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Post by bluatigro on Mar 30, 2020 3:12:06 GMT -5
error :
i can not find the wav file
i want to write a composer program
first i need to create 1 note
the code gives a short sound but i do not think that is myne note A
'' bluatigro 30 mrt 2020
'' wav creator try
''create wav struct
struct wav _
, chunkid as long _ ''RIFF
, chunksize as long _
, format as long _
, subchunk1id as long _
, subchunk1size as long _
, audioformat as short _ '' 1
, numchannels as short _ '' 1 = mono
, samplerate as long _ ''8000 41000
, byterate as long _
, blockalign as short _
, bitpersample as short _ ''8 16
, subchunk2id as long _
, subchunk2size as long
global numsamples ''1 sec
numsamples = 41000
global pi
pi = atn(1)*4
dim wavdata( numsamples )
wav.chunkid.struct = hexdec( "52494646" )''RIFF
wav.format.struct = hexdec( "57415645" )''WAVE
wav.subchunk1id.struct = hexdec( "666d7420" )''fmt
wav.subchunk1size.struct = 16
wav.audioformat.struct = 1
wav.numchannels.struct = 1 ''1 = mono 2 = stereo
wav.samplerate.struct = numsamples
wav.bitpersample.struct = 16
wav.blockalign.struct = wav.numchannels.struct _
* wav.bitpersample.struct / 8
wav.byterate.struct = numsamples _
* wav.numchannels.struct * wav.bitpersample.struct / 8
wav.subchunk2id = hexdec( "64617461" )''data
wav.subchunk2size = numsamples _
* wav.numchannels.struct * wav.bitpersample.struct / 8
wav.chunksize.struct = _
4 + ( 8 + wav.subchunk1size.struct ) _
+ ( 8 + wav.subchunk2size.struct )
for i = 0 to numsamples
'' note A
wavdata( i ) = int( sin( 440 * pi * 2 / wav.samplerate.struct ) _
* 256 * 127 + 256 * 128 )
next i
open "note_440_1000.wav" for output as #uit
print #uit , wav$()
close #uit
input "[ wav ready : push return ]" ; in$
playwave "note_440_1000.wav" , sync
print "[ game over ]"
end
function wav$()
uit$ = long$( wav.chunkid.struct ) _ ''RIFF
+ long$( wav.chunksize.struct ) _
+ long$( wav.format.struct ) _
+ long$( wav.subchunk1id.struct ) _
+ long$( wav.subchunk1size.struct ) _
+ short$( wav.audioformat.struct ) _ '' 1
+ short$( wav.numchannels.struct ) _ '' 1 = mono
+ long$( wav.samplerate.struct ) _ ''8000 41000
+ long$( wav.byterate.struct ) _
+ short$( wav.blockalign.struct ) _
+ short$( wav.bitpersample.struct ) _ ''8 16
+ long$( wav.subchunk2id.struct ) _
+ long$( wav.subchunk2size.struct )
for i = 0 to numsamples
uit$ = uit$ + short$( wavdata(i) )
next i
wav$ = uit$
end function
function long$( x )
a = int( x / 256 ^ 3 ) and 255
b = int( x / 256 ^ 2 ) and 255
c = int( x / 256 ) and 255
d = int( x ) and 255
long$ = chr$( a ) + chr$( b ) + chr$( c ) + chr$( d )
end function
function short$( x )
c = int( x / 256 ) and 255
d = int( x ) and 255
short$ = chr$( c ) + chr$( d )
end function
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Post by Rod on Mar 30, 2020 3:42:04 GMT -5
Well the header info has to be wrong and in a number of places. Here is your .wav and a normal .wav to compare.
A lot of your header values cannot be correct. 256 channels? audio format? byterate? and others. I am sure you will be able to find an example .wav of the correct format you are trying to create. Read that header and compare. Also way overkill on the bit rate. You can produce perfectly pleasing .wav at much lower bit rates.
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Post by bluatigro on Mar 30, 2020 5:20:37 GMT -5
i have the header from WWW
i gooled "wav file format"
and translated the results
the samplerate can be only 8000 or 41000 acording to the site i found
i fill numhanels whit 1 how does it change ?
update :
changed samplerate to 8000
'' bluatigro 30 mrt 2020
'' wav creator try JB version
'' WAV header
global chunkid ''RIFF
global chunksize
global format
global subchunk1id
global subchunk1size
global audioformat '' 1
global numchannels '' 1 = mono
global samplerate ''8000 41000
global byterate
global blockalign
global bitpersample ''8 16
global subchunk2id
global subchunk2size
'' WAV data
global numsamples ''1 sec
numsamples = 8000
global pi
pi = atn(1)*4
dim wavdata( numsamples )
chunkid = hexdec( "52494646" )''RIFF
format = hexdec( "57415645" )''WAVE
subchunk1id = hexdec( "666d7420" )''fmt
subchunk1size = 16
audioformat = 1
numchannels = 1 ''1 = mono 2 = stereo
samplerate = 8000
bitpersample = 16
blockalign = numchannels * bitpersample / 8
byterate = numsamples * numchannels * bitpersample / 8
subchunk2id = hexdec( "64617461" )''data
subchunk2size = numsamples * numchannels * bitpersample / 8
chunksize = _
4 + ( 8 + subchunk1size ) _
+ ( 8 + subchunk2size )
for i = 0 to numsamples
'' note A 1 sec
wavdata( i ) = int( sin( 440 * pi * 2 / samplerate ) _
* 256 * 127 + 256 * 128 )
next i
open "note_440_1000.wav" for output as #uit
print #uit , wav$()
close #uit
input "[ wav ready : push return ]" ; in$
playwave "note_440_1000.wav" , sync
print "[ game over ]"
end
function wav$()
uit$ = long$( chunkid ) _ ''RIFF
+ long$( chunksize ) _
+ long$( format ) _
+ long$( subchunk1id ) _
+ long$( subchunk1size ) _
+ short$( audioformat ) _ '' 1
+ short$( numchannels ) _ '' 1 = mono
+ long$( samplerate ) _ ''8000 41000
+ long$( byterate ) _
+ short$( lockalign ) _
+ short$( bitpersample ) _ ''8 16
+ long$( subchunk2id ) _
+ long$( subchunk2size )
for i = 0 to numsamples
uit$ = uit$ + short$( wavdata(i) )
next i
wav$ = uit$
end function
function long$( x )
a = int( x / 256 ^ 3 ) and 255
b = int( x / 256 ^ 2 ) and 255
c = int( x / 256 ) and 255
d = int( x ) and 255
long$ = chr$( a ) + chr$( b ) + chr$( c ) + chr$( d )
end function
function short$( x )
c = int( x / 256 ) and 255
d = int( x ) and 255
short$ = chr$( c ) + chr$( d )
end function
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Post by Rod on Mar 30, 2020 7:03:55 GMT -5
The sample rate can be a variety of values not just 8000 or 41000 You should also be sure your sound system can cope with 41000. The values are still way off. You need to go through the header step by step which is why you should find an example .wav and analyse its header. soundfile.sapp.org/doc/WaveFormat/Your header ChunkID =RIFF ChunkSize =2755526656 Format =WAVE Sub1ID =fmt Sub1Size =268435456 AudioFormat =256 NumChannels =256 SampleRate =1075773440 ByteRate =2151546880 BlockAlign =0 bitPerSample=4096 Sub2ID =data Sub2Size =2151546880 Example header at 44100 sample rate ChunkID =RIFF ChunkSize =414240 Format =WAVE Sub1ID =fmt Sub1Size =16 AudioFormat =1 NumChannels =2 SampleRate =44100 ByteRate =176400 BlockAlign =4 bitPerSample=16 Sub2ID =data Sub2Size =413440 sub1size should be 16 audioformat should be 1 numb channels should be 1, or 2 if you are writing stereo data sample rate should be 44100 the rest is way of as a consequence of these earlier errors. |
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Post by honkytonk on Mar 31, 2020 4:42:31 GMT -5
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Post by bluatigro on Apr 1, 2020 3:32:58 GMT -5
i never leanned that language
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Post by metro on Apr 1, 2020 3:58:33 GMT -5
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Post by honkytonk on Apr 1, 2020 11:47:24 GMT -5
i never leanned that language Firefox has an addon that translates with a single click on "T" and it's very nice. |
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Post by beta1973 on Feb 11, 2022 10:05:35 GMT -5
Hi,
Has anyone developed code to READ or WRITE a WAV file? I'm not talking about playing the file, but extracting the data into an array so it can be altered and then rewritten back to another WAV file. I am able to do this with the following Matlab code:
[Ywav,Fswav,Bitswav] = wavread(FILEX);
% Ywav = audio signal from WAV file
% Fswav = sampling frequency of WAV file
% Bitswav = bits/sample of WAV file
As you can see, Matlab returns the sampling frequency, bits/sample, and a time series array containing the audio. I am then able to resample at different rates, change the bits/sample, etc. I am doing this to create ADPCM files of various sample rates, bits/sample and quantization levels. It works fine in Matlab, BUT I want to make an executable available online and I don't have a Matlab compiler. I do have Liberty Basic and because it has a PLAYWAV function it seems there should be a way to create READWAV and WRITEWAV functions as well. Suggestions, comments, solutions?
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Post by Rod on Feb 11, 2022 11:38:56 GMT -5
I would start a new thread of your own. A .wav file is just a file and it’s contents are fairly well understood.
Reading the header is the first job to understand what .wav you have. Then it’s just a case of writing the new header then transposing the data.
So simple .wav files are doable but the processing time may be long.
The complications that will soon present are that most audio files are no longer .wav file format, mpg etc
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Post by tenochtitlanuk on Feb 11, 2022 16:40:24 GMT -5
You'll find various things that might help you on my website- eg www.diga.me.uk/makeMusic.htmlwww.diga.me.uk/fourier.htmlwww.diga.me.uk/ExplainFourier.htmlwww.diga.me.uk/2DfourierFFT.html...so all very do-able. Read the whole file into a single string, and parse out the sampling rate, bytes/sample, and address the data directly from the string. No need to move into an array. Manipulate and write to a new string. Save... Sampling to lower rate would be easy. To higher rates, harder. To save, write the header modified appropriately and then append. But might be easier to convert formats with something like 'Audacity' ( I use LB/Linux) |
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Post by beta1973 on Mar 4, 2022 10:33:46 GMT -5
Thanks tenochtitlanuk. Now that I am back from holiday I will investigate your web site.
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