| Index: webrtc/media/engine/webrtcvoiceengine.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
|
| index 3eb4c46700e51e4f89009d1aacc3f192896b6ad2..ba1bc7ff028a108682fd23e33776d3bd9530c8dd 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.cc
|
| @@ -1972,7 +1972,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
| // parameters.
|
| // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
|
| webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
|
| - {
|
| + do {
|
| send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
|
|
|
| // Find send codec (the first non-telephone-event/CN codec).
|
| @@ -1980,7 +1980,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
| codecs, &send_codec_spec.codec_inst);
|
| if (!codec) {
|
| LOG(LS_WARNING) << "Received empty list of codecs.";
|
| - return false;
|
| + break;
|
| }
|
|
|
| send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
|
| @@ -2050,7 +2050,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
|
| break;
|
| }
|
| }
|
| - }
|
| + } while (0);
|
|
|
| if (send_codec_spec_ != send_codec_spec) {
|
| send_codec_spec_ = std::move(send_codec_spec);
|
|
|