Remove last traces of deferred sequencing.
Bug: webrtc:11340
Change-Id: I761be67d42959192355f9f6f54ed1f735da1fe96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228646
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34770}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 37a14b7..6fd2161 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -502,8 +502,6 @@
configuration.rtcp_packet_type_counter_observer = this;
configuration.local_media_ssrc = ssrc;
- configuration.use_deferred_sequencing =
- !field_trial::IsDisabled("WebRTC-Audio-DeferredSequencing");
rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
rtp_rtcp_->SetSendingMediaStatus(false);
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index f95c743..b778563 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -231,8 +231,6 @@
crypto_options.sframe.require_frame_encryption;
configuration.extmap_allow_mixed = rtp_config.extmap_allow_mixed;
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
- configuration.use_deferred_sequencing = !absl::StartsWith(
- trials.Lookup("WebRTC-Video-DeferredSequencing"), "Disabled");
configuration.field_trials = &trials;
std::vector<RtpStreamSender> rtp_streams;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index c1a44fa..df5593c 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -146,10 +146,6 @@
// https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
bool non_sender_rtt_measurement = false;
- // If true, sequence numbers are not assigned until after the pacer stage,
- // in RtpSenderEgress.
- bool use_deferred_sequencing = false;
-
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
};
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index f2a74e1..89c5597 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -86,7 +86,6 @@
!IsTrialSetTo(config.field_trials,
"WebRTC-SendSideBwe-WithOverhead",
"Disabled")),
- deferred_sequencing_(config.use_deferred_sequencing),
clock_(config.clock),
packet_history_(packet_history),
transport_(config.outgoing_transport),
@@ -134,23 +133,19 @@
RTC_DCHECK_RUN_ON(&pacer_checker_);
RTC_DCHECK(packet);
- if (deferred_sequencing_) {
- // Strict increasing of sequence numbers can only be guaranteed with
- // deferred sequencing due to raciness with the pacer.
- if (packet->Ssrc() == ssrc_ &&
- packet->packet_type() != RtpPacketMediaType::kRetransmission) {
- if (last_sent_seq_.has_value()) {
- RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_seq_ + 1),
- packet->SequenceNumber());
- }
- last_sent_seq_ = packet->SequenceNumber();
- } else if (packet->Ssrc() == rtx_ssrc_) {
- if (last_sent_rtx_seq_.has_value()) {
- RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_rtx_seq_ + 1),
- packet->SequenceNumber());
- }
- last_sent_rtx_seq_ = packet->SequenceNumber();
+ if (packet->Ssrc() == ssrc_ &&
+ packet->packet_type() != RtpPacketMediaType::kRetransmission) {
+ if (last_sent_seq_.has_value()) {
+ RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_seq_ + 1),
+ packet->SequenceNumber());
}
+ last_sent_seq_ = packet->SequenceNumber();
+ } else if (packet->Ssrc() == rtx_ssrc_) {
+ if (last_sent_rtx_seq_.has_value()) {
+ RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_rtx_seq_ + 1),
+ packet->SequenceNumber());
+ }
+ last_sent_rtx_seq_ = packet->SequenceNumber();
}
RTC_DCHECK(packet->packet_type().has_value());
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index bf9076d..1fd7128 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -134,7 +134,6 @@
const absl::optional<uint32_t> flexfec_ssrc_;
const bool populate_network2_timestamp_;
const bool send_side_bwe_with_overhead_;
- const bool deferred_sequencing_;
Clock* const clock_;
RtpPacketHistory* const packet_history_;
Transport* const transport_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
index 4f3990c..d95a639 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -587,6 +587,7 @@
// Send a media packet.
std::unique_ptr<RtpPacketToSend> media_packet = BuildRtpPacket();
media_packet->SetPayloadSize(6);
+ media_packet->SetSequenceNumber(kStartSequenceNumber);
expected_transmitted_counter.packets += 1;
expected_transmitted_counter.payload_bytes += media_packet->payload_size();
expected_transmitted_counter.header_bytes += media_packet->headers_size();
@@ -606,8 +607,9 @@
// and retransmitted packet statistics.
std::unique_ptr<RtpPacketToSend> retransmission_packet = BuildRtpPacket();
retransmission_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+ retransmission_packet->SetSequenceNumber(kStartSequenceNumber);
retransmission_packet->set_retransmitted_sequence_number(
- retransmission_packet->SequenceNumber());
+ kStartSequenceNumber);
media_packet->SetPayloadSize(7);
expected_transmitted_counter.packets += 1;
expected_transmitted_counter.payload_bytes +=
@@ -636,6 +638,7 @@
std::unique_ptr<RtpPacketToSend> padding_packet = BuildRtpPacket();
padding_packet->set_packet_type(RtpPacketMediaType::kPadding);
padding_packet->SetPadding(224);
+ padding_packet->SetSequenceNumber(kStartSequenceNumber + 1);
expected_transmitted_counter.packets += 1;
expected_transmitted_counter.padding_bytes += padding_packet->padding_size();
expected_transmitted_counter.header_bytes += padding_packet->headers_size();
