I have a surveillance .avi video file that includes mostly still (unchanging) frames. However, certain objects (and people) enter the field of view and exit within short period of time. I need to extract those frames.
The entire video lasts very long (24+ hours) which prevents me from replaying realtime. Is there a Perl module that can parse the file by frames? With that I suppose I can quickly transform the frames to some histogram and compare the neighboring frames to find significant changes. This webpage spells out the general approach but I couldn't find either the script (keyframe.pl) or the modules used (granted that is for .mpeg files).
If the direct module to parse .avi does not exist, I can convert the video to .mpeg (or any other form), then please just direct me to some other module that can parse the file?
Thx!
I vote for the original comments, although I'm a Perl person, it's wrong for this. I'd just add that for surveillance and movement, there are two well-known tools: http://www.lavrsen.dk/foswiki/bin/view/Motion/WebHome motion and zoneminder: http://www.zoneminder.com/ that work in the area of detection. motion is built on top of ffmpeg, I'm not sure whether it can process files though.
Related
Is there a way to read .avi video file frame by frame using mmread or other function, similar to using videoReader, and readFrame functions?
I used mmread but it took a long time to read each frame as it read all the frames before the specified frame.
In general the media and movie reading in Matlab is cumbersome. I changed to python because of it. I would recommend to split the movie into single image files using tools like avconv or ffmpg and then work on the images, as the image manipulation toolbox is much faster.
If you want to go forward and backward through any movie, especially backwards is very slow. Very often it is implemented, by rewinding and than seeking out your frame, so basically to go from frame 100 to 99, Matlab rewinds to 0 and then seeks through the binary stuff to 99.
I am studying rat ultrasonic vocalisations (their speech in ultrasound). I have several audio wav files of the rats speeches. Ideally, I would import the whole file into matlab and just process it but I will get memory issues even with the smallest 70mb file. This is what I want help with.
[y, Fs, nbits] = audioread('T0000201.wav');
[S F T] = spectrogram(y,100,[],256,Fs,'yaxis');
..
..
..rest of program
I could consider breaking the audio (in one file) into blocks, and process the block before considering the next block, but I'm not sure what I would do for cases where rat calls are cut off half way through, at the end of the blocks (this might have a negative impact on the STFT spectrogram).
I came across another technique called "Mapreduce" which seems to allow me to use the entirety of my data without actually reading it in. While this seems most ideal, I don't quite understand how it works or can be implemented. "Hadoop" has also been mentioned. Can anyone provide any assistance?
I am currently using this (http://uk.mathworks.com/help/matlab/import_export/find-maximum-value-with-mapreduce.html) for reference. My first step was trying to use the wav file as the data store (like the csv file in the example) but that didn't work.
Since you're working primarily with a repository of audio (.wav) files, mapreduce might not be your best option. The datastore function only works with text files or key-value files.
Use the memory function to explore what the limits of memory are for MATLAB, and try processing the audio files in smaller blocks as you mentioned. Using a combination of audioread(), audioinfo(), and audiowrite(), you can break your collection of audio files up into a larger collection of smaller files that can then be individually processed.
If you have a small number of files to work with, then you can manually inspect the smaller blocks to make sure no important rat calls are cut off between blocks. Of course if you have thousands of files to work with then that approach won't be feasible.
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.
I'm making an Iphone game, we need to use a compressed format for sound, and we want to be able to loop SEAMLESSLY back to a specific sample in the audio file (so there is an intro, then it loops back to an offset)
currently THE ONLY export process I have found that will allow seamless looping (reports the right priming and padding frame numbers, no clicking when looping ect) is using apple's afconvert to a aac format in a caf file.
but when we try and encode to lower bitrates, it automatically re samples the sound! we do NOT want to have the sound re sampled, every other encoder I have encountered has an option to set the output sample rate, but I can't find it for this one.
on another note, if anyone has had any luck with seamless looping of a compressed file format using audio queues, let me know.
currently I'm working off the information found at:
http://developer.apple.com/mac/library/qa/qa2009/qa1636.html
note that this DID work PERFECTLY when I left the bitrate for the encode at default (~128kbs) but when I set it to 32kbps - with the -b option - it resampled, and looping clicks now.
It needs to be at least 48kbps. 32kbps will downsample to a lower sample rate.
I think you are confusing sample rate (typical values: 32kHz, 44.1kHz, 48kHz) and bit rate (typical values: 128kbps, 160kbps, 192kbps).
For a bit rate, 32kbps is extremely low. Sound will have bad quality at this bit rate. You probably intended to set the sample rate to 32kHz instead, which is also not outright typical, but makes more sense.
When compressing to AAC and uncompressing back to WAV, you will not get the same audio file back, because in AAC, the audio data is represented in a completely different format than in raw wave. E.g. you can have shifts by few microseconds, which are necessary to convert to the compressed format. You can not completely get around this with any highly compressed format.
The clicking sound originates from the sudden change between two samples which are played in direct succession. This is likely taking place because the offset to which you jump back in your loop does not end up to be at exactly the same position in the AAC file as it was in the WAV file (as explained above, there can shifts by microseconds).
You will not get around these slight changes when compressing. Instead, you have to compensate for them after compression by adjusting the offset. That means you have to open the compressed sound file in an audio editor, e.g. Audacity, and manually find another offset close to the original one, which is suitable for looping.
How to find an offset which is suitable for looping?
Zoom in to the waveform's end. Look at how the waveform looks there. Then zoom in to the waveform at the original offset and search in its neighbourhood for an offset at which the waveform connects seamlessly to the end of the waveform.
For an example how this shoud look like, open the uncompressed audio file in the audio editor and examine the end of the waveform and the offset there.
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.