Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(48)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/call/call.cc

Issue 2415943002: Revert of Add RtcpRttStats to AudioStream (Closed)
Patch Set: Created 4 years, 2 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 | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index a68a60e120c709f25e1bbd5f2c62d0794b2e5ce6..9515ac10f14efe7c87ae2708a42cf49d55ba5452 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -373,7 +373,7 @@
event_log_->LogAudioSendStreamConfig(config);
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
- bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
+ bitrate_allocator_.get(), event_log_);
{
WriteLockScoped write_lock(*send_crit_);
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
« no previous file with comments | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698