ++i) {
@@ -753,16 +745,14 @@
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;
- if (rtp_sequence_number_map_ && i == 0) {
- first_sequence_number = packet->SequenceNumber();
- }
-
// No FEC protection for upper temporal layers, if used.
bool protect_packet = temporal_id == 0 || temporal_id == kNoTemporalIdx;
@@ -802,13 +792,6 @@
}
}
- 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);
- }
-
LogAndSendToNetwork(std::move(rtp_packets), unpacketized_payload_size);
TRACE_EVENT_ASYNC_END1("webrtc", "Video", capture_time_ms, "timestamp",
@@ -832,37 +815,6 @@
.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;
-}
-
bool RTPSenderVideo::AllowRetransmission(
uint8_t temporal_id,
int32_t retransmission_settings,
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index 5f01803..2df62ad 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -28,7 +28,6 @@
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
-#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/rtp_rtcp/source/ulpfec_generator.h"
#include "rtc_base/critical_section.h"
@@ -74,7 +73,6 @@
PlayoutDelayOracle* playout_delay_oracle = nullptr;
FrameEncryptorInterface* frame_encryptor = nullptr;
bool require_frame_encryption = false;
- bool need_rtp_packet_infos = false;
bool enable_retransmit_all_layers = false;
absl::optional<int> red_payload_type;
absl::optional<int> ulpfec_payload_type;
@@ -90,7 +88,6 @@
PlayoutDelayOracle* playout_delay_oracle,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
- bool need_rtp_packet_infos,
bool enable_retransmit_all_layers,
const WebRtcKeyValueConfig& field_trials);
virtual ~RTPSenderVideo();
@@ -129,14 +126,6 @@
// or extension/
uint32_t PacketizationOverheadBps() const;
- // For each sequence number in |sequence_number|, recall the last RTP packet
- // which bore it - its timestamp and whether it was the first and/or last
- // packet in that frame. If all of the given sequence numbers could be
- // recalled, return a vector with all of them (in corresponding order).
- // If any could not be recalled, return an empty vector.
- std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
- rtc::ArrayView<const uint16_t> sequence_numbers) const;
-
protected:
static uint8_t GetTemporalId(const RTPVideoHeader& header);
bool AllowRetransmission(uint8_t temporal_id,
@@ -208,13 +197,6 @@
// Should never be held when calling out of this class.
rtc::CriticalSection crit_;
- // Maps sent packets' sequence numbers to a tuple consisting of:
- // 1. The timestamp, without the randomizing offset mandated by the RFC.
- // 2. Whether the packet was the first in its frame.
- // 3. Whether the packet was the last in its frame.
- const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_
- RTC_PT_GUARDED_BY(crit_);
-
// RED/ULPFEC.
const absl::optional<int> red_payload_type_;
const absl::optional<int> ulpfec_payload_type_;
