webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc - Issue 1362503003: Use suffixed {uint,int}{8,16,32,64}_t types.

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/voice_engine/test/auto_test/fakes/conference_transport.cc

Issue 1362503003: Use suffixed {uint,int}{8,16,32,64}_t types. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: rebase + revert basictypes.h (to be landed separately just in case of a revert due to unexpected us… Created 5 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:

Index: webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

index 581a7685b29e72ac47fb9b35c3c5b4c0ed0ba8f9..06770939598e0d27c7b19bb5029825029b99cb2f 100644

--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

@@ -207,8 +207,8 @@ bool ConferenceTransport::DispatchPackets() {

packet_queue_.pop_front();

}

- int32 elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);

- int32 sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;

+ int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);

+ int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;

if (sleep_ms > 0) {

// Every packet should be delayed by half of RTT.

webrtc::SleepMs(sleep_ms);