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Default-enable deferred sequence numbering for audio.

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Bug: webrtc:11340
Change-Id: I5aa2a1e35b007c6d4c039f42f09c48fd7871f6ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227775
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@{#34681}
This commit is contained in:
Erik Språng 2021-08-06 15:22:28 +02:00 committed by WebRTC LUCI CQ
parent 55542302b3
commit 2373bb9799
3 changed files with 11 additions and 3 deletions
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2
audio/channel_send.cc
View File

@ -502,6 +502,8 @@ ChannelSend::ChannelSend(
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);

4
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
View File

@ -384,7 +384,9 @@ bool ModuleRtpRtcpImpl2::TrySendPacket(RtpPacketToSend* packet,
RTC_DCHECK(rtp_sender_);
RTC_DCHECK_RUN_ON(&pacer_thread_checker_);
if (rtp_sender_->deferred_sequencing_) {
RTC_DCHECK(rtp_sender_->packet_generator.SendingMedia());
if (!rtp_sender_->packet_generator.SendingMedia()) {
return false;
}
if (packet->packet_type() == RtpPacketMediaType::kPadding &&
packet->Ssrc() == rtp_sender_->packet_generator.SSRC() &&
!rtp_sender_->sequencer_.CanSendPaddingOnMediaSsrc()) {

8
modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -304,8 +304,10 @@ bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
return false;
memcpy(payload, payload_data, payload_size);
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
if (!rtp_sender_->deferred_sequence_numbering() &&
!rtp_sender_->AssignSequenceNumber(packet.get())) {
return false;
}
{
MutexLock lock(&send_audio_mutex_);
@ -374,8 +376,10 @@ bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
packet->SetSsrc(rtp_sender_->SSRC());
packet->SetTimestamp(dtmf_timestamp);
packet->set_capture_time_ms(clock_->TimeInMilliseconds());
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
if (!rtp_sender_->deferred_sequence_numbering() &&
!rtp_sender_->AssignSequenceNumber(packet.get())) {
return false;
}
// Create DTMF data.
uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize);