| Index: webrtc/audio/audio_send_stream.cc
|
| diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
|
| index ad6366bc8095deebe830b33c9115cf77afe83adf..a6617d00abf030b08ee46a55f6eb185dd9fc7f65 100644
|
| --- a/webrtc/audio/audio_send_stream.cc
|
| +++ b/webrtc/audio/audio_send_stream.cc
|
| @@ -135,10 +135,12 @@ void AudioSendStream::Stop() {
|
| }
|
| }
|
|
|
| -bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
|
| +bool AudioSendStream::SendTelephoneEvent(int payload_type,
|
| + int payload_frequency, int event,
|
| int duration_ms) {
|
| RTC_DCHECK(thread_checker_.CalledOnValidThread());
|
| - return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
|
| + return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type,
|
| + payload_frequency) &&
|
| channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
|
| }
|
|
|
|
|