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());
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 057cb5a..f6ce463 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -305,7 +305,7 @@
   //    -round_trip_time_ms : estimate of the round-trip-time (in milliseconds).
   // Return value           : list of packets to be retransmitted.
   //
-  std::vector<uint16_t> GetNackList(int round_trip_time_ms) const;
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
 
   //
   // Get statistics of calls to GetAudio().
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index cbea050..4aa372f 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -2017,7 +2017,7 @@
 }
 
 std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
-    int round_trip_time_ms) const {
+    int64_t round_trip_time_ms) const {
   return receiver_.GetNackList(round_trip_time_ms);
 }
 
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 2d0f767..a06d877 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -246,7 +246,7 @@
   virtual void DisableNack() OVERRIDE;
 
   virtual std::vector<uint16_t> GetNackList(
-      int round_trip_time_ms) const OVERRIDE;
+      int64_t round_trip_time_ms) const OVERRIDE;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* stats) const OVERRIDE;
diff --git a/webrtc/modules/audio_coding/main/acm2/nack.cc b/webrtc/modules/audio_coding/main/acm2/nack.cc
index 7265fe6..4324cd2 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack.cc
+++ b/webrtc/modules/audio_coding/main/acm2/nack.cc
@@ -207,13 +207,13 @@
   nack_list_.erase(nack_list_.begin(), nack_list_.upper_bound(limit));
 }
 
-int Nack::TimeToPlay(uint32_t timestamp) const {
+int64_t Nack::TimeToPlay(uint32_t timestamp) const {
   uint32_t timestamp_increase = timestamp - timestamp_last_decoded_rtp_;
   return timestamp_increase / sample_rate_khz_;
 }
 
 // We don't erase elements with time-to-play shorter than round-trip-time.
-std::vector<uint16_t> Nack::GetNackList(int round_trip_time_ms) const {
+std::vector<uint16_t> Nack::GetNackList(int64_t round_trip_time_ms) const {
   std::vector<uint16_t> sequence_numbers;
   for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end();
       ++it) {
diff --git a/webrtc/modules/audio_coding/main/acm2/nack.h b/webrtc/modules/audio_coding/main/acm2/nack.h
index 3809327..d74bb1f 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack.h
+++ b/webrtc/modules/audio_coding/main/acm2/nack.h
@@ -87,7 +87,7 @@
   // Get a list of "missing" packets which have expected time-to-play larger
   // than the given round-trip-time (in milliseconds).
   // Note: Late packets are not included.
-  std::vector<uint16_t> GetNackList(int round_trip_time_ms) const;
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
 
   // Reset to default values. The NACK list is cleared.
   // |nack_threshold_packets_| & |max_nack_list_size_| preserve their values.
@@ -98,7 +98,7 @@
   FRIEND_TEST_ALL_PREFIXES(NackTest, EstimateTimestampAndTimeToPlay);
 
   struct NackElement {
-    NackElement(int initial_time_to_play_ms,
+    NackElement(int64_t initial_time_to_play_ms,
                 uint32_t initial_timestamp,
                 bool missing)
         : time_to_play_ms(initial_time_to_play_ms),
@@ -107,7 +107,7 @@
 
     // Estimated time (ms) left for this packet to be decoded. This estimate is
     // updated every time jitter buffer decodes a packet.
-    int time_to_play_ms;
+    int64_t time_to_play_ms;
 
     // A guess about the timestamp of the missing packet, it is used for
     // estimation of |time_to_play_ms|. The estimate might be slightly wrong if
@@ -171,7 +171,7 @@
   uint32_t EstimateTimestamp(uint16_t sequence_number);
 
   // Compute time-to-play given a timestamp.
-  int TimeToPlay(uint32_t timestamp) const;
+  int64_t TimeToPlay(uint32_t timestamp) const;
 
   // If packet N is arrived, any packet prior to N - |nack_threshold_packets_|
   // which is not arrived is considered missing, and should be in NACK list.
diff --git a/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc b/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
index 7863c75..c175908 100644
--- a/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/nack_unittest.cc
@@ -29,7 +29,7 @@
 const int kSampleRateHz = 16000;
 const int kPacketSizeMs = 30;
 const uint32_t kTimestampIncrement = 480;  // 30 ms.
-const int kShortRoundTripTimeMs = 1;
+const int64_t kShortRoundTripTimeMs = 1;
 
 bool IsNackListCorrect(const std::vector<uint16_t>& nack_list,
                        const uint16_t* lost_sequence_numbers,
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index 83e6dce..8826194 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -991,7 +991,8 @@
   // Negative |round_trip_time_ms| results is an error message and empty list
   // is returned.
   //
-  virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
+  virtual std::vector<uint16_t> GetNackList(
+      int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;