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); |