Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index 160a818323b139381c06ca16164b3d8b667e8491..24afcbcf58e94ad2582dd9fc88cf7860f27d3b2d 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -125,8 +125,8 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
return false; |
} |
-bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event, |
- uint32_t duration_ms) { |
+bool AudioSendStream::SendTelephoneEvent(int payload_type, int event, |
+ int duration_ms) { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) && |
channel_proxy_->SendTelephoneEventOutband(event, duration_ms); |