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  • How to get the contents of the wav file into array so as to cut the required segment and convert it

    - by kaushik
    How to get the contents of the wav file into array so as to cut the required segment and convert it back to wav format using python?? My prob is similar to "ROMANs" prob,i hav seen earlier in the post at this site.. Basically,i want to combine parts of different wav file into one wav file?? if there is ne other apporach thn takin the contents into an array and cuting part and combining and again converting bac? please suggest... edited: I prefer unpacking the contents of the wave file into an array and editing by cutting the required segment of sound from the wav file,as i am working on speech processing,and guess this way would be easy to enchance the quality of sound later... can ne one suggest a way for this?? Plz help.. Thanks in advance.

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  • using python,How to cut the wav file between certain time ranges??

    - by kaushik
    How to cut the wav file between certain time ranges from multiple wav files and paste the segments together in a single wav file in continous time ?? For this i thou of a way,to store the contents of the wav file in array form and cut the segments required from the array copy thm in another file and convert it back into wav formant. but i hav no idea how to code it in python as i am a beginner in it.. plz help...any alternative methods which serve the purpose are also welcome.. Quick reply,xpected plzz.. Thanks in advance..

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  • How can I convert audio files to this format?

    - by jeffamaphone
    I have a bunch of audio files that are named .wav but it seems not all .wavs are created equal. For example: $ file * file1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz file2.wav: Audio file with ID3 version 2.2.0, contains: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo file3.wav: Claris clip art? file4.wav: Audio file with ID3 version 2.2.0, contains: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo And for good measure, a non-wav: file5.m4a: ISO Media, MPEG v4 system, iTunes AAC-LC I would like to convert all of these files to the format that file1.wav is: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz What is the proper set of arguments to pass to afconvert to make that happen?

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  • MPlayer refuses to generate mono wav file

    - by JCCyC
    I want to downsample an existing audio file to 8KHz mono. This command line downsamples it to stereo: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1 -ao pcm:waveheader:file="/tmp/blah1.wav" ~/from_my_cellphone.3ga It generates a file that the file utility identifies as stereo: $ file /tmp/blah1.wav /tmp/blah1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Now, if I read the documentation correctly, I should add pan=1:0.5:0.5 so I get a file that's half the size: mplayer -quiet -vo null -vc dummy -af volume=0,resample=8000:0:1:pan=1:0.5:0.5 -ao pcm:waveheader:file="/tmp/blah2.wav" ~/from_my_cellphone.3ga But it doesn't! blah2.wav is identical to blah1.wav! What am I doing wrong?

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  • What is the easiest way to read wav-files using Python [summary]?

    - by Roman
    I want to use Python to access a wav-file and write its content in a form which allows me to analyze it (let's say arrays). I heard that "audiolab" is a suitable tool for that (it transforms numpy arrays into wav and vica versa). I have installed the "audiolab" but I had a problem with the version of numpy (I could not "from numpy.testing import Tester"). I had 1.1.1. version of numpy. I have installed a newer version on numpy (1.4.0). But then I got a new set of errors: Traceback (most recent call last): File "test.py", line 7, in import scikits.audiolab File "/usr/lib/python2.5/site-packages/scikits/audiolab/init.py", line 25, in from pysndfile import formatinfo, sndfile File "/usr/lib/python2.5/site-packages/scikits/audiolab/pysndfile/init.py", line 1, in from _sndfile import Sndfile, Format, available_file_formats, available_encodings File "numpy.pxd", line 30, in scikits.audiolab.pysndfile._sndfile (scikits/audiolab/pysndfile/_sndfile.c:9632) ValueError: numpy.dtype does not appear to be the correct type object I gave up to use audiolab and thought that I can use "wave" package to read in a wav-file. I asked a question about that but people recommended to use scipy instead. OK, I decided to focus on scipy (I have 0.6.0. version). But when I tried to do the following: from scipy.io import wavfile x = wavfile.read('/usr/share/sounds/purple/receive.wav') I get the following: Traceback (most recent call last): File "test3.py", line 4, in <module> from scipy.io import wavfile File "/usr/lib/python2.5/site-packages/scipy/io/__init__.py", line 23, in <module> from numpy.testing import NumpyTest ImportError: cannot import name NumpyTest So, I gave up to use scipy. Can I use just wave package? I do not need much. I just need to have content of wav-file in human readable format and than I will figure out what to do with that.

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  • How do I post a .wav file from CS5 Flash, AS3 to a Java servlet?

    - by Muostar
    Hi, I am trying to send a byteArray from my .fla to my running tomcat server integrated in Eclipse. From flash I am using the following code: var loader:URLLoader = new URLLoader(); var header:URLRequestHeader = new URLRequestHeader("audio/wav", "application/octet-stream"); var request:URLRequest = new URLRequest("http://localhost:8080/pdp/Server?wav=" + tableID); request.requestHeaders.push(header); request.method = URLRequestMethod.POST; request.data = file;//wav; loader.load(request); And my java servlet looks as follows: try{ int readBytes = -1; int lengthOfBuffer = request.getContentLength(); InputStream input = request.getInputStream(); byte[] buffer = new byte[lengthOfBuffer]; ByteArrayOutputStream output = new ByteArrayOutputStream(lengthOfBuffer); while((readBytes = input.read(buffer, 0, lengthOfBuffer)) != -1) { output.write(buffer, 0, readBytes); } byte[] finalOutput = output.toByteArray(); input.close(); FileOutputStream fos = new FileOutputStream(getServletContext().getRealPath(""+"/temp/"+wav+".wav")); fos.write(finalOutput); fos.close(); When i run the flash .swf file and send the bytearray to the server, I receive following in the server's console window:: (loads of loads of Chinese symbols) May 20, 2010 7:04:57 PM org.apache.tomcat.util.http.Parameters processParameters WARNING: Parameters: Character decoding failed. Parameter '? (loads of loads of Chinese symbols) and then looping this for a long time. It is like I recieve the bytes but not encoding/decoding them correctly. What can I do?

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  • SpeechSynthesizer in C# creates wav that has 22kHz... needs to be 16kHz

    - by Adrian
    My C# application needs to covert text to wav file and inject it into a Skype call. The code that creates the wav file is below. The problem is that the file has 22kHz sample rate and Skype accepts only 16kHz. Is there any way to adjust this setting? using (System.IO.FileStream stream = System.IO.File.Create("message.wav")) { System.Speech.Synthesis.SpeechSynthesizer speechEngine = new System.Speech.Synthesis.SpeechSynthesizer(); speechEngine.SetOutputToWaveStream(stream); speechEngine.Speak(number); stream.Flush(); }

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  • How to convert from wav or mp3 to raw PCM [on hold]

    - by Komyg
    I am developing a game using Cocos2d-X and Marmalade SDK, and I am looking for any recommendations of programs that can convert audio files in mp3 or wav format to raw PCM 16 format. The problem is that I am using the SimpleAudioEngine class to play sounds in my game and in Marmalade it only supports files that are encoded as raw PCM 16. Unfortunately I've been having a very hard time finding a program that can do this type of conversion, so I am looking for a recommendation.

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  • Convert Chinese character .wav song into .mp3 or .wma on English OS

    - by Jack
    I have bunch of Chinese .wav files on my hard disk that I'm trying to convert into .mp3 with Audacity but it appear that Audacity can not read Chinese character songs but the .wav file display correctly on my 32 bits Win7 Ultimate(English) pc. I have to rename these Chinese character songs into English file name in order to convert them. Does anyone know if there is any software (prefer open source) that will take Chinese character file name(.wav) and convert it into .mp3 without renaming the file?

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  • WAV and MP3 Streaming with ASP.Net and C#

    In this programming tutorial you will learn how to stream WAV and MP3 audio files in ASP.NET 3.5 using the C# server side language. This is particularly useful for music websites that are based on the ASP.NET 3.5 platform. The examples used in this article are tested to work on any major browser including Internet Explorer Chrome and Firefox. The scripts are tested on a Windows XP operating system using Visual Web Developer Express. For convenience an actual working example can be downloaded at the end of this tutorial. Finally this tutorial also highlights the use of the Google Flash player when streaming MP3s.... Autodesk? Inventor? Test Drive Autodesk? Inventor?. Download A Free 30-Day Trial Today.

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  • Best way to play wav files in the browser?

    - by Splatzone
    I have no choice but to play wav files directly in the browser (serverside encoding to mp3 isn't an option, unfortunately.) What's the best way to do this? I'd really like to take advantage of the HTML 5 audio tag but my target audience includes many, many teens using IE6. As far as I'm aware flash isn't an option, but speedy playback really is critical. Thanks.

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  • Any way to cache WAV files in IE?

    - by Dan Howard
    I'm seeing an issue with our web application. We have a few wave files which we can play (like ding.wav) and we have attempted to pre-load wave files but using Fiddler we're seeing that the WAV files are never cached like (js and css and image files). We always see an HTTP 200 instead of an HTTP 304. Any ideas on how to tell IE that it should cache wav files? We're inserting a div: <EMBED SRC='ding.wav' AUTOSTART='FALSE' HIDDEN='TRUE'>

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  • How do I play back a WAV in ActionScript?

    - by Jeremy White
    Please see the class I have created at http://textsnip.com/51013f for parsing a WAVE file in ActionScript 3.0. This class is correctly pulling apart info from the file header & fmt chunks, isolating the data chunk, and creating a new ByteArray to store the data chunk. It takes in an uncompressed WAVE file with a format tag of 1. The WAVE file is embedded into my SWF with the following Flex embed tag: [Embed(source="some_sound.wav", mimeType="application/octet-stream")] public var sound_class:Class; public var wave:WaveFile = new WaveFile(new sound_class()); After the data chunk is separated, the class attempts to make a Sound object that can stream the samples from the data chunk. I'm having issues with the streaming process, probably because I'm not good at math and don't really know what's happening with the bits/bytes, etc. Here are the two documents I'm using as a reference for the WAVE file format: http://www.lightlink.com/tjweber/StripWav/Canon.html https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ Right now, the file IS playing back! In real time, even! But...the sound is really distorted. What's going on?

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  • WAVEFORMATEX - how to read codecdata at the end??

    - by Roey
    Hi All. I've a WAVEFORMATEX struct with some codecdata at the end of it (10 bytes). I'm using C++. How do I access the data at the end? (this is a purely technical question). I tried : WAVEFORMATEX* wav = (WAVEFORMATEX*)pmt->pbFormat; WORD me = wav->cbSize; wav = wav + sizeof(WAVEFORMATEX); BYTE* arr = new BYTE[me]; memcpy(arr, (BYTE*)wav, me); Didnt work. Thanks Roey

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  • How to extract a Vorbis stream from a WAVE file?

    - by H.B.
    I would like to move the Vorbis stream into an ogg container but ffmpeg does not seem to recognize the stream. Even though MPlayer gives this output upon playback: Opening audio decoder: [acm] Win32/ACM decoders Loading codec DLL: 'vorbis.acm' Loaded DLL driver vorbis.acm at 10000000 Warning! ACM codec reports srcsize=0 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000-176400) Selected audio codec: [vorbisacm] afm: acm (OggVorbis ACM) ffmpeg: ffmpeg -i Source.wav -acodec copy Target.ogg Input #0, wav, from 'Source.wav': Duration: 00:02:15.17, bitrate: 128 kb/s Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s [ogg @ 00000000003096C0] Unsupported codec id in stream 0 Output #0, ogg, to 'Target.ogg': Metadata: encoder : Lavf53.6.0 Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Could not write header for output file #0 (incorrect codec parameters ?) Of course this does not necessarily need to be done via ffmpeg, any method that is workable would be fine... I have cut down one of the files to 512KB: sample.wav (Changed two chunk size fields in the wave header to account for this, the embedded stream is cut "without notice")

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  • Android:Playing bigger size audio wav sound file produces crash

    - by user187532
    Hi Android experts, I am trying to play the bigger size audio wav file(which is 20 mb) using the following code(AudioTrack) on my Android 1.6 HTC device which basically has less memory. But i found device crash as soon as it executes reading, writing and play. But the same code works fine and plays the lesser size audio wav files(10kb, 20 kb files etc) very well. P.S: I should play PCM(.wav) buffer sound, the reason behind why i use AudioTrack here. Though my device has lesser memory, how would i read bigger audio files bytes by bytes and play the sound to avoid crashing due to memory constraints. private void AudioTrackPlayPCM() throws IOException { String filePath = "/sdcard/myWav.wav"; // 8 kb file byte[] byteData = null; File file = null; file = new File(filePath); byteData = new byte[(int) file.length()]; FileInputStream in = null; try { in = new FileInputStream( file ); in.read( byteData ); in.close(); } catch (FileNotFoundException e) { // TODO Auto-generated catch block e.printStackTrace(); } int intSize = android.media.AudioTrack.getMinBufferSize(8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT); AudioTrack at = new AudioTrack(AudioManager.STREAM_MUSIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT, intSize, AudioTrack.MODE_STREAM); at.play(); at.write(byteData, 0, byteData.length); at.stop(); at.release(); } Could someone guide me please to play the AudioTrack code for bigger size wav files?

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  • MP3 codec for WAV files

    - by Don Reba
    Wav files support different encodings, including mp3. Is there a C/C++ library that would produce mp3-encoded wav files from uncompressed wav? If not, what would be the best place to start to implement one?

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  • Inserting a WAV at a certain point in an audio file using python

    - by Onion
    My problem is the following: I have a 2-minute long WAV file, and my aim is to insert another WAV file (7 seconds long), at a certain point in the first WAV file (say, 0:48), essentially combining the two WAVs, using python. Unfortunately I haven't been able to figure out how to do that, and was wondering if there was some obvious solution that I was missing, or if it is even feasible to do with python. Is there perhaps a library available that might provide a solution? Thanks to all in advance.

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  • Found your wavplayer but can't make it work...

    - by ifoks
    Hello man, I was looking for weeks an audio wav flash player and I found your blog where you post your WavPlayer, I download it and place it on my web site. I tried to read a wav file but it can't, I check it with the debug player and i found the problem, it come from that line : FileWav.hx:58 : Wrong RIFF magic! got 1974609456 instead of 0x46464952 But my audio files really are WAV files ! You're the only one who create a wav player you're my only hope ! Please if you see that message write me at [email protected] (e-mail adress), I really need your help on this man ! Thank's

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  • What trick will give most reliable/compatible sound alarm in a browser window for most browsers

    - by Dirk Paessler
    I want to be able to play an alarm sound using Javascript in a browser window, preferably with the requirement for any browser plugins (Quicktime/Flash). I have been experimenting with the tag and the new Audio object in Javascript, but results are mixed: As you can see, there is no variant that works on all browsers. Do I miss a trick that is more cross-browser compatible? This is my code: // mp3 with Audio object var snd = new Audio("/sounds/beep.mp3");snd.play(); // wav with Audio object var snd = new Audio("/sounds/beep.wav");snd.play(); // mp3 with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.mp3" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); // wav with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.wav" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); }

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