How can I do the following?
Sound object is selected
save sound object as wav (as a temporary filename)
run sox -i adsda.wav or soxi adsda.wav
output to the display the result of sox -i
delete temporary file
You can execute sox by calling the system directive, and then capture the output in a temporary file. You can then read this temporary file into Praat with readFile$(). Here's an example:
infile$ = "sound.wav"
outfile$ = "soxi.out"
sound = Create Sound as pure tone: "tone",
... 1, 0, 0.4, 44100, 440, 0.2, 0.01, 0.01
Save as WAV file: infile$
system soxi 'infile$' > 'outfile$'
soxi$ = readFile$(outfile$)
appendInfo: soxi$
deleteFile: infile$
deleteFile: outfile$
removeObject: sound
I'm not sure what changes (if any) this would need in Windows, but considering you are using sox, that probably is not a problem.
Please note that currently (Praat v.5.4) the system directive still uses the old syntax, which uses variable substitution. This will probably change in the future. But this script should remain usable even then.
Related
I have a long list of audio files, and some of them are longer than an hour. I am using Python 3.6, Jupyter notebook by connecting to a remote machine and using TinyTag library to get a duration of audio. Ffmpeg version is 2.8.14-0ubuntu0.16.04.1.
My code below goes over the files and if a file is longer than an hour, it splits the file into one-hour long pieces, and a leftover piece less than an hour, and copies the pieces as fname_0, fname_1,fname_2, etc. Before chopped, each file is .m4a but during chopping, they are converted to a .wav file. However, after this chopping process, when reading the duration of pieces, I realized that all the pieces have 'None' duration. Something must be wrong in the command line but I can`t see what that is. Thanks in advance.
# fpaths is the list of filepaths
for i in range(0,len(fpaths)):
fpath=fpaths[i]
fname=os.path.basename(fpath)
fname0=os.path.splitext(fname)[0] #name without extension
tag = TinyTag.get(fname)
if tag.duration > 3600:
cmd2 = "ffmpeg -i %s -f segment -segment_time 3600 -c copy %s" %(fpath, fname0) + "_%d.wav"
os.system(cmd2)
os.remove(fpath)
When I change to the extension from .wav to .m4a in the cmd2 command line, it works. Writing here just in case if someone has the same problem.
I've got a large (5Gb) WAV 64 file that I'm trying to analyse with SoX. On doing;
sox /file/ -n stats
it tells me;
sox FAIL formats: can't open input file '/file/': WAVE: RIFF header not found
On attempting to specify;
sox /file/ -t w64 -n stats
it says;
sox WARN sox: ignoring `-t w64'.
and proceeds to throw the same error as before. This file is W64. How do I convince SoX to accept it?
Looks like ffmpeg can handle wav64 files
Have you tried to create your own tiny wav64 file and see if you can read that ?
Looks like wav64 is targeted for multichannel audio is that your use case ? WAV format itself is very simple and lends itself to allowing people to write two pages of code to read any wav file ... if you exhaust all other options and your life depended on getting your file read, reading the ISO spec or similar on wav64 format then writing your own reader is an option.
Can you share how your wav64 file was created ? Maybe one of us can replicate its creation and battle how to read it back
If its not full of massive multi-channel audio is getting it converted to normal wav an option ?
Maybe, your wave64 file is invalid. A valid one should contain a riff header.
You've added -t w64 flag after the filename. That's why it's ignored.
# This would work! [CHECKED]
# If your file is valid!
sox -t w64 in_file -n stats
I currently have a list of file for which I need to change the sample rate for.
I'recently been aware that this is possible using sox But when I try do it, I keep on getting a error message that sox wav: Premature EOF on .wav input file And causes the audio file to be empty.. it seems like that sox is not able to resample an audio file which input = output... which I kinda need, if I have to convert a whole directory of audio files...
Currently used commands:
~/kaldi-trunk/egs/yesno/s5_k_added$ sox 0_0_0_0_1_1_1_1.wav -r 8000 0_0_0_0_1_1_1_1.wav
sox WARN wav: Premature EOF on .wav input file
:~/kaldi-trunk/egs/yesno/s5_k_added$ play 0_0_0_0_1_1_1_1.wav
0_0_0_0_1_1_1_1.wav:
File Size: 44
Encoding: Signed PCM
Channels: 1 # 16-bit
Samplerate: 8000Hz
Replaygain: off
Duration: unknown
In:0.00% 00:00:00.00 [00:00:00.00] Out:0 [ | ] Clip:0
Done.
How do I resample a directory of audio files?
Try changing the output file name, possible putting it in a different directory if you want to keep the same file name.
For example:
sox 0_0_0_0_1_1_1_1.wav -r 8000 ./out/0_0_0_0_1_1_1_1.wav
I believe sox is attempting to read the file while it is actively changing it, sometimes sox does not create a temporary file to output into.
EDIT: If you have a directory of files you all want to change, use this:
$ mkdir out
$ for file in *; do sox ${file} -r 8000 ./out/${file}; done
I am attempting to convert a directory full of .bmp files into a .mp4 file (or similar format).
The bitmap files have the following name scheme:
output_N_1024.bmp
Where N is an integer in the range 0 to 1023. (No zero padding / fixed width.)
The command I am using is:
avconv -r 25 -i output_{0..1023}_1024.bmp outputfile.mp4
This appears to run okay, and takes about a minute to convert all 1024, 1024 by 1024 resolution - (confusing?) bitmap images into a new file, outputfile.mp4.
However, when I attempt to open this file with VLC, a black window briefly flashes up and then closes. VLC then goes back to its mode where it waits for you to tell it which file to open next. No error or warning messages appear from VLC, which seems kind of strange since it seems to be refusing to play.
What can I do to fix this? Perhaps my converting command is incorrect?
The problem most likely is that you haven't actually passed the command to encode these files to avconv. This has happened because your shell has expanded the filenames already.
The command i have just managed to get to work on my machine is:
avconv -r 2 -i "%d.bmp" -s 600x400 -an out.ogv
Also for whatever reason it didn't want to work without explicitely giving it the size, but i don't think this is your problem.
In here quotes tell your shell not to touch this string. %d means digits from 1 to whatever the last file is (if you would want them to be 0-padded this would look like %000d to have maximum of three naughts in front).
VLC has then opened and ran my file just fine.
I am trying to convert midi files to ogg or mp3. Eventually this will happen on a linux webserver but currently I am using a Windows 7 machine. I am using timidity to convert the midi to wav and then either sox or ffmpeg to convert the wav to ogg/mp3.
When I use an intermediate file the process works fine (in the first line below timidity creates file.wav)
timidity.exe file.mid -Ow
sox.exe file.wav file.ogg
However, when I try to pipe the timidity output into sox (as below), the resulting file ogg is horribly distorted
timidity.exe file.mid -Ow -o - | sox.exe -t wav - file.ogg
and I get a warning
sox.exe WARN wav: Premature EOF on .wav input file
I also get the same distortion problem when I replace sox with ffmpeg (and the appropriate command line options), or when I replace ogg with mp3 as the output format.
So what am I doing wrong?
Thanks,
Chris
Regarding the warning itself, you're doing nothing wrong. You may also see a warning from timidity that reads something like
Warning: -: Illegal seek: Can't make valid header
What's happening there is explained in the timidity manual page:
If output is directed to a non-seekable file, or if TiMidity++ is interrupted before closing the file, the file header will contain 0xffffffff in the RIFF and data block length fields.
Note that RIFF is the encoding format commonly called by its file extension, .wav. When timidity writes a RIFF file, it doesn't know how long the file will be, so it writes some placeholder junk in the header and moves on to writing the data. When it finishes with the data, it knows how long the file is, so it goes back to the beginning of the file and writes over that junk in the header. When you write to a pipe, it has no way to go back and rewrite anything: the downstream program has to handle the placeholder junk. Also from the timidity manual page:
The popular sound conversion utility sox is able to read such malformed files, so you can pipe data directly to sox for on-the-fly conversion to other formats.
Thus, the message you mentioned. Sox is informing you that the chef prepared the file wrong BUT SOX IS HAPPY TO EAT IT ANYWAY BECAUSE SOX IS NOT PICKY. Sox is apparently passive-aggressive. Who knew?
You can ignore those warning messages, because now they are telling you something you already know. Or, you can use a raw format and explicitly tell timidity and sox how to play well with one another:
timidity file.midi -Or1Ssl -s44.1 -o- | sox -t raw -b 16 -e signed -r 44.1k -c 2 - file.ogg
As for the distortion, that may be caused in part by quirks in the audio libraries on the Windows system. I note that the pipeline in the question, sans .exe extensions, produces output with no notable distortion on a linux system. Using a well-defined raw format in the pipeline may also help with that issue.
Note that for Ogg output, you can now get that directly from timidity:
timidity file.midi -o file.ogg -Ov