iridium/neko-custom
Archived
2
0
Fork 0
Code Pull Requests Activity

opus useinbandfec.

Browse Source
This commit is contained in:
Miroslav Šedivý 2022-03-24 21:48:03 +01:00
parent 130832177c
commit 2189e4fd49
3 changed files with 3 additions and 2 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
docs/changelog.md
View File

@ -19,6 +19,7 @@
- Broadcast status change is sent to all admins now. - Broadcast status change is sent to all admins now.
- NordVPN replaced with Sponsorblock extension in default configuration #144. - NordVPN replaced with Sponsorblock extension in default configuration #144.
- Removed `vncviewer` image, as its functionality is replaced and extended by remmina. - Removed `vncviewer` image, as its functionality is replaced and extended by remmina.
- Opus uses `useinbandfec=1` from now on, hopefully fixes minor audio loss issues.
## [n.eko v2.5](https://github.com/m1k1o/neko/releases/tag/v2.5) ## [n.eko v2.5](https://github.com/m1k1o/neko/releases/tag/v2.5)

2
server/internal/gst/gst.go
View File

@ -160,7 +160,7 @@ func CreateAppPipeline(codecName string, pipelineDevice string, pipelineSrc stri
return nil, err return nil, err
} }
pipelineStr = fmt.Sprintf(audioSrc+"opusenc bitrate=%d"+pipelineStr, pipelineDevice, bitrate*1000) pipelineStr = fmt.Sprintf(audioSrc+"opusenc inband-fec=true bitrate=%d"+pipelineStr, pipelineDevice, bitrate*1000)
case "G722": case "G722":
// https://gstreamer.freedesktop.org/documentation/libav/avenc_g722.html?gi-language=c // https://gstreamer.freedesktop.org/documentation/libav/avenc_g722.html?gi-language=c
// gstreamer1.0-libav // gstreamer1.0-libav

2
server/internal/webrtc/webrtc.go
View File

@ -322,7 +322,7 @@ func (manager *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalS
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264, ClockRate: 90000, Channels: 0, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", RTCPFeedback: fb}, PayloadType: 102} codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264, ClockRate: 90000, Channels: 0, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f", RTCPFeedback: fb}, PayloadType: 102}
id = "video" id = "video"
case "Opus": case "Opus":
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 111} codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 2, SDPFmtpLine: "useinbandfec=1", RTCPFeedback: fb}, PayloadType: 111}
id = "audio" id = "audio"
case "G722": case "G722":
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeG722, ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 9} codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeG722, ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 9}