Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1782053002
Cr-Original-Commit-Position: refs/heads/master@{#11953}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 8842c3e41bcc4a2968d7c299f84f87099485a8e8
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 6788699..c04a3de 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -46,8 +46,8 @@
const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
const int kTelephoneEventPayloadType = 123;
-const uint8_t kTelephoneEventCode = 45;
-const uint32_t kTelephoneEventDuration = 6789;
+const int kTelephoneEventCode = 45;
+const int kTelephoneEventDuration = 6789;
struct ConfigHelper {
ConfigHelper()
