| Index: webrtc/media/engine/webrtcvoiceengine.h
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
|
| index 03b493744b857f903cbb09f530794787466e7b8f..76463739e81e78b7730b579678419df085d509b9 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine.h
|
| +++ b/webrtc/media/engine/webrtcvoiceengine.h
|
| @@ -259,6 +259,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
|
| bool playout_ = false;
|
| bool send_ = false;
|
| webrtc::Call* const call_ = nullptr;
|
| + webrtc::Call::Config::BitrateConfig bitrate_config_;
|
|
|
| // SSRC of unsignalled receive stream, or -1 if there isn't one.
|
| int64_t default_recv_ssrc_ = -1;
|
| @@ -280,6 +281,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
|
| std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
|
|
|
| webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
|
| + bool send_codec_spec_set_ = false;
|
|
|
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
|
| };
|
|
|