From 83afeebe9447c1153edb79634a54fc6b5dd0b20e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Erik=20Spr=C3=A5ng?= Date: Tue, 14 May 2019 15:57:19 +0200 Subject: [PATCH] Remove redundant capture time adjustment in RtpSender MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit webrtc::RealTimeClock::TimeInMilliseconds() and rtc::TimeMillis() have for some time been backed by the same clock, no need for adjustment. Bug: None Change-Id: I5962153d9f5aa5e58ccde26393c322972cb51d43 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136808 Reviewed-by: Danil Chapovalov Commit-Queue: Erik Språng Cr-Commit-Position: refs/heads/master@{#27939} --- modules/rtp_rtcp/source/rtp_sender.cc | 14 +++----------- modules/rtp_rtcp/source/rtp_sender.h | 1 - 2 files changed, 3 insertions(+), 12 deletions(-) diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc index acc2f15b36..56d42aad90 100644 --- a/modules/rtp_rtcp/source/rtp_sender.cc +++ b/modules/rtp_rtcp/source/rtp_sender.cc @@ -106,8 +106,6 @@ RTPSender::RTPSender( bool extmap_allow_mixed, const WebRtcKeyValueConfig& field_trials) : clock_(clock), - // TODO(holmer): Remove this conversion? - clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()), random_(clock_->TimeInMicroseconds()), audio_configured_(audio), flexfec_ssrc_(flexfec_ssrc), @@ -474,13 +472,9 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) { return 0; } - // Convert from TickTime to Clock since capture_time_ms is based on - // TickTime. - int64_t corrected_capture_tims_ms = - stored_packet->capture_time_ms + clock_delta_ms_; paced_sender_->InsertPacket( RtpPacketSender::kNormalPriority, stored_packet->ssrc, - stored_packet->rtp_sequence_number, corrected_capture_tims_ms, + stored_packet->rtp_sequence_number, stored_packet->capture_time_ms, stored_packet->packet_size, true); return packet_size; @@ -690,9 +684,7 @@ bool RTPSender::SendToNetwork(std::unique_ptr packet, uint32_t ssrc = packet->Ssrc(); if (paced_sender_) { uint16_t seq_no = packet->SequenceNumber(); - // Correct offset between implementations of millisecond time stamps in - // TickTime and Clock. - int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_; + int64_t capture_time_ms = packet->capture_time_ms(); size_t packet_size = send_side_bwe_with_overhead_ ? packet->size() : packet->payload_size(); if (ssrc == FlexfecSsrc()) { @@ -704,7 +696,7 @@ bool RTPSender::SendToNetwork(std::unique_ptr packet, packet_history_.PutRtpPacket(std::move(packet), storage, absl::nullopt); } - paced_sender_->InsertPacket(priority, ssrc, seq_no, corrected_time_ms, + paced_sender_->InsertPacket(priority, ssrc, seq_no, capture_time_ms, packet_size, false); return true; } diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h index 0dae94a339..66b821fc26 100644 --- a/modules/rtp_rtcp/source/rtp_sender.h +++ b/modules/rtp_rtcp/source/rtp_sender.h @@ -225,7 +225,6 @@ class RTPSender : public AcknowledgedPacketsObserver { void UpdateRtpOverhead(const RtpPacketToSend& packet); Clock* const clock_; - const int64_t clock_delta_ms_; Random random_ RTC_GUARDED_BY(send_critsect_); const bool audio_configured_;