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Remove redundant capture time adjustment in RtpSender

Browse Source 
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 <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27939}
This commit is contained in:
Erik Språng 2019-05-14 15:57:19 +02:00 committed by Commit Bot
parent 3b9aa66c7b
commit 83afeebe94
2 changed files with 3 additions and 12 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

14
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -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<RtpPacketToSend> 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<RtpPacketToSend> 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;
}

1
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -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_;