I have a server that sends data via a socket, the data is a wav 'file'. I can easily write the data to disk and then play it in WMP, but I have no idea how I can play it as I read it from the socket. Is it possible?
Bonus question: how would I do it if the stream was in mp3 or other format?
This is for windows in native C++.
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Given an InputStream called in which contains audio data in a compressed format (such as MP3 or OGG), I wish to create a byte array containing a WAV conversion of the input data. Unfortunately, if you try to do this, JavaSound hands you the following error:
java.io.IOException: stream length not specified
I managed to get it to work b...
I have a 10 second sound effect wave file. What I would like to do is take that file and repeat it n number of times and then save the longer WAV file to disk. This way I can create a much longer background effect rather than auto-repeat on the media player which is a bit stuttered between repeats. I am trying to do this in C#.
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We have a JavaScript construct that will play .wav files within Firefox on Windows and MacOSX, but it does not work for Red Hat Linux. What extension do I need?
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Hi, I have 2 wav files that I want to make/combine/merge into one file with the two sound tracks.
Is there any api for that task or some built inn commands in .NET that can I can use in som genius way to make this task possible?
Many thanks for the help. :)
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What is the easiest way to programatically convert a compressed wav file (MPEG codec for example, but could be any installed codec) to an uncompressed wav file (16 bit PCM)?
I've heard that using direct show and writing the code in native C++ would do it, but I've not had much experience with direct show.
Is there an easier way to do ...
How do you programmatically compress a WAV file to another format (PCM, 11,025 KHz sampling rate, etc.)?
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How can you convert wav files with MPeg layer 3 (MPeg3) audio format to PCM?
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I tried this:
sox -u -r 11.025k -b 8 -c 1 infile.wav outfile.wav
but the sound is garbled and unrecognizable when I play it. Help?
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Why do i get distorted output if I convert a wav file using libsox to:
&in->encoding.encoding = SOX_ENCODING_UNSIGNED;
&in->encoding.bits_per_sample = 8;
using the above code?
The input file has bits_per_sample = 16.
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The digital sound is playing using DirectSound device. It is necessary to display sound activity in decibels - like analog devices do.
What is the right way to calculate sound pressure from the WAVE PCM data (44100hz,16bit)?
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how do you convert a mp3wav (a compressed wav in mp3 form) to uncompressed wav (PCM) using sox?
mp3wav sample files can be downloaded here: http://www.clayloomis.com/simsong.html
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What is the difference between compressed and uncompressed .wav files?
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I'm trying to play a
PCM_UNSIGNED 11025.0 Hz, 8 bit, mono, 1 bytes/frame
file as described here (1) and here(2).
The first approach works, but I don't want to depend on sun.* stuff. The second results in just some leading frames being played, that sounds more like a click. Can't be an IO issue as I'm playing from a ByteArrayInputStre...
Is it possible to create a temporary file that contains a "loop" of a wav file?
Or is it possible to manipulate the stream sent to a stream reader/writer?
Basically I want to play some wav file for a period of time and if that time is greater than the length of time that the wav file provides I want to loop.
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I have a web service that returns an array of bytes and my intention is to convert said array to a .wav in the client (a handheld such as Blackberry). However I really have no idea of how to do that, I tried just making an FileOutputStream but of course that wouldn't play. So I am once again without knowing what to do. Any ideas?
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I am looking for a cross-platform (.Net and Mono on Windows, MacOSX, and Linux) way to decode wav, mp3, or ogg files such that I can then play the decoded streams through DirectSound or OpenAL as desired.
A solution that can decode either mp3 or ogg would be sufficient -- decoding both is not necessary. If it (or another solution) can d...
I'm trying to set artist information via variables with spaces in them. Lame craps out. Maybe I'm being retarded with bash?
#!/bin/bash
year=2008;
artist="New Kids On The Block";
album="The Block";
bitrate=320;
lame="lame -b $bitrate --ta \"$artist\" --tl \"$album\" --ty $year"
function first_half
{
for (( i=1;i<10;i++ )); do
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I need to load sound files to a Cocoa-based OpenAL app.
Progress:
The OpenAL utility function alutLoadWAVFile has been deprecated; the alut header is no longer included in Mac OS X SDKs. According to the TechNotes, the actual code is still there for binary compatibility. However, if I attempt to add a declaration for the function, t...
I'm interested in precisely extracting portions of a PCM WAV file, down to the sample level. Most audio modules seem to rely on platform-specific audio libraries. I want to make this cross platform and speed is not an issue, are there any native python audio modules that can do this?
If not, I'll have to interpret the PCM binary. Whi...