I wish to decode 608/708 captions from an HEVC encoded video wrapped within an MPEG TS stream. I cannot find any information as to how to locate the closed caption data. Is anybody able to provide any assistance with this regard?
I guess HEVC streams store CEA 608/708 captions in SEI T35 registered data, like H264 does.
May be you could add a link of the HEVC stream to help confirmation.
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In microsoft speaker recognition api what data we have to use for body parameter.
There written as "binary data" . Does this imply that we have to convert audio file to binary data and then paste there.
Currently, it's not possible to use this API testing console for posting application/octet-stream or application/form-data. I believe there's some work on this, and it should be available soon.
As an alternative, you can use PostMan or Fiddler. PostMan might be easier to use. Give it a try and let me know if you have a problem.
The binary data needs to be a WAV file of a specific format:
Container: WAV
Encoding: PCM
Rate: 16K
Sample Format: 16 bit
Channels: Mono
You can check out a working example web page over here - I've used an altered copy of recorderjs (altered by reverse engineering the Speaker Recognition API examples page) to get the right bit rate and sample rate WAV:
https://rposbo.github.io/speaker-recognition-api/
You could potentially use the test console, since you can send base64 encoded audio data (as the official demo page does):
https://azure.microsoft.com/en-gb/services/cognitive-services/speaker-recognition/
I am developing an iOS app that synchronises with GoPro cameras.
One of the feature requires downloading MP4 from the GoPro (potentially huge).
I basically have a url like: http://10.9.9.5/whatever/video.mp4.
However, I only need parts of the video, let's say between 1:00 and 1:05.
I am thinking on downloading just parts of the MP4, using HTTP "Range" header. I believe that it's possible and I will get a bunch of bytes.
However, is it a valid file? Will I be able to create a MP4 ? Do I need the MP4 header with meta information? Do any of you faced this kind of challenge?
I am using Objective C but I believe that this is a general question.
The MP4 file is a container for video that is structured around something called boxes. Probably you'll have h.264 video in that MP4 file, knowing that, you'll need to know the structure of the file you are trying to chunk.
Depending on the way it is encoded you'll have to look for a box with metadata that'll allow you to search for the correct part of the file either at the beginning or at the end, but you'll have to reconstruct a valid MP4 with the data you get from the original file.
You can see a reference of the file format here http://xhelmboyx.tripod.com/formats/mp4-layout.txt.
What is required to use SMIL file to utilize adaptive streaming in a videojs player. I have created the SMIL file in my wowza application and it is creating my 4 separate streams and making them available. However I cannot get my webpage, that uses videojs, to correctly play the SMIL file. Hints on that coding or where to go to find the correct documentation would be greatly appreciated.
There aren't many implementations of SMIL players. I'm sure I've seen wowza URLs that suggest it will output the SMIL as other formats, something like whatever.smil/manifest.m3u8. That's HLS which could be played on mobile and Safari natively and with videojs-contrib-hls elsewhere.
I know the question is old, but I've been struggling with this recently, so I want to share my experience in case anyone is interested. My scenario is very similar: want to deliver adaptive bitrate streaming from Wowza to clients using videojs.
There is a master link that explains how to setup and run Wowza Transcoder for live streaming, and how to set up your Adaptive Bitrate Streams using an SMIL file. Following the video in there you can achieve to have a stream that uses ABS, but the SMIL file is attached to the stream name, so it is not a solution if you have streams that come to Wowza from another Media Server origin and that need to be transcoded before being served to the clients. In the article there are a few key things mentioned (like the Stream Name Groups), but somehow things doesn't seem pretty clear, at least to me. So here is some clarification from what I understood from all articles I read and what I did to achieve ABS:
You can achieve ABS in Wowza either with SMIL files or with Stream Name Groups (NGRP). NGRP refres to a block of streams that is defined in the Transcoder template that can be played back using multi-bitrate streaming (dynamically) (<- this is what I used). And SMIL files are used to create a "static" list of streams for multi-bitrate VOD streaming. If you are using Wowza Origin-Edge Delivery you'll need the .smil file, because NGRP do not get forwarded to the edge. (Source for all this information: here).
In case you need the SMIL file, you probably need to generate a new one for every stream, and probably you want to do that in an automated way, so best way would be through an HTTP request (in the link above it is explained how to achieve this).
In case you can live with NGRP, things are a bit easier:
You need to enable Wowza Transcoder (this is pretty easy and steps are in the video I mention above).
You should create your own Transcoder Template with the different stream presets you want to deliver, as an example you can check the default ones that are already there. The more presets you add, the more work Wowza will need to do whenever a stream comes, since it will need to generate a new stream for every preset that you have defined.
Now is when we generate the NGRPs. In your Transcoder Template, you can generate as many NGRPs as you want (to clarify: these are like groups of streams, that you will be able to set in your clients video player. Each NGRP contains the streams that the video will be able to use when doing the adaptive bitrate streaming). For instance, these are the default NGRPs:
If you play the ngrp "_mobile" in the clients video player, the ABS algorithm in the player will be able to adapt itself to play either the 240p or the 160p streams based on the client capabilities.
So imagine you have these two NGRP. In order to play them in videoJS, you will need to set the source to:
http://[wowza-ip-address]:1935/<name-of-your-application>/ngrp:myStream_all/playlist.m3u8
or
http://[wowza-ip-address]:1935/<name-of-your-application>/ngrp:myStream_mobile/playlist.m3u8
... based on how many options you want to provide to the client player to use for the ABS. (For instance: if your targets are old mobile devices, you probably just want to offer a couple of low bitrate streams).
(This would be in case you're delivering an HLS stream. If other format, the extension would change, for instance if you are delivering a DASH stream you would have "/manifest.mpd" instead of "playlist.m3u8").
That is all, there is also a very helpful link in video.js documentation explaining how it does the bitrate switching: here.
I hope it helps someone! At least clarifying things! :)
Looking first for links to good documentation that correctly explains pseudostreaming, byte range requests and mp4 fragmenting. Note, I will be using only the mp4 container (h264 codec) and HTML5 video (no flash).
My understanding of pseudostreaming is that the client can send off a start parameter that the server "seeks" to in it's response. MOOV data must be upfront and it implicitly implies that buffering of the original source stops in favor of the new response starting at the "start"/seek position. How is the client forced to make pseudo calls? Does the MP4 have to formatted in a special way?
Byte range requests are different send rather than just a start parameter a range is sent. Sounds more like progressive downloading. How would "seeking" work? Does it with byte range? Can the segment size be pre-determined with movie box information?
How does MP4 fragmentation fit in? Looking like a construct originally designed by microsoft for silverlight. But is it applicable to other browser html5 video implementations?
Finding it difficult to sort out information on the web. Looking to both live feed and take historical segments of h264 files produced from rtp camera streams. Got a bunch of files time-ordered in a MongoDB. Created my own h264 decoder in JavaScript and can construct mpeg-dash boxes on the fly off a range query. Using Chrome's support for MSE to append segments. Works great but not a universal solution. Want to fall back on other techniques other than flash but with the html5 video.
I'm wondering if I can create an application that converts a youtube link to mp3 and save it on a desktop. Does anyone have any suggestions to do this? Any references of links?
Do you mean capturing video - extracting a soundtrack from it and save it as an mp3? There are tools for doing just that called audio extractors. It's possible that some of these are available as libs. Here's link to one of the tutorials that may give you some ideas
You could have a look at how the Easy YouTube Firefox add-on does it.
Try KissYouTube. Extract the audio stream from the flv file to whatever audio format you want.