I've successfully implemented the OpenTok. In group discussion (Three people, including moderator) video and audio transmission works well, but problem starts when moderator record stream. Video clip downloaded from OpenTok server has no sound.
Does anyone have any idea what can be wrong?
Thanks to Ankur (OpenTok Forum):
The audio stream in the downloaded videos is in the SPEEX codec. Many
desktop audio players don't recognize the codec. You may need to use
ffmpeg to transcode the audio before it is playable in programs like
VLC.
Related
I have a content creation site I am building and im confused on audio and video.
If I have a content creators audio or video stored in s3 and then I want to display their file will the html video player or audio player stream the media or will it download it fully then play it?
I ask because what if the video or audio is significantly long. like 2 hours for example. I need to know how to solve the use case.
Lastly what file type is most acceptable for viewing on webpages? It seems like MPEG-4 is the best bet. Is that true?
Most video player clients and browsers will attempt to stream the video if they can.
For an mp4 video file hosted on a server, so long as the header is at the start and the server accepts range requests, this will mean the player downloads the video in chunks and starts playing as soon as it has enough to decide the first frames.
For more professional streaming services, they will generally use an adaptive bit rate streaming protocol like DASH or HLS (see this answer: https://stackoverflow.com/a/42365034/334402) and again the video will be streamed in chunks, or segments, and will start playing while it is streaming.
To answer your last question you need to be aware that the raw video is encoded (e.g. h.264, VP9 etc) and the video, audio, subtitle etc tracks stored in a video container (e.g. mp4, Web etc).
The most common format is probaly h.264 encoded and mp4 containers at this time.
The particular profile for h.264 can matter also depending on the device - baseline is probably the most supported profile at this time. You can find examples of media support for different devices online, e.g. for Android: https://developer.android.com/guide/topics/media/media-formats
#Mick's answer is spot on. I'll just add that mp4 (with h264 encoding) will work in just about every browser out there.
The issue with mp4 files (especially with a 2 hour long movie) isn't so much the seeking & streaming. If your creator creates a 4K video - thats what you'll deliver to everyone (even mobile phones). HLS streaming on the other hand has adaptive bitrates - where the video adapts to both the screen & the available network speeds. You'll get better playback results with less buffering (and if you're using AWS - a LOT LESS data egress) with video streaming.
(there are a bunch of APIs and services that can help you do this - including api.video (where I work), Mux and others).
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?
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.
I have every stream type enabled on my wowza server.
a week ago i had it working in the wowza test players
now the only one that works is the RTSP in VLC
now every stream just shows a black screen.
If i try to access the m3u8 via Safari browser i can hear the audio but no visual
Any assistance on this would be a major help.
this is due to incompatible video, audio codec. you need to user right encoder or you can use wowza transcoder to transcode your stream (h.264 , AAC) . it will resolve your playback problem. if you are using flash media live encoder you can select h.264 encoding method, and audio codec should be aac or mp3.
Some quick troubleshooting steps:
What source encoder are you using? Can you playback the source URL on VLC as well? What codec info is displayed in VLC for the source stream? It should be one of the supported codecs for Wowza.
What codec info is displayed in VLC for the RTSP playback link generated by your Wowza server?
Do you see any errors/warnings in the access log when you publish and playback your HLS (m3u8) stream?
Usually, this kind of error is due to an incorrect video codec, or an encoding setting, or network saturation where the video packets are not coming through correctly. You can check what error or warning messages are being generated by tailing the access log found in logs/ folder of your Wowza installation directory.
Hope this helps.
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.