Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index f0531ed..07c2784 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -15,6 +15,7 @@
#include <algorithm> // sort
#include <vector>
+#include "webrtc/base/format_macros.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
@@ -727,12 +728,12 @@
}
std::vector<uint16_t> AcmReceiver::GetNackList(
- int round_trip_time_ms) const {
+ int64_t round_trip_time_ms) const {
CriticalSectionScoped lock(crit_sect_.get());
if (round_trip_time_ms < 0) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
"GetNackList: round trip time cannot be negative."
- " round_trip_time_ms=%d", round_trip_time_ms);
+ " round_trip_time_ms=%" PRId64, round_trip_time_ms);
}
if (nack_enabled_ && round_trip_time_ms >= 0) {
assert(nack_.get());
