Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(152)

Unified Diff: webrtc/video/audio_receive_stream.cc

Issue 1360913004: Fix BWE bug where audio has timestamps in us. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 5 years, 3 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
« no previous file with comments | « no previous file | webrtc/video/audio_receive_stream_unittest.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/video/audio_receive_stream.cc
diff --git a/webrtc/video/audio_receive_stream.cc b/webrtc/video/audio_receive_stream.cc
index b8da1bb6c35da0bd984b9c0a74f6f7b04180bbc4..a1cf2ca33ef07e8357aa23898e28d45f39fca3dc 100644
--- a/webrtc/video/audio_receive_stream.cc
+++ b/webrtc/video/audio_receive_stream.cc
@@ -102,7 +102,7 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
header.extension.hasAbsoluteSendTime) {
int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
if (packet_time.timestamp >= 0)
- arrival_time_ms = packet_time.timestamp;
+ arrival_time_ms = (packet_time.timestamp + 500) / 1000;
size_t payload_size = length - header.headerLength;
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
header, false);
« no previous file with comments | « no previous file | webrtc/video/audio_receive_stream_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698