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 !
Related
I'm trying to learn how to do pseudo streaming for MP4 files. I can't think of a good way to do it, but I just found a great example app has similar implementation (except I don't understand how it does it yet)
Here's the scenario:
Alice can send a video to Bob in the app
Bob can open it immediately and see Alice's video, from beginning, while Alice is still recording it
Also, Bob can choose to view the video later after Alice finished recording. But Bob should be able to view the video instantly without waiting too much time, even when the whole size of the video is large.
Thus, my hunch is, it's using some sort of pseudo streaming for mp4.
Here's the screenshots of the requests Alice's phone makes while using the example app:
The screenshot suggests, the example app is making an array of PATCH requests to their server, every 0.x seconds. And finally, the very last request will make a PATCH to update the moov information for this MP4.
Thus my question is, how is this implemented (any educated guess will be welcomed)? Or is there any sort of existing protocol/iOS encoder that I didn't know is doing this already?
Thanks a lot!
Reading the text of your question rather than the title, I think there are a number of likely steps:
Alice is recording video
She is ending the video to a streaming server
Alice notifies Bob that the stream is available and sends the URL on the streaming server that Bob can access to retrieve the stream
Bob's video client requests the stream, using range request to download it chunk by chunk
Have a server in the middle like this is a typical approach for any stream which may have more than one client watching it.
More sophisticated streaming servers may also support delivery the stream in different bit rates and even encoded with different codecs for maximum device reach.
There are commercial (e.g. https://www.wowza.com) and open source streaming servers (e.g. https://gstreamer.freedesktop.org) you can look at to get more info on streaming servers and to see some examples.
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.
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?
I am doing the functionality of downloading video and playing in offline mode. Here I am using NexPlayer with GCDWebServer my videos are encoded and needs to sync with server. I am using GCDWebServer for offline mode but after some video play its starts buffering.
So my question is this is offline server and we already having all data so why its buffering, I am not getting this. Please suggest something or can I use any other server instate of GCDWebServer.
Sorry to late reply but I found the solution for my problem.
Actually I set some buffering values with NexPlayer as online video playing but as we have a file locally stored, we don't want any buffing values to interrupt the process.
So I just removed them, simply set to 0 and it works perfectly as per expectations.
No buffing, happy life :)
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