Not sure if this is something obvious or not. After creating an YouTube LiveBroadcast, binding that to a LiveStream with a specific CDN format (let's say "720p"), and transitioning the broadcast from "ready" to "live" ... how can I change the stream quality without having to create a new broadcast?
Trying to unbind the current stream - exception is returned, cannot unbind the stream.
Trying to bind broadcast to another stream - same exception as above.
In addition, after looking through the support pages for YouTube live streaming, it is suggested that "ingest settings cannot be modified after the broadcast has started" - it says nothing about the actual API not being able to support this, but it looks like a major limitation from somewhere deeper. I only thought it applies to the web Live Control room.
I need this functionality so that I can change the stream quality for when a user switches from WiFi to mobile data. Currently streaming RTMP data in another resolution that what the LiveStream CDN format is configured for, results in health errors and encoding artifacts on YouTube's side. As suggested by the support pages, creating a "1080p" live stream ("maximum expected resolution") should work, but when that stream is receiving a 720p or 480p stream, depending on whether it was started or not, it either doesn't start at all, or goes to a gray scene with high-pitch audio (my stream is sent correctly, since I can output it to a dozen more outputs, like MP4, FLV, and other RTMP servers).
Solution?
Related
I am a newbie in video streaming and I just build a sample website which plays videos. Here i just give the video file location to the video tag in html5. I just noticed that in youtube the video tag contains the blob url and had a look into this. I found that the video data comes in segments and came across a term called pseudo streaming. Whereas it seems likes the website that i build downloads the whole file and plays the video. I am not trying to do any live streaming, just trying to stream local videos. I thought maybe the way video data is received in segments is done by a video streaming server. I came across RED5 open source streaming server, but most of the examples that is given is for live streaming which I am not experimenting on. Its been few days and I am not sure whether i am on the right track
The segmented approach you refer to is to support Adaptive Bit Rate streaming - ABR.
ABR allows the client device or player download the video in chunks, e.g 10 second chunks, and select the next chunk from the bit rate most appropriate to the current network conditions. See here for an example:
https://stackoverflow.com/a/42365034/334402
For your existing site, so long as your server supports range requests then you probably are not actually downloading the whole video. With Range Requests, the browser or player will request just part of the file at a time so it can start playback before the whole file is downloaded.
For MP4 files, it is worth noting that you need to have the header information, which is contained in a 'block' or 'atom' called MOOV atom, at the start of the file rather than the end - it is at the end for regular MP4 files. There are a number of tools which will allow you move it to the start - e.g.:
http://multimedia.cx/eggs/improving-qt-faststart/
You are definitely on the right track with your investigations - video hosting and streaming is a specialist area so it is generally easier to leverage existing streaming technologies and services rather than to build them your self. Some good places to look to get a feel for open source solutions:
https://gstreamer.freedesktop.org
http://www.videolan.org/vlc/streaming.html
I'm trying to get a livestream working on youtube. I want to stream 360° content with H264 video and AAC audio. The stream is started with the youtube live api from my mobile app and librtmp is used to deliver video and audio packets. I easily get to the point where the livestream health is good and my broadcast and stream are bound successfully.
However, when I try to transition to "testing" like this:
YoutubeManager.this.youtube.liveBroadcasts().transition("testing", liveBroadcast.getId(), "status").execute();
I get stuck on the "startTesting" status every time (100% reproducible) while I expect it to change to testing after few seconds to allow me to change it to live.
I don't know what's going on as in the youtube live control room everything seems to be fine but the encoder won't start.
Is it a common issue? Is there a mean to access the encoder logs? If you need more information feel free to ask me.
Regards.
I found a temporary fix !
I noticed 2 things :
When the autostart option was on, the stream changed its state to startLive as soon as I stopped sending data. It suggested that the encoder was trying to start but it was too slow to do it before some other data paket was received (I guess)
When I tried to stream to the "Stream now" URL, as #noogui suggested, it worked ! So I checked out what was the difference in the stream now & event configurations.
It turned out I just had to activate the low latency option as it's done by default in the stream now configuration.
I consider it as a temporary fix because I don't really know why the encoder isn't starting otherwise and because it doesn't work with the autostart option... So I hope it wont break again if Youtube does another change on their encoder.
So, if you have to work with the Youtube api, good luck guys !
I want to be able to load only the audio stream of the youtube video and process it (EQ, Effects, etc.) through a graph of Web Audio nodes.
Is this doable? Any open-source work out there, doing that?
Thanks in advance to all and any responses.
No, because you can't get audio streams cross-domain. (that is, if your code could be hosted on YouTube.com, sure, but not from mydomain.com.)
The reason for this (you CAN do it if CORS is set up, but it's not on YouTube) is because if you can get the audio stream, you can do a bit-copy of the data. Just like images, they don't want to leak the raw data.
i am streaming some FTA channels from
http://www.tbsdtv.com/products/tbs6985-dvb-s2-quad-tuner-pcie-card.html
using mediaportal
http://www.team-mediaportal.com/
and then i get rtsp url from mediaportal of channel i timeshift
and vlc i can send that stream to mediaserver FMS to get HLS, HDS, RTMP, RTSP
i have 3 servers running erlyvideo (flussonic)
so it take care of the delivery.
i want some alternate solution beside that
i have done some methods to work this our
including
VLC
IPTVL
Dvbdream
but the quality is better when i stream some thing as file, only FMLE works good with live stream, but for that we only can use directshow enabled devices like
http://www.viewcast.com/products/osprey-cards
i am doing it on windows.
if some one have any more methods or want to share his version please do so
There is a WWW page with Flash stream on it. I want to download and forward this stream to another streaming server, when possible - replace audio stream (e.g. translate), but without recompressing video stream. Usual way for this ATM is to capture and broadcast Flash player view from the web page, which is obviously suboptimal because video needs to be recompressed, making the quality notably worse and loading the cpu.
Has someone an idea how to do it? VLC seems to be able making relay, but it also seems not to support RTMP at all.
if you're ready to do this programmatically you can use crtmpserver (C++) or red5 (Java) with any RTMP client, otherwise this question doesn't belong to SO