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Fixing warning C4267 on Win (more_configs).

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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}
This commit is contained in:
Mirko Bonadei
2017-10-19 09:00:17 +02:00
committed by Commit Bot
parent 117c48291c
commit 737e073f8d
16 changed files with 70 additions and 43 deletions
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5
modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
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@ -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) {

4
modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
View File

@ -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);

7
modules/audio_coding/codecs/isac/unittest.cc
View File

@ -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));

6
modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
View File

@ -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);
}
}
}

7
modules/audio_coding/codecs/opus/opus_unittest.cc
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@ -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;

3
modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
View File

@ -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_;