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Replace labs with std::abs.

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Resolves clang 3.5 warnings on OS X for -Wabsolute-value.

BUG=chromium:351479
R=andrew@webrtc.org, thakis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/9869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5692 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2014-03-13 08:53:39 +00:00
parent 827faae0ec
commit b5f3029302
2 changed files with 3 additions and 3 deletions
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4
webrtc/modules/video_coding/main/source/receiver.cc
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@ -156,12 +156,12 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(
// Assume that render timing errors are due to changes in the video stream. // Assume that render timing errors are due to changes in the video stream.
if (next_render_time_ms < 0) { if (next_render_time_ms < 0) {
timing_error = true; timing_error = true;
} else if (labs(next_render_time_ms - now_ms) > max_video_delay_ms_) { } else if (std::abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_), VCMId(vcm_id_, receiver_id_),
"This frame is out of our delay bounds, resetting jitter " "This frame is out of our delay bounds, resetting jitter "
"buffer: %d > %d", "buffer: %d > %d",
static_cast<int>(labs(next_render_time_ms - now_ms)), static_cast<int>(std::abs(next_render_time_ms - now_ms)),
max_video_delay_ms_); max_video_delay_ms_);
timing_error = true; timing_error = true;
} else if (static_cast<int>(timing_->TargetVideoDelay()) > } else if (static_cast<int>(timing_->TargetVideoDelay()) >

2
webrtc/video/call_perf_tests.cc
View File

@ -196,7 +196,7 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
// estimated as being synchronized. We don't want to trigger on those. // estimated as being synchronized. We don't want to trigger on those.
if (time_since_creation < kStartupTimeMs) if (time_since_creation < kStartupTimeMs)
return; return;
if (labs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) { if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
if (first_time_in_sync_ == -1) { if (first_time_in_sync_ == -1) {
first_time_in_sync_ = now_ms; first_time_in_sync_ = now_ms;
webrtc::test::PrintResult("sync_convergence_time", webrtc::test::PrintResult("sync_convergence_time",