From 737e073f8da15cd05db43ca7da22ba030e14d27a Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Thu, 19 Oct 2017 09:00:17 +0200
Subject: [PATCH] Fixing warning C4267 on Win (more_configs).

This is a follow-up of https://webrtc-review.googlesource.com/c/src/+/12921.

Bug: chromium:759980
Change-Id: Ifd39adb6541c0c7e0337f587a8b34b84a07331ed
Reviewed-on: https://webrtc-review.googlesource.com/13122
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20341}
---
 .../acm2/acm_receiver_unittest.cc             |  9 +++++----
 .../acm2/audio_coding_module_unittest.cc      | 15 ++++++++------
 .../bitrate_controller_unittest.cc            | 20 +++++++++++--------
 .../builtin_audio_encoder_factory_unittest.cc |  5 +++--
 .../codecs/cng/audio_encoder_cng_unittest.cc  |  4 +++-
 modules/audio_coding/codecs/isac/unittest.cc  |  7 +++++--
 .../legacy_encoded_audio_frame_unittest.cc    |  6 ++++--
 .../audio_coding/codecs/opus/opus_unittest.cc |  7 +++++--
 .../red/audio_encoder_copy_red_unittest.cc    |  3 ++-
 .../neteq/audio_multi_vector_unittest.cc      |  3 ++-
 .../neteq/audio_vector_unittest.cc            | 10 ++++++----
 modules/audio_coding/neteq/expand_unittest.cc |  4 ++--
 .../audio_coding/neteq/neteq_impl_unittest.cc |  4 ++--
 .../neteq/red_payload_splitter_unittest.cc    |  4 +++-
 .../neteq/sync_buffer_unittest.cc             |  7 ++++---
 .../neteq/tools/input_audio_file_unittest.cc  |  5 +++--
 16 files changed, 70 insertions(+), 43 deletions(-)

diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 8fbea84413..c420c7eb6f 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -108,7 +108,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
     last_packet_send_timestamp_ = timestamp_;
     while (!packet_sent_) {
       frame.timestamp_ = timestamp_;
-      timestamp_ += frame.samples_per_channel_;
+      timestamp_ += rtc::checked_cast<uint32_t>(frame.samples_per_channel_);
       ASSERT_GE(acm_->Add10MsData(frame), 0);
     }
   }
@@ -175,8 +175,9 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
   for (size_t n = 0; n < codecs_.size(); ++n) {
     if (n & 0x1) {  // Just add codecs with odd index.
       EXPECT_EQ(
-          0, receiver_->AddCodec(n, codecs_[n].pltype, codecs_[n].channels,
-                                 codecs_[n].plfreq, NULL, codecs_[n].plname));
+          0, receiver_->AddCodec(rtc::checked_cast<int>(n), codecs_[n].pltype,
+                                 codecs_[n].channels, codecs_[n].plfreq, NULL,
+                                 codecs_[n].plname));
     }
   }
   // Get codec and compare.
@@ -338,7 +339,7 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
         EXPECT_EQ(0,
                   receiver_->GetAudio(output_sample_rate_hz, &frame, &muted));
         EXPECT_EQ(expected_output_ts, frame.timestamp_);
-        expected_output_ts += 10 * samples_per_ms;
+        expected_output_ts += rtc::checked_cast<uint32_t>(10 * samples_per_ms);
         EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
         EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
         EXPECT_EQ(output_channels, frame.num_channels_);
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index f95dca256c..bd1e884612 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -37,6 +37,7 @@
 #include "rtc_base/md5digest.h"
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/refcountedobject.h"
+#include "rtc_base/safe_conversions.h"
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/clock.h"
 #include "system_wrappers/include/event_wrapper.h"
@@ -122,7 +123,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
 
   int last_payload_len_bytes() const {
     rtc::CritScope lock(&crit_sect_);
-    return last_payload_vec_.size();
+    return rtc::checked_cast<int>(last_payload_vec_.size());
   }
 
   FrameType last_frame_type() const {
@@ -1158,9 +1159,9 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
   bool RegisterExternalSendCodec(AudioEncoder* external_speech_encoder,
                                  int payload_type) {
     payload_type_ = payload_type;
-    frame_size_rtp_timestamps_ =
+    frame_size_rtp_timestamps_ = rtc::checked_cast<uint32_t>(
         external_speech_encoder->Num10MsFramesInNextPacket() *
-        external_speech_encoder->RtpTimestampRateHz() / 100;
+        external_speech_encoder->RtpTimestampRateHz() / 100);
     return send_test_->RegisterExternalCodec(external_speech_encoder);
   }
 
@@ -1589,7 +1590,7 @@ class AcmSetBitRateTest : public ::testing::Test {
     int nr_bytes = 0;
     while (std::unique_ptr<test::Packet> next_packet =
                send_test_->NextPacket()) {
-      nr_bytes += next_packet->payload_length_bytes();
+      nr_bytes += rtc::checked_cast<int>(next_packet->payload_length_bytes());
     }
     EXPECT_EQ(expected_total_bits, nr_bytes * 8);
   }
@@ -1742,9 +1743,11 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
       if (packet_counter == nr_packets / 2)
         send_test_->acm()->SetBitRate(target_bitrate_bps);
       if (packet_counter < nr_packets / 2)
-        nr_bytes_before += next_packet->payload_length_bytes();
+        nr_bytes_before += rtc::checked_cast<int>(
+            next_packet->payload_length_bytes());
       else
-        nr_bytes_after += next_packet->payload_length_bytes();
+        nr_bytes_after += rtc::checked_cast<int>(
+            next_packet->payload_length_bytes());
       packet_counter++;
     }
     EXPECT_EQ(expected_before_switch_bits, nr_bytes_before * 8);
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index b6060a99e6..09d10665f4 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
 
@@ -213,8 +214,8 @@ TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
   int overall_bitrate = 34567;
   size_t overhead_bytes_per_packet = 64;
   int frame_length_ms = 20;
-  int current_bitrate =
-      overall_bitrate - overhead_bytes_per_packet * 8 * 1000 / frame_length_ms;
+  int current_bitrate = rtc::checked_cast<int>(
+      overall_bitrate - overhead_bytes_per_packet * 8 * 1000 / frame_length_ms);
 
   UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
                        rtc::Optional<size_t>(overhead_bytes_per_packet));
@@ -231,8 +232,9 @@ TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
 
   // Next: change frame length.
   frame_length_ms = 60;
-  current_bitrate += overhead_bytes_per_packet * 8 * 1000 / 20 -
-                     overhead_bytes_per_packet * 8 * 1000 / 60;
+  current_bitrate += rtc::checked_cast<int>(
+      overhead_bytes_per_packet * 8 * 1000 / 20 -
+      overhead_bytes_per_packet * 8 * 1000 / 60);
   UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
                        rtc::Optional<size_t>(overhead_bytes_per_packet));
   CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
@@ -248,8 +250,9 @@ TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
 
   // Next: change frame length.
   frame_length_ms = 20;
-  current_bitrate -= overhead_bytes_per_packet * 8 * 1000 / 20 -
-                     overhead_bytes_per_packet * 8 * 1000 / 60;
+  current_bitrate -= rtc::checked_cast<int>(
+      overhead_bytes_per_packet * 8 * 1000 / 20 -
+      overhead_bytes_per_packet * 8 * 1000 / 60);
   UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
                        rtc::Optional<size_t>(overhead_bytes_per_packet));
   CheckDecision(&controller, rtc::Optional<int>(frame_length_ms),
@@ -259,8 +262,9 @@ TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
   overall_bitrate -= 100;
   current_bitrate -= 100;
   frame_length_ms = 60;
-  current_bitrate += overhead_bytes_per_packet * 8 * 1000 / 20 -
-                     overhead_bytes_per_packet * 8 * 1000 / 60;
+  current_bitrate += rtc::checked_cast<int>(
+      overhead_bytes_per_packet * 8 * 1000 / 20 -
+      overhead_bytes_per_packet * 8 * 1000 / 60);
 
   UpdateNetworkMetrics(&controller, rtc::Optional<int>(overall_bitrate),
                        rtc::Optional<size_t>(overhead_bytes_per_packet));
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
index 3955e4ab65..ec79c28efe 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
+++ b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -13,6 +13,7 @@
 #include <vector>
 
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -58,8 +59,8 @@ TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
     auto encoder = factory->MakeAudioEncoder(kTestPayloadType, spec.format);
     EXPECT_TRUE(encoder);
     encoder->Reset();
-    const int num_samples =
-        encoder->SampleRateHz() * encoder->NumChannels() / 100;
+    const int num_samples = rtc::checked_cast<int>(
+        encoder->SampleRateHz() * encoder->NumChannels() / 100);
     rtc::Buffer out;
     rtc::BufferT<int16_t> audio;
     audio.SetData(num_samples, [](rtc::ArrayView<int16_t> audio) {
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index ef3ff31337..577962548d 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -14,6 +14,7 @@
 #include "common_audio/vad/mock/mock_vad.h"
 #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "rtc_base/constructormagic.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/gtest.h"
 #include "test/mock_audio_encoder.h"
 
@@ -290,7 +291,8 @@ TEST_F(AudioEncoderCngTest, EncodePassive) {
                   encoded_info_.encoded_bytes);
         EXPECT_EQ(expected_timestamp, encoded_info_.encoded_timestamp);
       }
-      expected_timestamp += kBlocksPerFrame * num_audio_samples_10ms_;
+      expected_timestamp += rtc::checked_cast<uint32_t>(
+          kBlocksPerFrame * num_audio_samples_10ms_);
     } else {
       // Otherwise, expect no output.
       EXPECT_EQ(0u, encoded_info_.encoded_bytes);
diff --git a/modules/audio_coding/codecs/isac/unittest.cc b/modules/audio_coding/codecs/isac/unittest.cc
index 7a811cf4b0..df78ab797c 100644
--- a/modules/audio_coding/codecs/isac/unittest.cc
+++ b/modules/audio_coding/codecs/isac/unittest.cc
@@ -17,6 +17,7 @@
 #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "rtc_base/buffer.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
@@ -163,10 +164,12 @@ void TestGetSetBandwidthInfo(const int16_t* speech_data,
     const int send_time = elapsed_time_ms * (sample_rate_hz / 1000);
     EXPECT_EQ(0, T::UpdateBwEstimate(
                      encdec, bitstream1.data(), bitstream1.size(), i, send_time,
-                     channel1.Send(send_time, bitstream1.size())));
+                     channel1.Send(send_time,
+                                   rtc::checked_cast<int>(bitstream1.size()))));
     EXPECT_EQ(0, T::UpdateBwEstimate(
                      dec, bitstream2.data(), bitstream2.size(), i, send_time,
-                     channel2.Send(send_time, bitstream2.size())));
+                     channel2.Send(send_time,
+                                   rtc::checked_cast<int>(bitstream2.size()))));
 
     // 3. Decode, and get new BW info from the separate decoder.
     ASSERT_EQ(0, T::SetDecSampRate(encdec, sample_rate_hz));
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
index 9fd60441a9..06182ee123 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "modules/audio_coding/acm2/rent_a_codec.h"
 #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -140,8 +141,9 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
         ASSERT_EQ(value, payload[i]);
       }
 
-      expected_timestamp += expected_split.frame_sizes[i] * samples_per_ms_;
-      expected_byte_offset += length_bytes;
+      expected_timestamp += rtc::checked_cast<uint32_t>(
+          expected_split.frame_sizes[i] * samples_per_ms_);
+      expected_byte_offset += rtc::checked_cast<uint32_t>(length_bytes);
     }
   }
 }
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index 9f2c1bbe72..c9e6ad150b 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -15,6 +15,7 @@
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "modules/audio_coding/neteq/tools/audio_loop.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
@@ -334,7 +335,7 @@ void OpusTest::TestCbrEffect(bool cbr, int block_length_ms) {
       int32_t diff = std::abs((int32_t)encoded_bytes_ - prev_pkt_size);
       max_pkt_size_diff = std::max(max_pkt_size_diff, diff);
     }
-    prev_pkt_size = encoded_bytes_;
+    prev_pkt_size = rtc::checked_cast<int32_t>(encoded_bytes_);
   }
 
   if (cbr) {
@@ -736,7 +737,9 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
         WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
                           rtc::CheckedDivExact(speech_block.size(), channels_),
                           kMaxBytes, bitstream_);
-    if (opus_repacketizer_cat(rp, bitstream_, encoded_bytes_) == OPUS_OK) {
+    if (opus_repacketizer_cat(
+            rp, bitstream_,
+            rtc::checked_cast<opus_int32>(encoded_bytes_)) == OPUS_OK) {
       ++num_packets;
       if (num_packets == kPackets) {
         break;
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 06ae2a70a5..8409b82fea 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -13,6 +13,7 @@
 
 #include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/gtest.h"
 #include "test/mock_audio_encoder.h"
 
@@ -59,7 +60,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
         timestamp_,
         rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
         &encoded_);
-    timestamp_ += num_audio_samples_10ms;
+    timestamp_ += rtc::checked_cast<uint32_t>(num_audio_samples_10ms);
   }
 
   MockAudioEncoder* mock_encoder_;
diff --git a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
index 051b8ee3d2..1489e80308 100644
--- a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
+++ b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/safe_conversions.h"
 
 #include <assert.h>
 #include <stdlib.h>
@@ -51,7 +52,7 @@ class AudioMultiVectorTest : public ::testing::TestWithParam<size_t> {
     // And so on.
     for (size_t i = 0; i < array_length(); ++i) {
       for (size_t j = 1; j <= num_channels_; ++j) {
-        *ptr = j * 100 + i;
+        *ptr = rtc::checked_cast<int16_t>(j * 100 + i);
         ++ptr;
       }
     }
diff --git a/modules/audio_coding/neteq/audio_vector_unittest.cc b/modules/audio_coding/neteq/audio_vector_unittest.cc
index e027ee8d13..1ff8b85f5d 100644
--- a/modules/audio_coding/neteq/audio_vector_unittest.cc
+++ b/modules/audio_coding/neteq/audio_vector_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "modules/audio_coding/neteq/audio_vector.h"
+#include "rtc_base/safe_conversions.h"
 
 #include <assert.h>
 #include <stdlib.h>
@@ -25,7 +26,7 @@ class AudioVectorTest : public ::testing::Test {
   virtual void SetUp() {
     // Populate test array.
     for (size_t i = 0; i < array_length(); ++i) {
-      array_[i] = i;
+      array_[i] = rtc::checked_cast<int16_t>(i);
     }
   }
 
@@ -253,7 +254,7 @@ TEST_F(AudioVectorTest, InsertAtEnd) {
   for (int i = 0; i < kNewLength; ++i) {
     new_array[i] = 100 + i;
   }
-  int insert_position = array_length();
+  int insert_position = rtc::checked_cast<int>(array_length());
   vec.InsertAt(new_array, kNewLength, insert_position);
   // Verify that the vector looks as follows:
   // {0, 1, ..., kLength - 1, 100, 101, ..., 100 + kNewLength - 1 }.
@@ -282,7 +283,8 @@ TEST_F(AudioVectorTest, InsertBeyondEnd) {
   for (int i = 0; i < kNewLength; ++i) {
     new_array[i] = 100 + i;
   }
-  int insert_position = array_length() + 10;  // Too large.
+  int insert_position = rtc::checked_cast<int>(
+      array_length() + 10); // Too large.
   vec.InsertAt(new_array, kNewLength, insert_position);
   // Verify that the vector looks as follows:
   // {0, 1, ..., kLength - 1, 100, 101, ..., 100 + kNewLength - 1 }.
@@ -338,7 +340,7 @@ TEST_F(AudioVectorTest, OverwriteBeyondEnd) {
   for (int i = 0; i < kNewLength; ++i) {
     new_array[i] = 100 + i;
   }
-  int insert_position = array_length() - 2;
+  int insert_position = rtc::checked_cast<int>(array_length() - 2);
   vec.OverwriteAt(new_array, kNewLength, insert_position);
   ASSERT_EQ(array_length() - 2u + kNewLength, vec.Size());
   // Verify that the vector looks as follows:
diff --git a/modules/audio_coding/neteq/expand_unittest.cc b/modules/audio_coding/neteq/expand_unittest.cc
index b52d626b8d..aeaa07bd01 100644
--- a/modules/audio_coding/neteq/expand_unittest.cc
+++ b/modules/audio_coding/neteq/expand_unittest.cc
@@ -88,8 +88,8 @@ class ExpandTest : public ::testing::Test {
   void SetUp() override {
     // Fast-forward the input file until there is speech (about 1.1 second into
     // the file).
-    const size_t speech_start_samples =
-        static_cast<size_t>(test_sample_rate_hz_ * 1.1f);
+    const int speech_start_samples =
+        static_cast<int>(test_sample_rate_hz_ * 1.1f);
     ASSERT_TRUE(input_file_.Seek(speech_start_samples));
 
     // Pre-load the sync buffer with speech data.
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index a6e48c7593..45d7f26c52 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -483,7 +483,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
         decoded[i] = next_value_++;
       }
       *speech_type = kSpeech;
-      return encoded_len;
+      return rtc::checked_cast<int>(encoded_len);
     }
 
     void Reset() override { next_value_ = 1; }
@@ -1312,7 +1312,7 @@ class Decoder120ms : public AudioDecoder {
       decoded[i] = next_value_++;
     }
     *speech_type = speech_type_;
-    return decoded_len;
+    return rtc::checked_cast<int>(decoded_len);
   }
 
   void Reset() override { next_value_ = 1; }
diff --git a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
index 7d97210595..153a18e30e 100644
--- a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
@@ -20,6 +20,7 @@
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
 #include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/safe_conversions.h"
 #include "test/gtest.h"
 #include "test/mock_audio_decoder_factory.h"
 
@@ -99,7 +100,8 @@ Packet CreateRedPayload(size_t num_payloads,
     // Not the last block; set F = 1.
     *payload_ptr |= 0x80;
     ++payload_ptr;
-    int this_offset = (num_payloads - i - 1) * timestamp_offset;
+    int this_offset = rtc::checked_cast<int>(
+        (num_payloads - i - 1) * timestamp_offset);
     *payload_ptr = this_offset >> 6;
     ++payload_ptr;
     assert(kPayloadLength <= 1023);  // Max length described by 10 bits.
diff --git a/modules/audio_coding/neteq/sync_buffer_unittest.cc b/modules/audio_coding/neteq/sync_buffer_unittest.cc
index f3f789527c..ad049420e4 100644
--- a/modules/audio_coding/neteq/sync_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/sync_buffer_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/safe_conversions.h"
 
 #include "test/gtest.h"
 
@@ -57,7 +58,7 @@ TEST(SyncBuffer, PushBackAndFlush) {
   // Populate |new_data|.
   for (size_t channel = 0; channel < kChannels; ++channel) {
     for (size_t i = 0; i < kNewLen; ++i) {
-      new_data[channel][i] = i;
+      new_data[channel][i] = rtc::checked_cast<int16_t>(i);
     }
   }
   // Push back |new_data| into |sync_buffer|. This operation should pop out
@@ -97,7 +98,7 @@ TEST(SyncBuffer, PushFrontZeros) {
   // Populate |new_data|.
   for (size_t channel = 0; channel < kChannels; ++channel) {
     for (size_t i = 0; i < kNewLen; ++i) {
-      new_data[channel][i] = 1000 + i;
+      new_data[channel][i] = rtc::checked_cast<int16_t>(1000 + i);
     }
   }
   sync_buffer.PushBack(new_data);
@@ -130,7 +131,7 @@ TEST(SyncBuffer, GetNextAudioInterleaved) {
   // Populate |new_data|.
   for (size_t channel = 0; channel < kChannels; ++channel) {
     for (size_t i = 0; i < kNewLen; ++i) {
-      new_data[channel][i] = i;
+      new_data[channel][i] = rtc::checked_cast<int16_t>(i);
     }
   }
   // Push back |new_data| into |sync_buffer|. This operation should pop out
diff --git a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
index e0ee265aae..32bccea9ea 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
+++ b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
@@ -11,6 +11,7 @@
 // Unit tests for test InputAudioFile class.
 
 #include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/safe_conversions.h"
 
 #include "test/gtest.h"
 
@@ -22,7 +23,7 @@ TEST(TestInputAudioFile, DuplicateInterleaveSeparateSrcDst) {
   static const size_t kChannels = 2;
   int16_t input[kSamples];
   for (size_t i = 0; i < kSamples; ++i) {
-    input[i] = i;
+    input[i] = rtc::checked_cast<int16_t>(i);
   }
   int16_t output[kSamples * kChannels];
   InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, output);
@@ -41,7 +42,7 @@ TEST(TestInputAudioFile, DuplicateInterleaveSameSrcDst) {
   static const size_t kChannels = 5;
   int16_t input[kSamples * kChannels];
   for (size_t i = 0; i < kSamples; ++i) {
-    input[i] = i;
+    input[i] = rtc::checked_cast<int16_t>(i);
   }
   InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, input);