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Fixing warning C4267 on Win (more_configs).

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We added a new bot to client.webrtc.fyi (https://build.chromium.org/p/client.webrtc.fyi/builders/Win%20%28more%20configs%29).

It seems it is spotting some unsafe conversions and this CL is a test to see if we can use rtc::dchecked_cast to fix them:
../../modules/audio_coding/neteq/neteq_unittest.cc(547): error C2220: warning treated as error - no 'object' file generated
../../modules/audio_coding/neteq/neteq_unittest.cc(547): warning C4267: '=': conversion from 'size_t' to 'uint16_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(548): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(977): warning C4267: '+=': conversion from 'size_t' to 'uint32_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(979): warning C4267: '+=': conversion from 'size_t' to 'uint32_t', possible loss 

Bug: chromium:759980
Change-Id: Icd0f32ccf620c7c6642fadff797dc2482918648d
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/12921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20335}
This commit is contained in:
Mirko Bonadei
2017-10-18 14:22:50 +02:00
committed by Commit Bot
parent 9ce0c7a2d2
commit a811027990
1 changed files with 7 additions and 4 deletions
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11
modules/audio_coding/neteq/neteq_unittest.cc
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@ -29,6 +29,7 @@
#include "rtc_base/flags.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/protobuf_utils.h"
#include "rtc_base/safe_conversions.h"
#include "rtc_base/sha1digest.h"
#include "rtc_base/stringencode.h"
#include "test/gtest.h"
@ -544,8 +545,8 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
for (size_t i = 0; i < num_frames; ++i) {
const uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
rtp_info.sequenceNumber = i;
rtp_info.timestamp = i * kSamples;
rtp_info.sequenceNumber = rtc::checked_cast<uint16_t>(i);
rtp_info.timestamp = rtc::checked_cast<uint32_t>(i * kSamples);
rtp_info.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
@ -974,9 +975,11 @@ class NetEqBgnTest : public NetEqDecodingTest {
ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Next packet.
rtp_info.timestamp += expected_samples_per_channel;
rtp_info.timestamp += rtc::checked_cast<uint32_t>(
expected_samples_per_channel);
rtp_info.sequenceNumber++;
receive_timestamp += expected_samples_per_channel;
receive_timestamp += rtc::checked_cast<uint32_t>(
expected_samples_per_channel);
}
output.Reset();