| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| index dc7b20e9db31fb673471d012d9e6610dcca16d29..024e028393057c54e2772945c8d49f0f17cb1aea 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -1132,13 +1132,6 @@ int32_t RTPSender::SendTelephoneEvent(uint8_t key,
|
| return audio_->SendTelephoneEvent(key, time_ms, level);
|
| }
|
|
|
| -int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) {
|
| - if (!audio_configured_) {
|
| - return -1;
|
| - }
|
| - return 0;
|
| -}
|
| -
|
| int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
|
| return audio_->SetAudioLevel(level_d_bov);
|
| }
|
|
|