I am new in j2me developing world.
I just want to know that how to get audio frequency from the audio recording application which stores data in .amr file.
Please help me, I tried a lot, but I am helpless.
So any idea regarding this will be appreciated.
thanks in advance.
im gonna ad here what i have found from the other sites that may be useful to you and me(as a newbie)
http://www.developer.nokia.com/Community/Discussion/showthread.php?154169-Getting-Recorded-Audio-Frequency-in-J2ME
If you want frequency of sound in Hz then it is actually not a single value but a series of values as a function of time.
You will have to calculate fourier transform of the sound samples which will give you frequency.
Read about this on wikipedia on how to calculate fourier transform and frequency graph...
http://www.developer.nokia.com/Community/Discussion/showthread.php?95262-Frequency-Analysis-in-J2ME-MMAPI
this forum says something about fft(fast fourrier transform) and analysing recorded amr sound rather than processing live stream and provides 3 link about fft which are right underneat this line have a look at them:..
look at the site mobile-tuner.com. (im new too. in fact i know nothing about java.)
but the site says that tuner function enabled phones are s60 phones. i was trying to write guitar tuner program. since my phone is nokia 5310 express music which is s40 i gave up.
so good luck to you
note: javax.microedition.media.control.RecordControl
--i don't know too much but i have a hunch about that ""Record Control"" class or function is related to audio frequency function in j2me. and the frequency analysis thing is inside the "sound processing"
Related
I am using Media Recorder class to record sound . and I am getting amplitude on some interval of time and converting it into decibel, but what I want I also want to get frequency of audio on that interval with corresponding to that amplitude or decibel. I searched, but I did not get proper idea that how to do it.
please if someone can guide me then please help
The process most often used to determine pitch is called the "fast fourier transform". Try using those keywords or the common abbreviation "FFT" along with the language or platform you are working on and that should bring up libraries to incorporate that can do this. Coding an FFT is pretty complex, so you'll probably want to use a library. But if you are curious about the math and how they work, check out The Scientists and Engineers Guide to Digital Signal Processing.
I'm a beginner with tensorflow and Python and I'm trying to build an app that automatically detects, in a football (soccer) match some key moments (yellow/red cards, goals, etc).
I'm starting to understand how to do a video analysis training the program on a dataset built by me, downloading images from the web and tagging them. In order to obtain some better results for the analysis, I was wondering if someone had some suggestions on tutorials to follow in order to understand how to train my app also on audio files, to make the program able to understand when there is a pitch variation in the audio of the video and combine both video and audio analysis in order to get better results.
Thank you in advance
Since you are new to Python and to tensorflow, I recommend you focus on just audio for now, especially since its a strong indicator of events of importance in a football match (red/yellow cards, nasty fouls, goals, strong chances, good plays, etc).
Very simply, without using much ML at all, you can use the average volume of a time period to infer significance. If you want to get a little more sophisticated, you can consider speech-to-text libraries to look for keywords in commentator speech.
Using video to try to determine when something important is happening is much, much more challenging.
This page can help you get started with audio signal processing in Python.
https://bastibe.de/2012-11-02-real-time-signal-processing-in-python.html
I am quite new to LabVIEW and NI devices.
I am working on Active Noise Cancellation Project, where I will be using two microphones input and one loud speaker as output. I have NI myRIO 1900 and CDAQ 9178 devices in our university lab. I need to do real time audio processing, I will collect data from microphone and process it using filtered XLMS algorithm to produce anti noise from loud speaker and other microphone is error microphone. I want to process data so quickly( within 1.7 msec ) so I will have real time response at 44100 sample rate !! My question is , 'is it possible to do with labview ?? and is stream processing possible in labVIEW?? and can I achieve so small audio latencies as mentioned above ??'
I have searched for audio processing objects in labview help. I can only find 'Acquire Sound', 'Play Waveform', surprisingly 'Acquire Sound configuration ' will work only for duration of minimum of 1 second not less than that !!! I can't input the time milli seconds !!!( I am still facing problem installing myRIO, so I have used host computed VI to do this.)
Please help !! Thank You
The thing you should be looking into is the FPGA part of the myRIO. You’re never going to be able to get 1.7ms response time via the host computer. The FPGA can access the Analogue inputs and outputs, so if you can get your algorithm to compile onto the FPGA then it should work.
Yes, it is possible with LabVIEW, insofar as any algorithm you want to code up can be executed by LabVIEW. If you're asking whether there is a library that already exists to do the filtering you're wanting to do, you may want to explore the NI Sound & Vibration toolkit, which is sold separate from LabVIEW, or explore third-party libraries.
The raw waveform mathematics abilities that come with LabVIEW are fairly extensive. You should be able to code whatever transforms you want if you know the base math.
I have a situation where I have a video capture of HD content via HDMI with audio from a sound board that goes through a impedance drop into a microphone input of a camcorder. That same signal is split at line level to a 'line in' jack on the same computer that is capturing the HDMI. Alternatively I can capture the audio via USB from the soundboard which is probably the best plan, but carries with it the same issue.
The point is that the line in or usb capture will be much higher quality than the one on HDMI because the line out -> impedance change -> mic in path generates inferior quality in that simply brushing the mic jack on the camera while trying to change the zoom (close proximity) can cause noise on the recording.
So I can do this today:
Take the good sound and the camera captured sound and load each into
audacity and pretty quickly use the timeshift toot to perfectly fit
the good audio to the questionable audio from the HDMI capture and
cut the good audio to the exact size of the video. Then I can use
ffmpeg or other video editing software to replace the questionable
audio with the better audio.
But while somewhat quick and easy, it always carries with it a bit of human error and time. I'd like to automate this if possible as this process is repeated at least weekly throughout the year.
Does anyone have a suggestion if any of these ideas have merit or could suggest another approach?
I suspect but have yet to confirm that the system timestamp of the start time may be recorded in both audio captured with something like Audacity, or the USB capture tool from the sound board as well as the HDMI mpeg-2 video. I tried ffprobe on a couple audacity captured .wav files but didn't see anything in the results about such a time code, but perhaps other audio formats or other probing tools may include this info. Can anyone advise if this is common with any particular capture tools or file formats?
if so, I think I could get best results by extracting this information and then using simple adelay and atrim filters in ffmpeg to sync reliably directly from the two sources in one ffmpeg call. This is all theoretical for me right now-- I've never tried either of these filters yet-- just trying to optimize against blind alleys by asking for advice up front.
If such timestamps are not embedded, possibly I can use the file system timestamp for the same idea expressed in 1a, but I suspect the file open of the two capture tools may have different inherant delays. Possibly these delays will be found to be nearly constant and the approach can work with a built-in constant anticipation delay but sounds messy and less reliable than idea 1. Still, I'd take it, if it turns out reasonably reliable
Are there any ffmpeg or general digital audio experts out there that know of particular filters that can be employed on the actual data to look for similarities like normalizing the peak amplitudes or normalizing the amplification of the two to some RMS value and then stepping through a short 10 second snippet of audio, moving one time stream .01s left against the other repeatedly and subtracting the two and looking for a minimum? Sounds like it could take a while, but if it could do this in less than a minute and be reliable, I suspect it could work. But I have only rudimentary knowledge of audio streams and perhaps what I suggest is just not plausible-- but since each stream starts with the same source I think there should be a chance. I am just way out of my depth as to how to go down this road, so if someone out there knows such magic or can throw me some names of filters and example calls, I can explore if I can make it work.
any hardware level suggestions to take a line level output down to a mic level input and not have the problems I am seeing using a simple in-line impedance drop module, so that I can simply rely on the audio from the HDMI?
Thanks in advance for any pointers or suggestinons!
Having just witnessed Sound Load technology on the Nintendo DS game Bangai-O Spritis. I was curious as to how this technology works? Does anyone have any links, documentation or sample code on implementing such a feature, that would allow the state of an application to be saved and loaded via audio?
Its the same old thing used in ZX Spectrum era. You load programs/games from tape.Only the sound quality and the filters are probably better.
In my opinion something like Bluetooth or WiFi is better. You can also send files that can be put on some storage and then load them. I find these methods much easier than sound because if there is a lot of noise around you cannot do much.
It is just a conversion of data to audio and then back from audio to data.
Search for Zotyocopy and Copy86M on google - these are the utilities used for saving a game to tape after loading it into memory on zx spectrum.
If you want to pass data as audio through the air there are a few things you need to be aware of though, such as how the speaker and microphone interact for example. It is important that they don't distort or alter the sound too much as what you are sending are in fact the raw bytes.
Some audio software will let you open any file as audio so that you may listen to it. If you record audio as data do not use lossy compression such as mp3 on the audio file!