We recently built a demo application utilizing Kurento Media Server to record applicant video interview, but the audio quality is not well , some audio is not recognizable and some of it had high pitch noise. We've been test it on several models of PC or Mac, so this should not be device problem.
We've been using RecorderEndpoint with media profile MediaProfileSpecType.WEBM ,and all other setting remain as default.
To fix this problem, we tried:
We upgrade to Kurento 6.2.1 which use Opus as the audio encoder.
Try to using setMaxOuputBitrate of the recorder, we don't see it has improvements or I don't know which bit rate range can be used.
Change SDPOffer to setup a high bit rate audio for Opus which we don't know where to modify
None of it is working so far, so please tell us where to look.
Thanks.
Please check with this recording tutorial. The audio should be fine. Just make sure you are only sending audio, and not video. That should help.
If the audio is not being recorded correctly, I would try and hear what's coming out of your box through your browser. Try and run the hello-world tutorial, with a pair of headphones connected to your box so you don't have echoes.
About #2, if you want to raise the bitrate exchanged between the webrtc endpoint and the recorder, you need to invoke the setOutputBitrate command on the webrtc endpoint.
Related
I am trying to build an open-source in-ear monitoring system. I have created the UI and was wondering how I would get the channels that are on an audio mixing console so that I can edit the channels and stream them to each musician. Is there a certain protocol that all the mixers use? You can find the project at https://gitlab.com/openstagemix. We would love to have contributors.
I can't really test whether this is the correct answer as I am trapped in my house during the coronavirus time. But, all mixers use something called OSC which is a protocol between mixers, synthesizers, etc. to computers. You can find more information here http://opensoundcontrol.org/introduction-osc.
Update:
It's neither! I am going to use the AES67 standard to receive information from my mixer and with that process the audio. This is because my mixer is ethernet capable.
This is the details of the video
when i play my gameplay in vlc media player it plays fine and the audio syncs with the video but when i import the video into premiere pro the audio gets out of sync. i don't know why is this happening. i researched about it and tried converting my video into CFR using Handbrake software but that didn't helped.
I'm thinking that maybe you don't have a setting selecting one your timeline that keeps the audio and video tracks synced. I personally like my timeline settings like this
Let me know if that works, if not maybe try adding a screen recording of the issue or some screenshots?
Check it out here:
https://webrtc.github.io/samples/src/content/devices/input-output/
All the audio recording sounds like aliens trying to communicate with you. I could of swore WebRTC was working just last week but this site of samples is completely freaking out on both my laptop and my phone.
Does anyone know why the audio is freaking out like that?
Is it possible to see the live stream of an IP camera using RTSP ?
Example URL: rtsp://public ip:554/1363e66e.mp4
The encoding is mp4 h.264 baseline profile at 320 x 240 resolution.
I followed the Wiki link here.
But I get the error: Prefetch error -2
When I try to play using real player on the nokia e72, I get the error: 'General: System Error'.
Please let me know what I can do about this.
There are no video players on Ovi store that can play the stream either but I am able to play the stream on VLC on the desktop.
You can stream it using ReaPlayer if you don't have VLC player in Ovi store. See the port address range supported by your IP camera. Try the range of 1024 - 2000. RTSP supports VLC, Quicktime and Real player. Using any of these objects you can stream it.
So I think here is the case,
There are a few different mp4 containers. Standard one wont allow you to wrap a real time data into a mp4 container because mp4 needs to have a field/atom in its header called
MDAT and it has information about the file and its size as well.(which is written after the file is completely encoded. )
So unless you wake that you can not stream live stuff in mp4 UNLESS it is fragmented mp4.
Media Foundation will allow you to do this when windows 8 is out( i got the intel from the msdn forum so I dont know how true it is).
I dont know what ffmpeg/Gstreamer is capable of. Again if this is a commercial product you are working on you might run into some licensing issues with ffmpeg.
Look at webrtc.
I am guessing best bet it to use webm or ogg/theora but I am not sure if theora can do what you want, This is something I am also working on.
Please share your findings
Thanks.
I was wondering if anyone knows of a way of recording audio played on a website, for example a client would play a series of sound samples on my site and he/she was able to record it and play it back (still on my website). I found tons of questions/answers of this type but in regards to recording sounds from client's mic not sounds played by a sound card.
SoundCloud has something like that:
http://eu.techcrunch.com/2010/12/01/soundcloud-launches-super-cool-ability-to-record-on-site-and-iphone-app/
In addition I am aware of red5--but I'm not exactly sure how or if it allows recording audio from a sound card. There are also other flv recorders but they all seem to do one thing: record sounds from a microphone through flash.
Thank you very much for your time.