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• What is a lightweight cross platform WAV playing library?

  - by Lokkju

  I'm looking for a lightweight way to make my program (written in C) be able to play audio files on either windows or linux. I am currently using windows native calls, which is essentially just a single call that is passed a filename. I would like something similar that works on linux. The audio files are Microsoft PCM, Single channel, 22Khz Any Suggestions?

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• error in qemu monitor wavcapture with virsh

  - by Aniket Awati

  I have VM running on qemu-kvm. I am managing it with libvirt and command line tool virsh. I want to record the audio output of the VM. Here is what I am trying - virsh qemu-monitor-command -hmp VM_NAME wavcapture VM.wav This is the output I am getting : Failed to open wave file `vm.wav' Reason: Permission denied Failed to add wave capture I have tried to create a dummy vm.wav with 777 permissions. But I still get the same error.

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• Is it possible to auto update php.ini via a bash script?

  - by Tada.wav

  I'm trying to write an install script and i need to change the sendmail line in php.ini but I want to do this automatically at the moment I'm doing this manually: sudo nano /etc/php5/apache2/php.ini finding the line containing sendmail_path = then editing it to be sendmail_path = /usr/bin/msmtp -t then saving the file. Is it possible to just auto script this to make the change? Thanks a lot

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• FileUtils.mv adding linebreaks in Windows

  - by Lowgain

  I am streaming wav data from a flash application. If I get the data and do the following: f = File.open('c:/test.wav') f << wav_data.pack('c'*wav_data.length) f.close The wav file works perfectly. If I do this: f = Tempfile.new('test.wav') f << wav_data.pack('c'*wav_data.length) f.close FileUtils.mv(f.path, 'c:/') The file is there, but sounds all garbled. Checking in a hex editor shows that everywhere the working file had an 0A (or \n), the garbled version had 0D0A (or \r\n) I am using this in conjuction with rails+paperclip, and am going to be using a combination of Heroku and S3 for the live app, so I am hoping this problem will solve itself, but I'd like to get this working on my local machine for the time being. Does anybody know of any reason FileUtils.mv would be doing this, and if there is a way to change its behaviour?

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• Sound Manager Classes for Windows

  - by Yakov

  I need some classes for playing short wav sounds, this classes would load this wav files into memory when an instance created, play sounds in background when needed, release this wav files from memory when an instance disposed. How can I do this on C# for windows (.Net 2.0)? (Win API's sndPlaySound, OpenAL or may be any wrapper) Ideally I would love to find an exist solution that simple and able to solve my task. Do you know any solutions for this issue?

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• can the python wave module accept StringIO object

  - by user368005

  i'm trying to use the wave module to read wav files in python. whats not typical of my applications is that I'm NOT using a file or a filename to read the wav file, but instead i have the wav file in a buffer. And here's what i'm doing import StringIO buffer = StringIO.StringIO() buffer.output(wav_buffer) file = wave.open(buffer, 'r') but i'm getting a EOFError when i run it... File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 493, in open return Wave_read(f) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 163, in __init__ self.initfp(f) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 128, in initfp self._file = Chunk(file, bigendian = 0) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/chunk.py", line 63, in __init__ raise EOFError i know the StringIO stuff works for creation of wav file and i tried the following and it works import StringIO buffer = StringIO.StringIO() audio_out = wave.open(buffer, 'w') audio_out.setframerate(m.getRate()) audio_out.setsampwidth(2) audio_out.setcomptype('NONE', 'not compressed') audio_out.setnchannels(1) audio_out.writeframes(raw_audio) audio_out.close() buffer.flush() # these lines do not work... # buffer.output(wav_buffer) # file = wave.open(buffer, 'r') # this file plays out fine in VLC file = open(FILE_NAME + ".wav", 'w') file.write(buffer.getvalue()) file.close() buffer.close()

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• Conditional Batch file renaming with mysql data

  - by Paul Stevens

  Hello, I wonder if anyone knows how could I rename multiple files, all of them originally named with same structure, and add some data extracted from a mysql DB according to specifics rules. For example I have 500 files named with this vars: ID NAME ADDRESS PHONE.wav = 1234567 PAULSIMON WESTDR122 9942213456.wav Now I need to rename files taking some data from the databases for each file, and append the data from a query appended to the filename. For example add the data resulting from a query Where some conditions match, and the data to build the query is taked from original file name, as ID or NAME. i other words, lets say that I want to build a query taking ID & NAME from file 1234567 PAULSIMON WESTDR123 9942213456.wav as WHERE statements to take another value as BirthDATE and add this to new filename, so final result should be: ID NAME ADDRESS PHONE BIRTHDATE.wav I will appreciate any help on this. I need this to be done on a LINUX server.

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• REBOL: How to do another task while waiting an alarm to trigger ?

  - by Rebol Tutorial

  So I have created an alarm function: alarm: func[seconds message [string! unset!]][ wav: load %attention.wav sound-port: open sound:// wait seconds do [ insert sound-port wav wait sound-port close sound-port if (value? 'message) [ print message ] ] ] which works like this: alarm 30 "Will trigger in 30 seconds" Now how can I for example show a timer which increments while waiting for the alarm since Rebol doesn't support thread ?

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• Visual Studio Play Sound With No File Present

  - by jb

  It's fairly simple to play a sound file, say temp.wav, from Visual Studio C#. I'm looking for a way to do this without temp.wav being locatable from the machine running the program. IE - I want to read the .WAV file into my solution so that it is somehow inside of my .EXE and be played by it. Is this possible?

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• problem with taking file name from list and openning it?

  - by kaushik

  i have list similar to this m=[['qw','wew','23','C:/xyz/s.wav'],['qw','wew','23','C:/xyz/s2.wav'],['qw','wew','23','C:/xyz/s1.wav']] now i want to these files win=wave.open(m[0][3],'rb') it is giving error how can i use this in this way... i want to take the files name from the list please suggest???

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• For improving the join of two wave files

  - by kaki

  i want to get the values of the last 30 frames of the first wav file and first thirty frames of the second wave file in integer format and stored in a list or array. i have written the code for joining but during this manupalation i am getting in byte format and tried to convert it to integer but couldn't. as told before i want to get the frame detail of 1st 30 and last 30 in integer format,and by performing other operations join can be more successful looking for your help in this,please... thanking you, import wave m=['C:/begpython/S0001_0002.wav', 'C:/begpython/S0001_0001.wav'] i=1 a=m[i] infiles = [a] outfile = "C:/begpython/S0001_00367.wav" data= [] data1=[] for infile in infiles: w = wave.open(infile, 'rb') data1=[w.getnframes] #print w.readframes(100) data.append( [w.getparams(), w.readframes(w.getnframes())] ) #print w.readframes(1) #data1 = [ord(character) for character in data1] #print data1 #data1 = ''.join(chr(character) for character in data1) w.close() print data output = wave.open(outfile, 'wb') output.setparams(data[0][0]) output.writeframes(data[0][1]) output.writeframes(data[1][1]) output.writeframes(data[2][1]) output.close()

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• Python Threading, loading one thread after another

  - by Michael

  Hi, I'm working on a media player and am able to load in a single .wav and play it. As seen in the code below. foo = wx.FileDialog(self, message="Open a .wav file...", defaultDir=os.getcwd(), defaultFile="", style=wx.FD_MULTIPLE) foo.ShowModal() queue = foo.GetPaths() self.playing_thread = threading.Thread(target=self.playFile, args=(queue[0], 'msg')) self.playing_thread.start() But the problem is, when I try to make the above code into a loop for multiple .wav files. Such that while playing_thread.isActive == True, create and .start() the thread. Then if .isActive == False, pop queue[0] and load the next .wav file. Problem is, my UI will lock up and I'll have to terminate the program. Any ideas would be appreciated.

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• Audio processing libraries for Ruby?

  - by J. Pablo Fernández

  Any recommendation on libraries to do audio processing in Ruby. I need to do the following two tasks: Find silences, for which I'm happy to just be able to iterate over each sample in the wave. Cut and paste pieces of wav files to form a new wav file. Convert wav to mp3, which I will probably leave to lame anyway. I'm looking for the equivalent of NAudio, a C# library.

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• play sound clips in iphone

  - by LaGrad

  I'm new to iphone development. i want to write an app that has 3 buttons. touching each will trigger a sound. 1.wav 2.wav 3.wav Can someone give me some hints and guide me to the right path?

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• [c#] SoundPlayer.PlaySync stopping prematurely

  - by JeffE

  I want to play a wav file synchronously on the gui thread, but my call to PlaySync is returning early (and prematurely stopping playback). The wav file is 2-3 minutes. Here's what my code looks like: //in gui code (event handler) //play first audio file JE_SP.playSound("example1.wav"); //do a few other statements doSomethingUnrelated(); //play another audio file JE_SP.playSound("example2.wav"); //library method written by me, called in gui code, but located in another assembly public static int playSound(string wavFile, bool synchronous = true, bool debug = true, string logFile = "", int loadTimeout = FIVE_MINUTES_IN_MS) { SoundPlayer sp = new SoundPlayer(); sp.LoadTimeout = loadTimeout; sp.SoundLocation = wavFile; sp.Load(); switch (synchronous) { case true: sp.PlaySync(); break; case false: sp.Play(); break; } if (debug) { string writeMe = "JE_SP: \r\n\tSoundLocation = " + sp.SoundLocation + "\r\n\t" + "Synchronous = " + synchronous.ToString(); JE_Log.logMessage(writeMe); } sp.Dispose(); sp = null; return 0; } Some things I've thought of are the load timeout, and playing the audio on another thread and then manually 'freeze' the gui by forcing the gui thread to wait for the duration of the sound file. I tried lengthening the load timeout, but that did nothing. I'm not quite sure what the best way to get the duration of a wav file is without using code written by somebody who isn't me/Microsoft. I suppose this can be calculated since I know the file size, and all of the encoding properties (bitrate, sample rate, sample size, etc) are consistent across all files I intend to play. Can somebody elaborate on how to calculate the duration of a wav file using this info? That is, if nobody has an idea about why PlaySync is returning early. Of Note: I encountered a similar problem in VB 6 a while ago, but that was caused by a timeout, which I don't suspect to be a problem here. Shorter (< 1min) files seem to play fine, so I might decide to manually edit the longer files down, then play them separately with multiple calls.

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• ffmpeg hangs when creating a video

  - by FearUs

  I am trying to insert an audio channel with a video: first of all I extract the audio from the original video for processing: ffmpeg -i lotr.mp4 lotr.wav I then extract all frames for later processing too: ffmpeg -i lotr.mp4 -f image2 %d.jpg When done processing audio and video streams, I try to create the video ffmpeg -f image2 -r 15 -i %d.jpg new.mp4 then merge with the audio: ffmpeg -i new.mp4 -i lotr.wav -map 0:0 -map 1:0 new_w_audio.mp4 Result: CPU activity = 100%, the process hangs and never returns. PS: I even tried it without modifying the images or the audio (so just trying to unpack then repack the video) but still the same output FFmpeg version SVN-r26400, Copyright (c) 2000-2011 the FFmpeg developers built on Jan 18 2011 04:07:05 with gcc 4.4.2 configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect - -enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads -- cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack libavutil 50.36. 0 / 50.36. 0 libavcore 0.16. 1 / 0.16. 1 libavcodec 52.108. 0 / 52.108. 0 libavformat 52.93. 0 / 52.93. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1.74. 0 / 1.74. 0 libswscale 0.12. 0 / 0.12. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'new.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Duration: 00:00:29.66, start: 0.000000, bitrate: 193 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], 192 k b/s, 15 fps, 15 tbr, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 [wav @ 01fed010] max_analyze_duration reached Input #1, wav, from 'lotr.wav': Duration: 00:00:29.90, bitrate: 176 kb/s Stream #1.0: Audio: pcm_s16le, 11025 Hz, 1 channels, s16, 176 kb/s File 'new_w_audio.mp4' already exists. Overwrite ? [y/N] y [buffer @ 01b03820] w:200 h:134 pixfmt:yuv420p Output #0, mp4, to 'new_w_audio.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf52.93.0 Stream #0.0(und): Video: mpeg4, yuv420p, 200x134 [PAR 1:1 DAR 100:67], q=2-3 1, 200 kb/s, 15 tbn, 15 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding

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• How to record both audio, Where i have one music running and my microphone is in use?

  - by YumYumYum

  I have one music playing, and i have microphone open, already the microphone is used by other application. In such case, how can i record that music and the microphone audio to a file? (if possible with command line). Follow up: $ rec new-file.wav Input File : 'default' (alsa) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:25.94 [00:00:00.00] Out:1.24M [ | ] Clip:0 ^C $ sox -d new-file.wav

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• Multiple Control Templates for a custom control in Silverlight

  - by Tada

  I am creating a custom control. The contents of the control will differ a lot when in different visual states. Can I to achieve the above, apply different control templates to the same custom control? That is define more than one control template for a custom control? If not, any clues as to how I can do this, without have as many custom/user controls as there are states?

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• Convert MP3 to AAC,FLAC to AAC (.NET/C#) FREE :)

  - by PearlFactory

  So I was tasked with looking at converting 10 million tracks from mp3 320k to AAC and also Converting from mp3 320k to mp3 128k After a bit of hunting around the tool you need to use is FFMPEG Download x64 WindowsAlso for the best results get the Nero AACEncoder Download Now the command line STEP 1(From Flac)ffmpeg -i input.flac -f wav - | neroAacEnc -ignorelength -q 0.5 -if - -of output.m4aor (From mp3)ffmpeg -i input.mp3 -f wav - | neroAacEnc -ignorelength -q 0.5 -if - -of output.m4aNow the output.m4a is a intermediate state that we now put a ACC wrapper on via FFMpeg STEP 2ffmpeg -i output.m4a -vn -acodec copy final.aacDone :) There are a couple of options with the FFMPEG library as in we can look at importing the librarys and manipulation the API for the direct result FFMPEG has this support. You can get the relevant librarys from HereThey even have the source if you are that keen :-)In this case I am going to wrap the command lines into c# external process threads.( For the app that i am building to convert the 10 million tracks there is a complex multithreaded app to support this novel code )//Arrange Metadata about Call Process myProcess = new Process();ProcessStartInfo p = new ProcessStartInfo();string sArgs = string.format(" -i {0} -f wav - | neroAacEnc -ignorelength -q 0.5 -if - -of {1}",inputfile,outputfil) ; p.FileName = "ffmpeg.exe" ; p.CreateNoWindow = true; p.RedirectStandardOutput = true; //p.WindowStyle = ProcessWindowStyle.Normal p.UseShellExecute = false;//Execute p.Arguments = sArgs; myProcess.StartInfo = p; myProcess.Start(); myProcess.WaitForExit();//Write details about call  myProcess.StandardOutput.ReadToEnd();Now in this case we would execute a 2nd call using the same code but with different sArgs to put the AAC wrapper on the m4a file. Thats it. So if you need to do some conversions of any kind for you ASP.net sites/apps this is a great start and super fast.. With conversion times of around 2-3 seconds all of this can be done on the fly:-)Justin Oehlmannref : StackOverflow.com

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• Acceptable sound quality: stereo needed for an Android game?

  - by Thomas Calc

  I have various simple short sound effects (damage sound, dying sound, thunderbolt, fanfare, breaking) for a game that is developed for Android currently. I use OGG files: 96kbps VBR, 44.1KHz, 2 channels (that means stereo, right?). I read the other stackexchange topics about "acceptable sound quality", but they're too general, address too many things. My experience is that even with 80kbps, my effects sound OK. But I tested it on a limited number of Android devices (including a Sony Ericsson Xperia Neo and a HTC Desire HD). My questions: For mobile phones and tablets, generally, what parameters are recommended? Won't my 80kbps sounds be bad on a newer device (such as a modern tablet)? I don't hear any difference between stereo and mono (2 channels vs. 1 channel, right?), is there any noticeable difference at all for mobile phones / tablets? (in terms of the player experience) May it worth it at all? I assume that stereo sounds take much more in memory (when they're decoded to PCM), despite of the fact that the compressed OGG size is practically the same. Reacting to Roy T.'s great comment: Actually, I couldn't measure the PCM size (Android decodes OGG internally), but I thought that stereo will take more space than mono when uncompressed After throwing out one of the WAV channels in Audacity, and re-exporting it: The new WAV file size is half than before The OGG file size is practically the same as before The sound effects and game music was recorded by my friend who is an experienced hobby musician/composer, but he knows little about computers & software so he just gave me some high-quality WAV files generated via his hardware.These were stereo, but if I check them in Audacity, both channels appear to be exactly the same.Can I consider them the same (= moving to mono), or might there be some unnoticeable differences to the human eye?

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• Sound not playing on Windows XP - SoundEffect or Song: Monogame

  - by ashes999

  I'm trying to integrate sound into my Monogame game. I don't have the content pipeline hack -- just straight Monogame (Beta 3) at this point. (I tried adding the content pipeline, but ran into some issues.) I added a .wav file to my /Content directory, and I can create and instantiate both SoundEffect and Song classes. However, both show durations of 00:00:00 (on a ten-second long file), and neither plays. I can call LoadContent without any issue. But when I call Play, nothing plays. I've tried a couple of different sounds, and different formats (MP3 and WAV) to rule that out. Only WAV seems to even load without crashing out, but it doesn't play. There seems to be a GitHub issue that fixes this problem in 2.5.1. Downgrading to 2.5.1 doesn't fix this problem; it seems like it's fixed in 3.0 (_data is set in the SoundEffect instance). This issue only occurs on Windows XP. I tested it on a Windows 7 laptop, and the sound plays fine.

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• hdmi audio works only with aplay -D alsa test wavs; open source radeon drivers; kernel 3.5 vgaswitcheroo

  - by user108754

  I've trolled the internets to make hdmi work on my system Ubuntu 12.04 software center kernel 3.5 uname: Linux ubuntu 3.5.0-18-generic #29~precise1-Ubuntu SMP...x86_64 x86_64 x86_64 GNU/Linux open source radeon drivers vgaswitcheroo (hybrid intel/radeon gpu): I boot with intel, not radeon, running. (and recall that with kernel 3.5, vgaswitcheroo now gives info on a third item, "DIS-Audio"; it indicates pwr on my system) ( /etc/rc.local: chown user:user /sys/kernel/debug/ # change "username" with your user name echo OFF /sys/kernel/debug/vgaswitcheroo/switch ) grub indeed now has "radeon.audio=1" for testing audio, I did aplay -l which gave me the card and device, which made me try aplay -D plughw:1,3 /usr/share/sounds/alsa/Front_Center.wav and lo! I get crystal clear sound on my hdtv. If I play an mp3 file as the argument to that command, I get noise as, I guess, aplay interprets the mp3 code as a wav. If I play a .wav that is not in the /usr/share/sounds/alsa/ directory, I get nothing. Internet flash video in browser plays no sound over hdmi. Both system sounds control and pavucontrol have hdmi cedar selected. Alas, I can not get sound for any gui test (left, right). Why would only aplay, and only when directed with "-D plughw", yield sound over hdmi? I've also tried only using one sound program at a time, if it was a limitation of alsa, so I tried aplay with web browser and even the sound control gui closed. I tried each of the last two, running alone. No improvement. alsamixer only shows hda intel and I think it's only the intel audio, not the hdmi.

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• After upgrading to trusty, ALSA midi connection (aconnect) doesn't seem to work right

  - by SougonNaTakumi

  Previously in kubuntu 13.10 I was able to open vmpk or plug in a midi keyboard, and provided that TiMidity was running in server mode, I could run aconnect [keyboard port (129:0 for vmpk)] 14:0 aconnect 14:0 128:0 and I could play the keyboard and get sound. But now, a while after upgrading to trusty, I tried to do that, and didn't get any sound. TiMidity itself still plays files fine, but if I try to play them with aplaymidi, I still just get silence. Oddly, the midi files are clearly being read. When I ran (where 130:0 was vmpk's input port) aplaymidi -p 130:0 ~/path/to/midi.mid vmpk was highlighting notes on the piano as if it were playing the midi. One time I tried this, TiMidity (?) very briefly played a fraction of a second of the first chord of my song before everything went silent and vmpk just highlighted the first voice on the keyboard as usual. Now the weirdest part of this is that probably about 40% of the time, when I've played at least one note with either aplaymidi or vmpk, when I run aconnect -x I get a sudden burst of a note or chord from my speakers (that is, if I played one note, I get a note; if I played multiple sequential notes, they turn into a chord), as if the notes were being queued up but not being played and that somehow liberated them. I have no idea what's going on there. A little while ago I remember having a problem with Audacity playing wav files sped up and also locking up if I tried to pause it, which it stopped doing when I set the audio devices to the actual audio devices rather than pulse. But now when I checked again, it's doing the opposite: it won't play audio at all and/or acts weirdly if I don't set the audio devices to pulse, and either way will very occasionally randomly do the speeding up thing regardless. Oddly in the midst of what's looking like a pretty screwed up sound system, sound in VLC and Firefox has been working fine and if I play a wav file with aplay ~/path/to/sound.wav that works fine too. Any idea what I could do to figure out what's wrong with ALSA and/or fix it?

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• mix audio with h264 mp4 video with ffmpeg

  - by user2362912

  I have 2 files : Input #0, wav, from '105426_1.wav': Duration: 00:00:09.98, bitrate: 1312 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 41000 Hz, stereo, s16, 1312 kb/s and: Duration: 00:00:41.29, start: 0.000000, bitrate: 1313 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 1211 kb/s, 24.42 fps, 25 tbr, 90k tbn, 48 tbc Metadata: handler_name : VideoHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 99 kb/s Metadata: handler_name : SoundHandler I want to insert first audio file into video in special place (for example in 10 secunde of video) and mix it with audio stream of video file. I try to /usr/local/bin/ffmpeg -i 105426_1.wav -i 105426.mp4 -map 0:0 -map 1:1 -map 1:0 video_finale.mp4 but result is : Duration: 00:00:41.31, start: 0.046440, bitrate: 755 kb/s Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s Metadata: handler_name : SoundHandler Stream #0:2(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 588 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc Metadata: handler_name : VideoHandler I need only one audio stream and first stream play not from beginig but from 10 sec

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• SoX on Windows 7 64-Bit Outfile Missing

  - by Christian

  I have come across the strangest problem when trying to run sox.exe on my Windows 7 installation. Whenever I try and record audio it works without any issues but it will not output an audio file. The crazy thing is that when I use the play command it successfully plays what I just recorded. Has anyone ever heard of this happening? Here are the commands (and output) that I'm using: C:\Program Files (x86)\Vox\sox-14-4-0>sox -d test.wav trim 0 00:05 Input File : 'default' (waveaudio) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:05.03 [00:00:00.00] Out:240k [ | ] Clip:0 Done. C:\Program Files (x86)\Vox\sox-14-4-0>play test.wav test.wav: File Size: 960k Bit Rate: 1.54M Encoding: Signed PCM Channels: 2 @ 16-bit Samplerate: 48000Hz Replaygain: off Duration: 00:00:05.00 In:100% 00:00:05.00 [00:00:00.00] Out:240k [ | ] Clip:0 Done. Am I losing my mind or is something up here?

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