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Unified Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 1722253002: - Clean up unused voice engine DTMF code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_1
Patch Set: more remove Created 4 years, 10 months ago
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Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 6788699cd5003037536ffecc2872583ec908727c..fad1da898b4e68db45d6be4a3e220fc871ada7bc 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -45,9 +45,9 @@ const CallStatistics kCallStats = {
1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123};
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 kTelephoneEventPayloadType = 123;
const uint8_t kTelephoneEventCode = 45;
-const uint32_t kTelephoneEventDuration = 6789;
+const uint16_t kTelephoneEventDuration = 6789;
struct ConfigHelper {
ConfigHelper()

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