Index: webrtc/modules/video_coding/test/video_rtp_play.cc |
diff --git a/webrtc/modules/video_coding/test/video_rtp_play.cc b/webrtc/modules/video_coding/test/video_rtp_play.cc |
index 0a6b7d13e8bea23210a9d413618b8e8bf640f5af..cb092e381e0e7fd0ac87d3182b678ea0e7d30e6e 100644 |
--- a/webrtc/modules/video_coding/test/video_rtp_play.cc |
+++ b/webrtc/modules/video_coding/test/video_rtp_play.cc |
@@ -48,9 +48,9 @@ int RtpPlay(const CmdArgs& args) { |
output_file = webrtc::test::OutputPath() + "RtpPlay_decoded.yuv"; |
webrtc::SimulatedClock clock(0); |
- webrtc::rtpplayer::VcmPayloadSinkFactory factory(output_file, &clock, |
- kConfigProtectionEnabled, kConfigProtectionMethod, kConfigRttMs, |
- kConfigRenderDelayMs, kConfigMinPlayoutDelayMs); |
+ webrtc::rtpplayer::VcmPayloadSinkFactory factory( |
+ output_file, &clock, kConfigProtectionEnabled, kConfigProtectionMethod, |
+ kConfigRttMs, kConfigRenderDelayMs, kConfigMinPlayoutDelayMs); |
rtc::scoped_ptr<webrtc::rtpplayer::RtpPlayerInterface> rtp_player( |
webrtc::rtpplayer::Create(args.inputFile, &factory, &clock, payload_types, |
kConfigLossRate, kConfigRttMs, |
@@ -63,7 +63,7 @@ int RtpPlay(const CmdArgs& args) { |
while ((ret = rtp_player->NextPacket(clock.TimeInMilliseconds())) == 0) { |
ret = factory.DecodeAndProcessAll(true); |
if (ret < 0 || (kConfigMaxRuntimeMs > -1 && |
- clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) { |
+ clock.TimeInMilliseconds() >= kConfigMaxRuntimeMs)) { |
break; |
} |
clock.AdvanceTimeMilliseconds(1); |