similar to: SpeexEncoder requires 320 samples to process a Frame, not 160

Here is the patch for making syslinux enable opening partitions with fat filesystems (well, the architecture is in place to make it open other filesystems too). Usage is like this: #include <syslinux/io.h> #include <syslinux/partitions.h> #include <fs/fat/libfat.h> syslinux_devdesc dfd; struct libfat_filesystem* fs; diskinfo disk_info; char mbr[512]; static part_entry

Hey Alex, Found a solution to the problem that was occuring with wxMSW and the Wx::TextCtrl, have yet to verify that this will work in linux, but my little trick to the matter, was simple, thanks to the things that are provided with both Wx::TextCtrl, and Wx::Window classes. I''m providing this bit of information for thoes who may have trouble with dealing with Wx::TextCtrl, _AND_ have

hi, i did some tests about performance in jspeex ... uls@mainframe:~/eclipse/avrelay$ java TestClient Diff1: 0 10 Diff2: 141 started. uls@mainframe:~/eclipse/avrelay$ <p>{ SpeexEncoder e1=new SpeexEncoder(); e1.init(0,1,44100,1); byte[] by1=new byte[320]; long l1=System.currentTimeMillis(); long l2=System.currentTimeMillis(); System.out.println("Diff1:

Hi guys I succesfully got my encoder and decoder working after much hassles, but when I use the same code in another project, I get these following errors: Error ---> Invalid sideband mode encountered (1st sideband): 7 Error ---> Invalid sideband mode encountered (1st sideband): 7 Error ---> Invalid sideband mode encountered (1st sideband): 6 Error ---> More than two

hi there, I am new to mailing list so excuse me if I don't obey to the 'netiquette'. i am writing voice chat and speex is in the root of it. i write it in Java and use JNI to link with 'C'-based Speex 1.2beta. [I know of JSpeex but there are not implemented some features] recently i decided to use DTX feature of speex as well. the code follows. The problem is that no matter

Okay, how do I drop a changeset/patchset/tag for you folks from SVN? At this point, I have written three examples of how to use the basics of the ogg streaming and decoding in Tremor. I heartily welcome any suggestions, improvements and corrections that you can point out in the code. The examples required me to make some small modifications to the main tremor library. However, the changes

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

Hi all, First of all, thanks to Speex developper for the all the job. I am trying to implement my own Speex DirectShow fitlers for VoIP following the documentation and sample code's. I am facing audio glitches when encoding - decoding PCM data. The encoder and decoder procedures are copied below. What about the lookahead size ? how shouw we apply it in encoding stage ? thanks in advance

Hello, I'm having a problem using Speex - I'm getting white noise/static from the following process: (Load File) -> (Encode) -> (Decode) -> (Play) I know there's no issue with the loading or playing because (Load File)-> (Play) plays the file properly. I believe the problem lies with the speex_bits_write and speex_bits_read_from functions, because if I change my encode

hi, i'm using jspeex to encode signed little endian pcm 16bits data to speex, so i can then publish it to a flash client. On rtmp spec is said that is used network byte order, so in a speex frame consisted in more than 1 byte, i guess i have to send it in big endian order. By the source code of jspeex i see that when i pass pcm data as a parameter to SpeexEncoder.processData they use it as

Hey there =] I intend to develop a voip application using the Jspeex (Java implementation of Speex) library. At this point, the progrm can capture the voice from the microphone to an "ByteArrayOutputStream" object and play it back. I`m willing to add the encode\decode features of Jspeex but I can't make it to work, I`m probably missing somethnig. The code (Part of it) : -----

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

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(),

Is There any intension to deal with mixing two or more streams encoded with "speex". Most voice wise applications (such as conference) need this feature. Oded Rephael <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org'

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

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);

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

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

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

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