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}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 3955e4a..ec79c28 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 @@
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 ef3ff31..5779625 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 @@
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 7a811cf..df78ab7 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 @@
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 9fd6044..06182ee 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 @@
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 9f2c1bb..c9e6ad1 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 @@
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 @@
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 06ae2a7..8409b82 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 @@
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_;