so, there's a "trick" that lets you manipulate video file durations and I would like to know if there is a way to 100% set the exact duration you want without to much trial and error(?). thanks in advance.
After searching and trying stuff for hours, I found this video.
Look in the comments for UNICORNTEARZ13's thread, and someone named Stop! You violated the law! explained that the duration value after marker 44 89 in WebM files is IEEE754 float hexadecimal. I then used to site to get the exact value I wanted and voilà !
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I have coded two versions of video player based on QMediaPlayer and on Vlc-qt. In both cases I have the incorrect value for the full time video. The player show me that total time is 7 seconds, but in fact the time approx. 5 minutes. And of course, the slider of position shows not correct.
I was confused, so that maybe I did something wrong. But this video file was tested with MS video player, and I see the same problem.
Video for testing can be found at https://1drv.ms/u/s!AgCzZ90Ttbz65jqiluS2NS95Id0U
My guess is that the file contains the wrong information about the time. Or maybe not codec provides such information in not correct way.
Can anybody clarify to me what the reason of the problem and how it should be fixed.
Let's say I have some numbers, like
5,10,7,8,9,6,2,4,8,5,3,9,78,5,6
I need to send this to another computer, but as the least number of possible bytes. I know what there is a way to do that, I just forgot what it's called and how it works, but generally doing some math with those numbers, getting a big number that, from this number, I'll be able to export the data and get this numbers from this number. Thanks in advance.
EDIT
OK so I need to send this text in UDP but I need it as less bits as possible. I'm sending some options, like firstcolor-secondcolor, let's say I have 15 colors. Every color is just number, from 1 to 199, but maybe there is a better way to send this data? thanks.
No one can say which compression scheme is the best for you. We don't have any information about the numbers. But as a first try, you could just write them into a file and use gzip compression on it. Or bzip2, or 7zip.
And only if all these don't help, you should think about doing the compression yourself.
You also didn't tell us your operating systems (source computer, destination computer) and from where you get the data.
[Update, based on the edit in the question:] So basically you want to send some numbers in the range of 1 to 199. This is pretty close to what a single byte can hold.
If it is ok that you use 8 bits per number (meaning you waste 0.4 bits per number), this is trivial but highly depends on the programming language. Here is how it might look like in Java syntax:
ByteBuffer buf = new ByteBuffer();
buf.add(1);
buf.add(199);
buf.add(78);
buf.add(7);
udpSocket.send(buf.toArray());
Get a compression library (like zlib, for example) and feed your numbers in (as an array of integers, for example). This is compressing your data. That same library should allow you to reverse the process and decompress the data at the other end to get your values back out.
If you want to improve your algorithmic knowledge and your requirements are simple and non-critical I'd recommend having a go at writing your own compression/decompression code. If not, grab some code off the shelf - there are loads of good libraries around.
I am using AVAudioPlayer.
I have a short sound in my application.
Randomly it does not play , Play API does not return any error,
Volume is also proper that moment. I am not getting what is going wrong.
Interesting thing is If I play some other sound file, It does play properly every time.
Then I did check properties of both the sound files.
The both the sound files have same bit rate, only difference is
The one which does not play randomly has 00.02 duration.
and the one which play properly has 00.00 duration.
Can anybody give me idea to fix this issue?.
I'm writing software which is demonstraiting video on demand service. One of the feature is something similiar to IIS Smooth Streaming - I want to adjust quality to the bandwith of the client. My idea is, to split single movie into many, let's say - 2 seconds parts, in different qualities and then send it to the client and play them. The point is that for example first part can be in very high quality, and second in really poor (if the bandwith seems to be poor). The question is - do you know any software that allows me to cut movies precisly? For example ffmpeg splits movies in a way that join is visible and really annoying (seconds are the measure of precision). I use qt + phonon as a player if it matters. Or maybe you know any better way to provide such feature, without splitting movie into parts?
Are you sure ffmpeg's precision is in seconds? Here's an excerpt from the man page:
-t duration
Restrict the transcoded/captured video sequence to the duration specified in seconds. "hh:mm:ss[.xxx]" syntax is also supported.
-ss position
Seek to given time position in seconds. "hh:mm:ss[.xxx]" syntax is also supported.
-itsoffset offset
Set the input time offset in seconds. "[-]hh:mm:ss[.xxx]" syntax is also supported. This option affects all the input files that follow it. The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by 'offset' seconds.
Looks like it supports up to millisecond precision, and since most video is not +1000 frames per second, this would be more than enough precision to accurately seek through any video stream.
Are you sure this is a good idea? Checking the bandwidth and switching out clips every two seconds seems like it will only allow you to buffer two seconds into the future at any given point, and unless the client has some Godly connection, it will appear extremely jumpy.
And what about playback, if the user replays the video? Would it recalculate the quality as it replays, or do you build the video file while streaming?
I am not experienced in the field of streaming video, but it seems what I see most often is that the provider has several different quality versions of their video (from extremely low to HD), and they test the user's bandwidth and then stream at an appropriate quality.
(I apologize if I misunderstood the question.)
I am using the below jQuery plugin for playing mp3
www.happyworm.com/jquery/jplayer
However, there is a bug in Flash that the total play (track) time won't show up correctly UNTIL AFTER the whole mp3 is completed downloaded.
I wonder if there is a way to work around this to get the correct total time using either javascript / another flash / even backend library in ASP.NET. Any suggestion helps. Thanks
You sure that's a bug? Looking at the header definition for the MP3 format I don't see any values for the length of the file. Generally applications that play MP3s would have to calculate the time, and that may not be doable until the entire file is downloaded. So the behavior you're seeing from Flash might be expected.
Theoretically if it's a fixed bitrate file (as opposed to VBR) then knowing the bitrate (gotten from the header) and the total size of the file should be enough to calculate it. However, the server would have to report the size of the file in the response headers (and that's not guaranteed to be accurate).
My guess is you'd need some service on the server that could calculate the length and report that to you in a separate request.