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  • sox mix nested scripts on amazon ec2 rhel linux

    - by Ray
    I'm trying to use "sox" to mix some audio files. The command works great on my Mac terminal sox -m audio.wav "| sox upload.wav -p trim 2 1 pad 6" final.wav This mixes (not concatenate) audio.wav and a section of upload.wav from the 2nd second to the 3rd second and adds 6 seconds of padding in the front, and outputs to final.wav Now the problem is, the SAME EXACT command does NOT work my Amazon EC2 RHEL box. (sox is installed correctly). I get the following error sox soxio: Can't open input file `| sox upload.wav -p trim 2 2 pad 6': No such file or directory For some reason RedHat doesn't like the double quotes. Even though it is documented to be used this way. Thanks for your help!

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  • How would i down-sample a .wav file then reconstruct it using nyquist? - in MATLAB

    - by Andrew
    This is all done in MATLAB 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the sampling at verious frequencies?

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  • Why do I get this error "EMCIDeviceError" when opening some wav files in my program.

    - by Roy
    Hey I have this program that has been working fine until I tried to open this one wav file? Not sure what the problem is or that I understand it? Do I need to find a new component to use for this file or what? I am using Delphi 4 Pro and the standard VCL component for Media Player. I am looking for a good new component that offers more help with wav and mp3 files too but not found what I am looking for yet?

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  • How to play non buffered .wav with MediaStreamSource implementation in Silverlight 4?

    - by kyrisu
    Background I'm trying to stream a wave file in Silverlight 4 using MediaStreamSource implementation found here. The problem is I want to play the file while it's still buffering, or at least give user some visual feedback while it's buffering. For now my code looks like that: private void button1_Click(object sender, RoutedEventArgs e) { HttpWebRequest request = (HttpWebRequest)HttpWebRequest.Create(new Uri(App.Current.Host.Source, "../test.wav")); //request.ContentType = "audio/x-wav"; request.AllowReadStreamBuffering = false; request.BeginGetResponse(new AsyncCallback(RequestCallback), request); } private void RequestCallback(IAsyncResult ar) { this.Dispatcher.BeginInvoke(delegate() { HttpWebRequest request = (HttpWebRequest)ar.AsyncState; HttpWebResponse response = (HttpWebResponse)request.EndGetResponse(ar); WaveMediaStreamSource wavMss = new WaveMediaStreamSource(response.GetResponseStream()); try { me.SetSource(wavMss); } catch (InvalidOperationException) { // This file is not valid } me.Play(); }); } The problem is that after settings request.AllowREadStreamBuffer to false the stream does not support seeking and the above mentioned implementation throws an exception (keep in mind I've put some of the position setting logic into if(stream.CanSeek) block): Read is not supported on the main thread when buffering is disabled Question Is there a way to play wav stream without buffering it in advance?

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  • how to record mic input and pipe the output to another program

    - by acrs
    Hi everyone Im trying to follow a tutorial on generating truly random bits How To Generate Truly Random Bits This is the command from the tutorial but it does not work rec -c 1 -d /dev/dsp -r 8000 -t wav -s w - | ./noise-filter >bits I know i can record my mic input using rec -c 1 no.wav this is the command i tried using rec -c 1 -r 8000 -t wav -s noise.wav | ./noise-filter >bits but i get root@xxc:~/cc# rec -c 1 -r 8000 -t wav -s noise.wav - | ./noise-filter >bits rec WARN formats: can't set sample rate 8000; using 48000 rec FAIL sox: Input files must have the same sample-rate I have complied noise-filter noise-filter I think the tutorial is using an older version of SOX and REC I'm using sox: SoX v14.3.2 on Ubuntu 12.04 server Can someone please help me ?

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  • Need a little help with sound in a JApplet

    - by jacob schuschel
    I am working on a solitaire game in Java, and i need to implement sound when the desk is shuffled, card flipped, etc. I used the following sites as reference to try and get it to work, but i am getting Null Pointer Exceptions or mishandled URL exception (depending on what i tweak). here Also, i am using netbeans 6.7.1 as my IDE. I will try to break down the code and explain: package cardgame; import java.applet.*; import java.util.logging.Level; import java.util.logging.Logger; import javax.swing.JApplet; import javax.swing.*; import java.io.*; import java.net.*; /** * * @author jacob */ public class Sound extends JApplet { private AudioClip song; // Sound player private String URL = null; private URL songPath; // Sound path /* *sound_1 = shuffling cards *sound_2 = to discard *sound_3 = from discard *sound_4 = cardflip 1 *sound_5 = cardflip 2 */ Sound(String filename) { try { songPath = new URL(getCodeBase(),filename); // Get the Sound URL } catch (MalformedURLException ex) { Logger.getLogger(Sound.class.getName()).log(Level.SEVERE, null, ex); } song = getAudioClip(songPath); // Load the Sound } Sound(int i) { URL = "./sounds/sound_" + i + ".wav"; System.out.println(URL); try { songPath = new URL(URL); // Get the Sound URL song = getAudioClip(songPath); } catch (MalformedURLException ex) { Logger.getLogger(Sound.class.getName()).log(Level.SEVERE, null, ex); } } public void playSound() { song.loop(); // Play } public void stopSound() { song.stop(); // Stop } public void playSoundOnce() { song.play(); // Play only once } } The 2 different construcors are for different ways i tried to implement this. The first one creates the filepath, and passes it in. The second one builds the filepath in the constructor, given a sound # (i made a list of what numbers correspond to what sound for reference). I am getting the followig errors out: ./sounds/sound_1.wav Nov 16, 2009 4:14:13 PM cardgame.Sound ./sounds/sound_2.wav SEVERE: null java.net.MalformedURLException: no protocol: ./sounds/sound_1.wav ./sounds/sound_3.wav at java.net.URL.(URL.java:583) at java.net.URL.(URL.java:480) at java.net.URL.(URL.java:429) ./sounds/sound_4.wav ./sounds/sound_5.wav at cardgame.Sound.(Sound.java:46) at cardgame.Game.loadSounds(Game.java:712) at cardgame.Game.(Game.java:62) at cardgame.Main.main(Main.java:25) Nov 16, 2009 4:14:13 PM cardgame.Sound SEVERE: null java.net.MalformedURLException: no protocol: ./sounds/sound_2.wav at java.net.URL.(URL.java:583) at java.net.URL.(URL.java:480) at java.net.URL.(URL.java:429) at cardgame.Sound.(Sound.java:46) at cardgame.Game.loadSounds(Game.java:712) at cardgame.Game.(Game.java:62) at cardgame.Main.main(Main.java:25) Nov 16, 2009 4:14:13 PM cardgame.Sound SEVERE: null java.net.MalformedURLException: no protocol: ./sounds/sound_3.wav at java.net.URL.(URL.java:583) at java.net.URL.(URL.java:480) at java.net.URL.(URL.java:429) at cardgame.Sound.(Sound.java:46) at cardgame.Game.loadSounds(Game.java:712) at cardgame.Game.(Game.java:62) at cardgame.Main.main(Main.java:25) Nov 16, 2009 4:14:13 PM cardgame.Sound SEVERE: null java.net.MalformedURLException: no protocol: ./sounds/sound_4.wav at java.net.URL.(URL.java:583) at java.net.URL.(URL.java:480) at java.net.URL.(URL.java:429) at cardgame.Sound.(Sound.java:46) at cardgame.Game.loadSounds(Game.java:712) at cardgame.Game.(Game.java:62) at cardgame.Main.main(Main.java:25) Nov 16, 2009 4:14:13 PM cardgame.Sound SEVERE: null java.net.MalformedURLException: no protocol: ./sounds/sound_5.wav at java.net.URL.(URL.java:583) at java.net.URL.(URL.java:480) at java.net.URL.(URL.java:429) at cardgame.Sound.(Sound.java:46) at cardgame.Game.loadSounds(Game.java:712) at cardgame.Game.(Game.java:62) at cardgame.Main.main(Main.java:25) Thanks for those who read and more thanks to those who help. I know it is somewhat long, but i would rather get it all out there, than have 50 questions that come back or have people not answer due to lack of initial info. also only lets me post a single link right now, so the links are given below dreamincode.net/forums/showtopic14083.htm stackoverflow.com/questions/512436/java-playing-wav-sounds deitel.com/articles/java_tutorials/20060422/LoadingPlayingAudioClips/index.html

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  • Faster way to convert from 24 bit wav pcm format to float?

    - by LMO
    I need to read data in from a wav file in 24 bit pcm format, and convert to float. I'm using Python 2.7.2. The wave package reads the data in as a string, so what I've tried is: # read in entire wav file wdata = f.readframes(nFrames) # unpack into signed integers and convert to float data = array.array('f') for i in range(0,nFrames*3,3): data.append(float(struct.unpack('<i', '\x00'+ wdata[i:i+3])[0])) # normalize sample values data = data / 0x800000 This is quite a bit faster than my earlier approaches, but still quite slow. Can anyone suggest a more efficient method?

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  • wav stopped working with AudioServices on my iPhone app?

    - by user157733
    I have a wav file that was working fine. It is played using the AudioServices methods. Suddenly it stopped working. The weird thing is if i change he wav file to a different one that works. Any idea what is going on? The non working sound is slightly longer (still <10seconds) but it was originally working so I just can't figure it out. Any suggestions of what to try would be most appreciated. Thanks :-)

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  • issue getting dynamic Config parameter in Grails taglib

    - by Mick Knutson
    I have a dynamic config parameter I want to get like: String srcProperty = "${attrs ['src']}.audio" + ((attrs['locale'])? "_${attrs['locale']}" : '') assert srcProperty == "prompt.welcomeMessageOverrideGreeting.audio" where my config has: prompt{ welcomeMessageOverrideGreeting { audio = "/en/someFileName.wav" txt = "Text alternative for /en/someFileName.wav" audio_es = "/es/promptFileName.wav" txt_es = "Texto alternativo para /es/someFileName.wav" } } While this works fine: String audio = "${config.prompt.welcomeMessageOverrideGreeting.audio}" and: assert "${config.prompt.welcomeMessageOverrideGreeting.audio}" == "/en/someFileName.wav" I can not get this to work: String audio = config.getProperty("prompt.welcomeMessageOverrideGreeting.audio")

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  • How do I use WAV files as a voicemail greeting in Exchange UM?

    - by Doug Luxem
    We are in the process of ditching Cisco Unity for Exchange 2007 Unified Messaging; however, I just came to realization that Exchange doesn't seem to provide a way to upload a WAV file to be used as a voicemail greeting. This could be a problem, since we have several mailboxes that use professionally recorded greetings. I found this article which shows how to access the data through MAPI, but it does not provide a way to upload new files. Note, this is not for the auto attendants, but for actual voicemail greetings.

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  • How to download a wav file from the web to a location on iPhone using NSFileHandle and NSURLConnecti

    - by Jinru
    Hi all, I want to download a wav file from a web service, cache it on the iphone and playback it using AVAudioPlayer. Using NSFileHandle and NSURLConnection seems a viable solution when dealing with relatively large files. However, after running the app in the simulator I don't see any saved file under the defined directory (NSHomeDirectory/tmp). Below is my basic code. Where am I doing wrong? Any thoughts are appreciated! #define TEMP_FOLDER [NSHomeDirectory() stringByAppendingPathComponent:@"tmp"] - (void)downloadToFile:(NSString*)name { NSString* filePath = [[NSString stringWithFormat:@"%@/%@.wav", TEMP_FOLDER, name] retain]; self.localFilePath = filePath; // set up FileHandle self.audioFile = [[NSFileHandle fileHandleForWritingAtPath:localFilePath] retain]; [filePath release]; // Open the connection NSURLRequest* request = [NSURLRequest requestWithURL:self.webURL cachePolicy:NSURLRequestUseProtocolCachePolicy timeoutInterval:60.0]; NSURLConnection* connection = [[NSURLConnection alloc] initWithRequest:request delegate:self]; } #pragma mark - #pragma mark NSURLConnection methods - (void)connection:(NSURLConnection *)connection didReceiveData:(NSData*)data { [self.audioFile writeData:data]; } - (void)connection:(NSURLConnection *)connection didFailWithError:(NSError*)error { NSLog(@"Connection failed to downloading sound: %@", [error description]); } - (void)connectionDidFinishLoading:(NSURLConnection *)connection { [connection release]; [audioFile closeFile]; }

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  • Play a wav file retrieved from a database on the iPhone?

    - by user312917
    I have alot of wav files stored in sqlite3, but when I retrieve one of them, I can't play it. The retrieve code is NSData *soundData = (NSDATA *)sqlite3_column_blob(statement, 0); mPlayer = [[AVAudioPlayer alloc] initWithData:soundData error:&error]; The data is stored as binary and it's there when I search for it using sqlite3.

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  • Creating stereo file from two mono files with sox

    - by haimg
    I'm using sox 14.2.0 on Centos 6.0. I have two mono wav files, left.wav and right.wav. I need to combine them into one stereo.ogg file, with left.wav pan 80% to the left, and right.wav pan 80% to the right. I was unable to come up with the sox options needed for this. How do I do this? This is going to be executed repeatably for many files, so I'd prefer an efficient solution. From what I understand there should be a way to do it in one pass (one invocation of sox).

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  • Bash script runs fine, but not in cron

    - by radiotech
    I have a script that's supposed to record a shoutcast stream for an hour, convert it to mp3, and then save it. The script runs correctly when I run it from the terminal, but I can't seem to get it to run in cron (where it should run every hour at the top of the hour). Here's the line in crontab: 0 * * * * /medialib/tech/bin/recordstream 2>&1 >> /medialib/tech/cron.log and here's the script: #!/bin/bash name="$(date +%s)" mp3_name=$name.mp3 wav_name=$name.wav timeout -sHUP 60m vlc -I dummy --sout "#transcode{channels=2}:std{access=file,mux=wav,dst=/medialib/stream_backup/wav/$wav_name" /medialib/tech/lib/listen.m3u lame --mp3input /medialib/stream_backup/wav/$wav_name /medialib/stream_backup/$mp3_name rm /medialib/stream_backup/wav/$wav_name Thank you! EDIT: Contents of cron.log (This text has been in the log file since it was transferred from an old server where it was working). VLC media player 2.0.8 Twoflower Command Line Interface initialized. Type `help' for help. > Shutting down. VLC media player 2.0.8 Twoflower Command Line Interface initialized. Type `help' for help. > Shutting down.

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  • Encoding with FFmpeg using a FIFO

    - by Ashot Martirosyan
    Hello everyone. I'm trying to convert Flac audio file to AAC file using command line. So I wrote this ffmpeg -i input.flac temp.wav faac -q 120 -o output.m4a temp.wav It's working fine. Now I want to do the same using fifo, so I'm writing this mkfifo temp.wav ffmpeg -i input.flac temp.wav & faac -q 120 -o output.m4a temp.wav And it's freezing. So could you tall me what I'm doing wrong. Thanks a lot, and sorry for my English.

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  • Mix Audio tracks with offset in SOX

    - by Laramie
    From ASP.Net, I am using FFMPEG to convert flv files on a Flash Media Server to wavs that I need to mix into a single MP3 file. I originally attempted this entirely with FFMPEG but eventually gave up on the mixing step because I don't believe it it possible to combine audio only tracks into a single result file. I would love to be wrong. I am now using FFMPEG to access the FLV files and extract the audio track to wav so that SOX can mix them. The problem is that I must offset one of the audio tracks by a few seconds so that they are synchronized. Each file is one half of a conversation between a student and a teacher. For example teacher.wav might need to begin 3.3 seconds after student.wav. I can only figure out how to mix the files with SOX where both tracks begin at the same time. My best attempt at this point is: ffmpeg -y -i rtmp://server/appName/instance/student.flv -ac 1 student.wav ffmpeg -y -i rtmp://server/appName/instance/teacher.flv -ac 1 teacher.wav sox -m student.wav teacher.wav combined.mp3 splice 3.3 These tools (FFMEG/SoX) were chosen based on my best research, but are not required. Any working solution would allow an ASP.Net service to input the two FMS flvs and create a combined MP3 using open-source or free tools.

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  • Playing Multiple Sounds at Once in Qt

    - by Ben
    I'm trying to play background music along with sound effects using Qt. However, I can't get more than one sound to play at once. For example: QSound::play("Music.wav"); QSound::play("Effect.wav"); When this code is run (on Windows), you can hear Music.wav just start to play, but then it stops and Effect.wav plays. Is there any way to get the two sounds to play at once?

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  • Play .WAV under Mono on Mac OS X (Snow Leopard)?

    - by Bob Denny
    The Mono 2.6 distribution contains System.Media.SoundPlayer, but attempts to play result in no sound (and no errors) on Mac OS X. All I can find with Google search is obscure references to ALSA. I posted to the Mono-OSX list, but there have been on replies there. I hope someone here has an answer. I think I need to tap into CoreAudio, but don't know how from Mono/C#.

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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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