Fix build breakage introduce with r5665.
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5666 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index 13a15c7..461c3863 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -9,6 +9,7 @@
*/
#include <stdio.h>
+#include <sstream>
#include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
@@ -58,7 +59,9 @@
if (header.extension.transmissionTimeOffset != 0)
++non_zero_ts_offsets;
if (arrival_time_only) {
- fprintf(stdout, "%ld\n", static_cast<int64_t>(time_ms) * 1000000);
+ std::stringstream ss;
+ ss << static_cast<int64_t>(time_ms) * 1000000;
+ fprintf(stdout, "%s\n", ss.str().c_str());
} else {
fprintf(stdout, "%u %u %d %u %u %d %u %u\n", header.sequenceNumber,
header.timestamp, header.extension.transmissionTimeOffset,