I have a server that stream mp4(h264). I use MP4Box to put the moov atom at the beginning of a file and interleave default 500 ms.
However, I noticed that at peak times when the server is busy, the files start streaming slower, but not same slower, large videos (one hour or more) start much slower than small files.
I read about Atom Moov being processing slower in a lighttpd with h264 streaming module like mine...
Any way I can speed up playback start to about 2 seconds, right now it's about 7 for large files...
You can use mp4parser to see which part of moov box is getting bigger with the increasing file size. Then may be you can look for an optimal way of representing the box. I think it is sample size box (stsz). Also, I can think of segmenting MP4 so that the header overhead is spread across the file. MP4Box does support segmenting MP4 files. But then you need to check whether your client is capable of understanding this format.
Related
We have a project which will generate lots (hundreds of thousands) of .PNG images that are around 1mb. Rapid serving is not a priority as we use the images internally, not front end.
We know to use filesystem not DB to store.
We'd like to know how best to compress these images on the server to minimise long term storage costs.
linux server
They already are compressed, so you would need to recode the images into another lossless format, while preserving all of the information present in the PNG files. I don't know of a format that will do that, but you can roll your own by recoding the image data using a better lossless compressor (you can see benchmarks here), and have a separate metadata file that retains the other information from the original .png files, so that you can reconstruct the original.
The best you could get losslessly, based on the benchmarks, would be about 2/3 of their current size. You would need to test the compressors on your actual data. Your mileage may vary.
I have two stereo wav files that I would like to take the left channel of the first audio file and take the right channel of the second audio file and join them into one new wave file.
Here's an image of what I'm trying to do.
I know I can read files into matlab / octave and get the separate left right channels with the code below:
[imported_sig_1, fs_rate, nbitsraw] = wavread(strcat('/tmp/01a.wav'));
imported_sig_L=imported_sig_1(:,1)';
[imported_sig_2, fs_rate, nbitsraw] = wavread(strcat('/tmp/02a.wav'));
imported_sig_R=imported_sig_2(:,2)';
I can then write the new channels that I want out using the code
wavwrite([(imported_sig_L)' (imported_sig_R)'] ,fs_rate,16,'newfile.wav'); %
The problem I'm running into is the time it takes to import the file and size of the array the wave files take up. The files I'm importing are about 1-4 hours long and it takes a while to import and it takes a lot of memory in the array is there away around importing the full file and then exporting them?
I'm using octave 3.8.1 on Ubuntu 14.04 which is like matlab but I also have access to sox
I assume the bottleneck is your hard drive and your system has sufficient memory to keep all thee files in memory at the same time. If so you won't gain an speed reading only one channel. With a 16 bit wav your HDD would have to skip 2 bytes, read 2 bytes, skip 2 bytes, read 2 bytes... For such a read operation it is much faster to copy the full file into the memory and remove the unwanted channels afterwards.
I am trying to read many video files from a database and process them. I am using Matlab and my problem is that when I want to read a 10 minutes long full HD video I should wait so much and my computer stops performing well. I use this command
VideoReader('movie.mp4')
I have seen that it takes 47 seconds to read a 30 seconds long video in the same format. I do not need to load all frames into my memory I just need 11 frames for each step of my process and really got stock here. Any help will be appreciated.
Also here is my output when I run this command
disp(videoObj);
output:
Summary of Multimedia Reader Object for 'movie.mp4'.
Video Parameters: 30.00 frames per second, RGB24 1280x720.
1482 total video frames available.
By the way I am running my code on Matlab R2014a and my OS is ubuntu 14.0.4.
Siavash,
The long loading time is because the entire file is scanned to determine the number of frames. This process is necessary to support frame indexed based access. In R2014b and higher, the frame counting during construction has been disabled. Additionally, you can seek to specific locations in the file using the CurrentTime property and use the hasFrame/readFrame methods for reading to avoid this performance penalty
Dinesh
I'm using version R2015b and I find the same slow processing of mp4 files with the VideoReader and readFrame functions. However, I find that those functions perform much faster on an avi file than an mp4, so I first convert the mp4 to avi using an independent program from https://www.ffmpeg.org. I don't know why there's such a difference in speed between the two...perhaps someone from MATLAB can provide some insight into that question.
What are similar compressors to the RAR algorithm?
I'm interested in compressing videos (for example, avi) and images (for example, jpg)
Winrar reduced an avi video (1 frame/sec) to .88% of it's original size (i.e. it was 49.8MB, and it went down to 442KB)
It finished the compression in less than 4 seconds.
So, I'm looking to a similar (open) algorithm. I don't care about decompression time.
Compressing "already compressed" formats are meaningless. Because, you can't get anything further. Even some archivers refuse to compress such files and stores as it is. If you really need to compress image and video files you need to "recompress" them. It's not meant to simply convert file format. I mean decode image or video file to some extent (not require to fully decoding), and apply your specific models instead of formats' model with a stronger entropy coder. There are several good attempts for such usages. Here is a few list:
PackJPG: Open source and fast performer JPEG recompressor.
Dell's Experimental MPEG1 and MPEG2 Compressor: Closed source and proprietry. But, you can at least test that experimental compressor strength.
Precomp: Closed source free software (but, it'll be open in near future). It recompress GIF, BZIP2, JPEG (with PackJPG) and Deflate (only generated with ZLIB library) streams.
Note that recompression is usually very time consuming process. Because, you have to ensure bit-identical restoration. Some programs even check every possible parameter to ensure stability (like Precomp). Also, their models have to be more and more complex to gain something negligible.
Compressed formats like (jpg) can't really be compressed anymore since they have reached entropy; however, uncompressed formats like bmp, wav, and avi can.
Take a look at LZMA
I need to know if it is possible to create a 30 second sample MP3 from a WAV file. The generated MP3 file must feature a fade at the start and end.
Currently using ffmpeg, but can not find any documentation that would support being able to do such a thing.
Could someone please provide me the name of software (CLI, *nix only) that could achieve this?
This will
trim out from Position 45 sec. the next 30 seconds (0:45.0 30) and
fade the first 5 seconds (0:5) and the last 5 seconds (0 0:5) and
convert from wav to mp3
sox infile.wav outfile.mp3 trim 0:45.0 30 fade h 0:5 0 0:5
Check out SoX - Sound eXchange
I have not used it myself but one of my friends speaks highly of it.
From web page (highlighted my me):
SoX is a cross-platform (Windows,
Linux, MacOS X, etc.) command line
utility that can convert various
formats of computer audio files in to
other formats. It can also apply
various effects to these sound files,
and, as an added bonus, SoX can play
and record audio files on most
platforms.
The best way to do this is to apply the 30-second truncation, fade in and fade out to the WAV audio data before converting it to an MP3. If your conversion library has a method that takes an array of samples, this is very easy to do. If the method only accepts a WAV file (either in-memory or on disk), then this is slightly less easy as you have to learn the WAV file format (which is easy to write but somewhat more difficult to read). Either way, applying gain and/or attenuation to time-domain sample data (as in a WAV file) is much easier than trying to apply these effects to frequency-domain data (as in an MP3 file).
Of course, if your conversion library already does all this, it's best to just use that and not worry about it yourself.