Displaying 20 results from an estimated 200 matches similar to: "help with jspeex"
2008 Jan 07
0
JSpeex - Unsupported conversion
I'm having difficulty encoding audio using the JSpeex Speex Encoder. My
program throws an "Unsupported conversion" exception and I can't figure
out why. I've read the related posts and I think I'm doing everything that
was recommended. I'm working on Linux, by the way.
Any help would be greatly appreciated.
Richard
Here is my program output:
$ java -classpath
2008 Jan 08
0
JSpeex - Unsupported conversion
This is best answered by the jspeex folks, as it is the interfacing
code that you are having problems with.
That being said, it looks like, just as it says, you have asked for a
conversion it cannot do -- you want it to convert PCM data to speex
data at 44.1kHz sample rate -- speex only supports 8kHz in standard
operation. It can also support 16kHz (wb) and 32kHz (uwb), but nothing
outside of
2010 May 06
1
Encoding a wave file with a bad header
If I use Speex, JSpeex actually, to compress an otherwise valid wave file with zero lengths in the header would it impact the compression at all? Here's what I'm doing during compression in Java:
AudioFormat wavFormat = ais.getFormat();
AudioFormat speexFormat =
new AudioFormat(SpeexEncoding.SPEEX_Q5,
wavFormat.getSampleRate(),
2010 Mar 20
0
Decode file written from JSpeex using C/C++ API?
I'm new to Speex and I'm trying to compress audio using JSpeex in a servlet then play it back on the iPhone. I've managed to get Speex to compile on the iPhone by copying the speex and libspeex folders into XCode. I've read the sample code and the PDF documentation and I have a rough idea how to decode a raw stream. (I'm assuming the sampledec source works with raw speex audio
2004 Aug 06
0
Lost ogg sync using jspeex
Hi
I am getting lost ogg sync exception when I tried to decompress a speex stream. I am using the following program for both compressing as well as decompressing. I am just planning to test how jspeex works.
// File AudioCapture.java
import javax.sound.sampled.spi.*;
import javax.sound.sampled.*;
import org.xiph.speex.spi.*;
import java.util.*;
import java.net.*;
import java.io.*;
public
2004 Aug 06
0
Lost ogg sync using jspeex
Hi
I am getting lost ogg sync exception when I tried to decompress a speex stream. I am using the following program for both compressing as well as decomressing. I am just planning to test how jspeex works.
// File AudioCapture.java
import javax.sound.sampled.spi.*;
import javax.sound.sampled.*;
import org.xiph.speex.spi.*;
import java.util.*;
import java.net.*;
import java.io.*;
public
2004 Aug 06
0
q about jspeex
Hi,
It would appear the the 'pcm2speex.read(frame, 0, frame.length)' is
blocking which means that it is waiting for data from the underlying
inputstream (i.e.AudioInputStream(t.input)). If it could read sufficient
data it would transcode it. If it recieved an EOF, it should do some
zero padding and then transcode it. Are you sure that you are receiving
data from the underlying
2004 Aug 06
0
JSpeex help
I am not sure if this is right place to ask for help on jspeex or can some one suggest. I have tried on jspeex sourceforge.net page.
I am trying to use Pcm2SpeexAudioInputStream by creating it from tdl which received pcm_signed audio data as follows:
AudioFormat format = new AudioFormat (AudioFormat.Encoding.PCM_SIGNED, 8000, 16, 1, 2, 8000, true);
DataLine.Info targetInfo = new DataLine.Info
2004 Aug 06
0
Please Help, Lost ogg sync using jspeex
<HTML dir=ltr><HEAD></HEAD>
<BODY>
<DIV id=idOWAReplyText8852 dir=ltr>
<DIV dir=ltr><FONT face=Arial color=#000000 size=2><FONT face=Arial color=#000000 size=2>Hi</FONT></DIV></DIV>
<DIV dir=ltr>
<DIV dir=ltr>
<DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial
2004 Aug 06
0
JSpeex help
Hi Sanjiv,
There does indeed appear to be some trouble with
the Javasound SPI encoder (the SPI decoder works well).
I am working on it, and hope to have a fix soon. In the
mean time, you can use the encoder outside of
JavaSound, that one works well (see the command line
encoder as example code of how to implement it).
sincerely
Marc Gimpel
Head of research
Wimba
<p>On
2004 Aug 06
2
q about jspeex
Ulrich B. Staudinger wrote:
> Hi,
>
> i have:
>
> public void run(){
> try{
> System.out.println("Opening
> mic"); // AudioInput
> ai=new AudioInput(t);
> // ai.start();
> if(t.input==null){
> AudioFormat format = new
2004 Aug 06
0
q about jspeex
Hi,
i have:
public void run(){
try{
System.out.println("Opening mic");
// AudioInput ai=new AudioInput(t);
// ai.start();
if(t.input==null){
AudioFormat format = new
AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, false);
2004 Aug 06
0
q about jspeex
Hi,
i changed the read method after constructing a
pipedinputstream/opipedoutputstream to
int n=auin.read(frame, 0, frame.length);
System.out.println(""+n+" bytes read.");
pout.write(frame);
int m=pcm2speex.read();
System.out.println("something
2004 Aug 06
3
q about jspeex
Hi Marc,
thanks for the quick reply.
Marc Gimpel wrote:
> It would appear the the 'pcm2speex.read(frame, 0, frame.length)' is
> blocking which means that it is waiting for data from the underlying
> inputstream (i.e.AudioInputStream(t.input)). If it could read
> sufficient data it would transcode it. If it recieved an EOF, it
> should do some zero padding and then
2007 Jun 24
0
JSpeex help
Have you looked at MoodleSpeex?
see a demo
http://www.artofart.info/index.php/projects/post-by-post-audio-comments-system/
contact me if you want a copy
----- Original Message -----
> Message: 2
> Date: Fri, 22 Jun 2007 12:21:09 +0400
> From: <mail-box@nm.ru>
> Subject: [speex-dev] JSpeex help
> To: speex-dev@xiph.org
> Message-ID:
2009 Mar 02
2
[Fwd: RE: Mixing inputstreams.]
Thanks Ross,
So you think that I can just numerically add up the appropriate Bytes
from each of the streams? If that's so would we be better add first,
then to normalise after the additon, so the final byte size is the same
as the individual ones, what ever that may be.
-------- Original Message --------
Subject: RE: [Vorbis] Mixing inputstreams.
Date: Tue, 3 Mar 2009 01:12:35 +1300
2006 Jul 18
1
SpeexEncoder requires 320 samples to process a Frame, not 160
Hi guys
I have tried compiling this attached code, I made all the buffers 320, there is no trace of a 160 buffer, but I get a "
SpeexEncoder requires 320 samples to process a Frame, not 160" error.
Maybe there's something I'm missing, here's my code:
import java.io.IOException;
import java.io.FileOutputStream;
import java.io.File;
import
2012 Sep 30
0
Speex (in ios) really poor quality (and robotic) sound
Hi everyone,
I'm trying to encode/decode with speex, when I do not, the audio is loud and clear, but when I encode/decode to test audio quality, I get a really poor audio quality and a robotic sound.
Here's my init audio method :
#define AUDIO_QUALITY 10
- (void) initAudio {
try {
//SPEEX CONFIG
speex_bits_init(&bits_in);
2004 Aug 06
0
Getting a Side band error when decoding with Jspeex
Hi.,
I am trying to implement speex. Currently i am capturIng sound and encoding it using Pcm2SpeexAudioInputStream. I am feeding bytes from this stream to Speex2PcmAudioInputStream and trying to read decoded bytes. But here the decoder is giving me an error "saying sideband error" and values after it vary with every message.
I donno the reason for this, but I guess it is because
2011 Nov 24
1
Wrong WAV AudioFormat
Hi,
I am using vorbis-tools 1.4.0 on Fedora 14 to convert Ogg Vorbis files
to MS WAV format. It seems that ogg123 is encoding the wrong AudioFormat
value. For Uncompressed PCM, the AudioFormat should be 1 but for some
reason, it is being encoded in the WAV file as -1.
This used to work on some prior version of either Fedora or
vorbis-tools. I thought I'd ask in the email list in case