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Removes rtp level keep alive support.

Browse Source 
This is not used in practice as there's functionality on
other levels that serves the same purpose.

Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
This commit is contained in:
Sebastian Jansson
2019-03-04 18:59:32 +01:00
committed by Commit Bot
parent 9ffb5df04e
commit d155d686f8
17 changed files with 4 additions and 200 deletions
Download Patch File Download Diff File Expand all files Collapse all files

15
modules/rtp_rtcp/source/rtp_sender.cc
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@ -1229,21 +1229,6 @@ int64_t RTPSender::LastTimestampTimeMs() const {
return last_timestamp_time_ms_;
}
void RTPSender::SendKeepAlive(uint8_t payload_type) {
std::unique_ptr<RtpPacketToSend> packet = AllocatePacket();
packet->SetPayloadType(payload_type);
// Set marker bit and timestamps in the same manner as plain padding packets.
packet->SetMarker(false);
{
rtc::CritScope lock(&send_critsect_);
packet->SetTimestamp(last_rtp_timestamp_);
packet->set_capture_time_ms(capture_time_ms_);
}
AssignSequenceNumber(packet.get());
SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
RtpPacketSender::Priority::kLowPriority);
}
void RTPSender::SetRtt(int64_t rtt_ms) {
packet_history_.SetRtt(rtt_ms);
flexfec_packet_history_.SetRtt(rtt_ms);