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 1b6df005d6c9a2cb2ab6eb1ae7055261679dbb89..6dcff3ab37eafda66c444a37a84be40f29fdbab3 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -1153,7 +1153,7 @@ int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) { |
if (!audio_configured_) { |
return -1; |
} |
- return audio_->SetAudioPacketSize(packet_size_samples); |
+ return 0; |
} |
int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) { |