| Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| index abe4b5313ddf77853fbe13e86e6f903c996a9105..612d5c2e6b0fb27eef815b91f8c7272c767877d2 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| @@ -14,6 +14,7 @@
|
| #include "webrtc/base/arraysize.h"
|
| #include "webrtc/base/byteorder.h"
|
| #include "webrtc/base/gunit.h"
|
| +#include "webrtc/base/safe_conversions.h"
|
| #include "webrtc/call/call.h"
|
| #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
|
| #include "webrtc/media/base/fakemediaengine.h"
|
| @@ -3724,7 +3725,7 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
|
| }
|
|
|
| // Find the index of a codec, or -1 if not found, so that we can easily check
|
| - // supplementary codecs are orderd after the general codecs.
|
| + // supplementary codecs are ordered after the general codecs.
|
| auto find_codec =
|
| [&codecs](const webrtc::SdpAudioFormat& format) -> int {
|
| for (size_t i = 0; i != codecs.size(); ++i) {
|
| @@ -3732,7 +3733,7 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
|
| if (STR_CASE_CMP(codec.name.c_str(), format.name.c_str()) == 0 &&
|
| codec.clockrate == format.clockrate_hz &&
|
| codec.channels == format.num_channels) {
|
| - return static_cast<int>(i);
|
| + return rtc::checked_cast<int>(i);
|
| }
|
| }
|
| return -1;
|
|
|