Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index f57db27a3177f494813af1465657fcc6a7dfd074..a496316b42149637757846ec3e20f89b32915f02 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -1178,8 +1178,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
RTC_CHECK(stream_); |
} |
- bool SendTelephoneEvent(int payload_type, uint8_t event, |
- uint32_t duration_ms) { |
+ bool SendTelephoneEvent(int payload_type, int event, int duration_ms) { |
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
RTC_DCHECK(stream_); |
return stream_->SendTelephoneEvent(payload_type, event, duration_ms); |