commit a811027990e7eb674f1c89e09c524530947c26d2
author Mirko Bonadei <mbonadei@webrtc.org> Wed Oct 18 14:22:50 2017 +0200
committer Commit Bot <commit-bot@chromium.org> Wed Oct 18 12:28:58 2017 +0000
tree c6d1e913e71556c57f501409256f6c29e12dc701
parent 9ce0c7a2d2b4e1f6f85b44dd6face675309e4f07

Fixing warning C4267 on Win (more_configs).

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}

modules/audio_coding/neteq/neteq_unittest.cc

diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 8b3f28f..592323e 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -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 @@
   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 @@
       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();