| 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);
|
|
|