I want to split a Multichannel (2,8 or 16) wav file into its channels and save every channel in another wav-File.
So far I've accomplished to get libsox up and running in my c++, objective c++ project.
Libsox isnt well documented and there aren many examples on how to do it :(
I started by first openning the Inputfile
sox_format_t * in, * out;
assert(sox_init() == SOX_SUCCESS);
assert(in = sox_open_read((const char*)filename.c_str(),NULL,NULL,NULL));
Now I must find a way to get the number of channels of this file. Then I have to create the same amount of out-files and save every channel itself inside them.
How to do?
Thanks!
I think I will do it the old fashioned way.
Determine the channel count of the file.
Determine the length of the Data block.
Length of Data block / channelCount = Size of each channelBlock
Channels are saved like that inside a WavFile-Datablock (for 4Channel WavFile)
CH1/Ch2/Ch3/Ch4 CH1/Ch2/Ch3/Ch4.
I run through the datablock extract the channels and put them into a mono-wav-file
Related
I need to get waveform data from the wav file,but my code returns not right waveform (i compare my results with waveform from fl studio)
This is my code:
path = "/storage/emulated/0/FLM User
Files/My Samples/808 (16).wav";
waveb = FileUtil.readFile(path);
waveb = waveb.substring((int) (waveb.indexOf("data") + 4), (int)(waveb.length()));
byte[] b = waveb.getBytes();
for(int i= 0; i < (int)(b.length/4); i++) {
map = new HashMap<>();
map.put("value", String.valueOf((long)((b[i*4] & 0xFF) +
((b[i*4+1] & 0xFF) << 8))));
map.put("byte", String.valueOf((long)(b[i*4])));
l.add(map);
}
listview1.setAdapter(new
Listview1Adapter(l));
( (BaseAdapter)listview1.getAdapter()).notifyDataSetChanged();
My results:
Fl studio mobile results:
I'm not sure I can help, given what I know off of the top of my head, but perhaps this will trigger some ideas in your search for a solution.
It looks to me like you are assuming the sound file is 16-bit stereo, little-endian, and that you are only attempting to inspect one track of the stereo frame. Can you confirm this?
There's at least one way this plan could go awry: the .wav header may be an odd number of bytes in length, and you might not be properly parsing frame boundaries as a result. As an experiment, maybe try adding a different increment when you reference the b[] array? For example b[i4 + 1] and b[i4 + 2] instead of b[i4] and b[i4 + 1]. This won't solve the general problem of parsing .wav headers, but it could at least get you closer to understanding the situation.
It sure looks like Java's AudioInputStream is not accessible in Android, and all searches that I have that ask if there is an Android equivalent are turning up unanswered.
I've used AudioTrack for the playback of raw PCM, but I don't know an Android equivalent for reading wav files. The AudioRecord class and read() methods look interesting as the read methods store PCM data in a short array, but I've never used them, and they seem to be hard-coded to the microphone for input.
There used to be a Google Group: andraudio#googlegroups.com. IDK if it is still around. I used to go there and occasionally ask about things.
Maybe there is code you can use from Oboe or libGDX? The latter makes use of OpenAL and is for cross-platform development, with Android as one of the target platforms. I have not looked into either for this question.
If you do find the answer, it would be great to post it as a solution. This seems to be a matter that many have tried to solve and given up on.
I'm looking for a way to shorten a sentence (a text of few lines) to produce a "readable" (not too long) file name.
The application scenario is a chatbot where user can submit a media, say a video, with some paired description text (a caption). The application would assign to the video a readable file name, to retrieve afterward the video by his file name.
Imagine a video paired with a more or less long text description of the scene, like by example:
const videoDescription = 'beautiful yellow flowers on foreground, with a background with countryside meadows and many cows'
How could I shorten the description above with a "suitable" short file name?
Ok, I could just give the sentence as a name, maybe something a bit sanitized, like:
const videoFileName = 'beautiful_yellow_flowers_on_foreground_with_a_background_with_countryside_meadows_and_many_cows.MP4'
but in that way I could exceed the 255 limit of file name size (e.g. on Linux)
Any idea for a shortener algo?
Maybe I could build the shortened filename with word abbreviations?
Maybe I could remove from sentence articles, prepositions, etc.?
BTW, a minor issue: I'm working with Italian language, so a bit of chars sanitize is required to produce good filenames.
Last but not least, I'd looking for JavaScript/Node.js code
You can check if the length is larger than 255 and shorten if necessary. You should also check for duplicates and append -1, -2 and so on if necessary.
let filename='some_flowers_on_foreground_with_a_background_with_countryside_meadows_and_few_cows.MP4'
if(filename.length>255)
filename=filename.slice(0,255-4)+'.MP4'
I've recently started Coding a program that will replace sound effects from a default directory, in the Source-Engine Game, Garry's Mod.
This is the current code:
function GM:PlayerFootstep( ply, pos, foot, sound, volume, rf )
ply:EmitSound("gear1")
return true
end
I want to emit multiple .wav Sound effects, without them overlapping, and being selected at random.
I have not found any Source helpful enough on the Internet to assist, so i resorted to Stack Overflow.
I would appreciate assistance with the topic.
You'll want to look at the file.Find function for this.
I'd recommend having a custom folder such as sound/customsteps/ where you can put all your custom sounds. I would also recommend using the .wav format for the sound files, but some others do work (.mp3 and .ogg if I recall correctly).
In your code, simply call local snds=file.Find( "sound/customsteps/*", "GAME" ) which gives you a table, then you can simply choose a random one from the list using local snd=snds[math.random(1,#snds)] and play it as you do in your above code - ply:EmitSound(snd).
Make sure you create the table of sounds outside of the GM:PlayerFootstep function, so that it only runs once. I would also recommend precaching all the sounds. You can do this by looping through the table and calling util.PrecacheSound(path) on them, like so:
for k,v in pairs(snds) do
util.PrecacheSound(v)
end
So, with all that in mind - your final code should look something like this:
local snds=file.Find( "sound/customsteps/*", "GAME" )
for k,v in pairs(snds) do
util.PrecacheSound(v)
end
function GM:PlayerFootstep( ply, pos, foot, sound, volume, rf )
ply:EmitSound(snds[math.random(1,#snds)])
return true
end
Source: personal experience
I want to record the call in asterisk with the loop one minute for each audio file. That means there are many audio recording files per one call but they have the length of 1 minute. For example, the recording file names for the call in 10 minuts are: audioRec1.wav, audioRec2.wav, audioRec3.wav, audioRec4.wav ... audioRec10.wav.
Is it possible to do this in asterisk? if not, are there any program doing this job?
Thank you very much!
You have 4 options
1) Create external controller app, which will remember all record states and if callmore then minute do stopmonitor/startmonitor.
Connect via ami http://www.voip-info.org/wiki/view/Asterisk+manager+API
Stop call monitor http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+StopMonitor
Change name http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ChangeMonitor
Start new monitoring http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Monitor
2) Change asterisk source code to do what you expect. Very complex,require guru levle in asterisk.
3) Record by asterisk Record app using chanspy in a loop, always changing files
http://www.voip-info.org/wiki/view/Asterisk+cmd+Record
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
4) split file after end of file. this is very easy method.
So here is what I've got:
The problem that I face requires me to take a specialized header from WAV1 , and put it as the header for WAV2, in order to make WAV2 work with the API that I'm using. However, whenever I try to replace the first 38 characters of WAV2 with the first 38 of WAV1, I get an error when I try to play the file, I get an error saying that it is not formatted properly. Both WAV1 and WAV2 play properly before the edit.
Do you guys have any idea on what I'm doing wrong?
Thanks so much for your help.
-Rhynorater.
Wav format is a standardised format (see https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ for details about file format). I'm not sure what a "specialized" header is (perhaps you could clarify what your specialised header is?) as the format is standard - any variation would not be a wav file.
The first 38 bytes of a wav file are the header and should adhere to the standard. You cannot copy the header from one file and use it for another as the header contains information specific to the individual file (number of channels, sample rate, file length, etc).
If you both files playback normally (how are you testing this?) I'm not sure why the API you are using is not compatible (which API are you using?).