webrtc/modules/rtp_rtcp/source/rtp_sender.h - Issue 2249223005: Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData

Keyboard Shortcuts

	File
u :	up to issue
j / k :	jump to file after / before current file
J / K :	jump to next file with a comment after / before current file
	Side-by-side diff
i :	toggle intra-line diffs
e :	expand all comments
c :	collapse all comments
s :	toggle showing all comments
n / p :	next / previous diff chunk or comment
N / P :	next / previous comment
<Up> / <Down> :	next / previous line

	Issue
u :	up to list of issues
j / k :	jump to patch after / before current patch
o / <Enter> :	open current patch in side-by-side view
i :	open current patch in unified diff view

	Issue List
j / k :	jump to issue after / before current issue
o / <Enter> :	open current issue

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.h

Issue 2249223005: Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 4 years, 4 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Download patch

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h

index 540c6f18ec0bc7509138e82e95ad4100f5374bde..e1dfa9bb32b84fa1adba139eb9c892d624a8276e 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h

@@ -222,8 +222,6 @@ class RTPSender {

uint16_t AllocateSequenceNumber(uint16_t packets_to_send);

size_t MaxPayloadLength() const;

- // Current timestamp.

- uint32_t Timestamp() const;

uint32_t SSRC() const;

// Deprecated. Create RtpPacketToSend instead and use next function.

@@ -410,7 +408,7 @@ class RTPSender {

uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);

bool ssrc_forced_ GUARDED_BY(send_critsect_);

uint32_t ssrc_ GUARDED_BY(send_critsect_);

- uint32_t timestamp_ GUARDED_BY(send_critsect_);

+ uint32_t last_rtp_timestamp_ GUARDED_BY(send_critsect_);

int64_t capture_time_ms_ GUARDED_BY(send_critsect_);

int64_t last_timestamp_time_ms_ GUARDED_BY(send_critsect_);

bool media_has_been_sent_ GUARDED_BY(send_critsect_);

« no previous file with comments | « no previous file | webrtc/modules/rtp_rtcp/source/rtp_sender.cc » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc » ('J')