Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(391)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/call/rampup_tests.h

Issue 2165743003: Variable audio bitrate. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 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
« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/call/call.cc ('k') | webrtc/call/rampup_tests.cc » ('j') | webrtc/call/rampup_tests.cc » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/call/rampup_tests.h
diff --git a/webrtc/call/rampup_tests.h b/webrtc/call/rampup_tests.h
index fa73c63f9aba0ae0af0f5139ca212d3fd732812c..8179512d439f26e0c2a8f1e05aa646946eced491 100644
--- a/webrtc/call/rampup_tests.h
+++ b/webrtc/call/rampup_tests.h
@@ -64,6 +64,7 @@ class RampUpTester : public test::EndToEndTest {
const size_t num_audio_streams_;
const bool rtx_;
const bool red_;
+ Call* sender_call_ ;
stefan-webrtc 2016/07/20 10:14:59 Remove spaces at the end.
mflodman 2016/07/22 13:50:29 Done.
VideoSendStream* send_stream_;
test::PacketTransport* send_transport_;
@@ -99,7 +100,6 @@ class RampUpTester : public test::EndToEndTest {
SsrcMap rtx_ssrc_map_;
rtc::PlatformThread poller_thread_;
- Call* sender_call_;
};
class RampUpDownUpTester : public RampUpTester {
« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/call/call.cc ('k') | webrtc/call/rampup_tests.cc » ('j') | webrtc/call/rampup_tests.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698