similar to: Invalid sideband mode encountered

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

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

FLV contains TC messages? TC message payload contains a format byte and speex frames (up to eight). In the format byte 0xb0 indicates speex. Speex is always 16 kHz, 16 bit, mono. Jozsef Message: 1 Date: Mon, 16 Nov 2009 14:40:20 -0600 From: Jeff Ramin <jeff.ramin at singlewire.com> Subject: [Speex-dev] jspeex question To: speex-dev at xiph.org Message-ID: <4B01B8B4.8020904 at

The link is http://www.adobe.com/devnet/rtmp/. TC Message stands for TinCan message. It is 11 bytes long, first byte is message type, three bytes of payload length four bytes of timestamp and three bytes of stream ID. The first byte of the payload for audio message is the format byte and the rest of the byte is the payload. Jozsef ----- Original Message ---- From: Jeff Ramin <jeff.ramin

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

Jean-Marc Valin wrote: >>a simple question which i can't answer through the manuals ... is the >>header always 80 bytes long? >>thanks in advance, >> >> > >Right now it is. However, I may add some fields (or even additional >headers) in the future if needed. This shouldn't be a problem since the >header is always alone in his packet and

hi, a simple question which i can't answer through the manuals ... is the header always 80 bytes long? thanks in advance, ulrich -- Ulrich B. Staudinger http://www.die-horde.de email: us@die-horde.de jid: uls@jabber.org current project: REDHORN http://redhorn.sourceforge.net Blog: uls@jabber.org">http://jabber.linux.it/jogger/user.php?jid=uls@jabber.org</a> <p>---

Unfortunately the DP MST helpers do not have much in the way of debugging utilities. So, let's add some! This adds basic debugging output for down sideband requests that we send from the driver, so that we can actually discern what's happening when sideband requests timeout. Since there wasn't really a good way of testing that any of this worked, I ended up writing simple selftests

Hi everyone, I just changed the meaning of the timestamp in the jitter buffer. If you don't know what I'm talking about, then you're not affected. If you do use the jitter buffer, than you will need to change your code accordingly, so instead of bumping the timestamp by 20 (ms) for each frame, you'll have to increase by 160 (samples) for narrowband or 320 for wideband. The new

> what happens if this number flows over? It is just a "int", so it might reach > its limits at 2^15 = 32768, that happens after 102 puts... I would say that an int is 32 bits :-) Actually, RTP defines the timestamp as a 32-bit value. Now, what happens when it overflows (3 days for narrowband), I don't know what the RFC says about it. > In my current > implementation

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

On Saturday 01 October 2005 09:19, Jean-Marc Valin wrote: > I just changed the meaning of the timestamp in the jitter buffer. If you > don't know what I'm talking about, then you're not affected. If you do > use the jitter buffer, than you will need to change your code > accordingly, so instead of bumping the timestamp by 20 (ms) for each > frame, you'll have to

On Wednesday 05 October 2005 13:53, Jean-Marc Valin wrote: > > what happens if this number flows over? It is just a "int", so it might > > reach its limits at 2^15 = 32768, that happens after 102 puts... > > I would say that an int is 32 bits :-) Actually, RTP defines the > timestamp as a 32-bit value. Now, what happens when it overflows (3 days > for narrowband),

Another thing i noted when looking at the java code is a package counter ... is it important to know at which package the stream pointer stands? I mean, if it is important to continously increment the package counter i would have to repackage the audio data for every listener .... which can become quite CPU loady ... ulrich <p><p>Ulrich B. Staudinger wrote: > Jean-Marc Valin

Hi, I downloaded Ulrich's TestClient and have not been able to connect to the default server. I will try to modify and get to work locally(local echo). I wanted to test my SpeexDecoder as no matter what I try I alwasy get an java.io.StreamCorruptedException: Invalid sideband mode encountered. (2nd sideband): 7 Please note that the Invalid sideband modes differ per file but the end result

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

Is there a document somewhere that describes speex-encoded FLV files? What is a TC message? Thanks. Jozsef Vass wrote: > FLV contains TC messages? TC message payload contains a format byte and speex frames (up to eight). In the format byte 0xb0 indicates speex. Speex is always 16 kHz, 16 bit, mono. > > Jozsef > > > Message: 1 > Date: Mon, 16 Nov 2009 14:40:20 -0600 >

Thanks for the help folks, but I got this working a couple hours ago. =) I'm quite please after struggling with it for a few days. I just needed to take each audio tag from the FLV file and feed the contents of the tag (except for the first byte) to the jspeex decoder and write the results to a file. Jozsef - it is possible to specify 8KHz in the flash client and decode it as such. Speex

Hi, I'm Ossama Khayat, an Arab .Net Framework developer. I'm interested in building an API or COM component, of the speex codec, that is suitable for use in a Windows Forms application. This will be used in an Open Source project that will hopefully be hosted in Source Forge one done. I tried downloading the source code and compiling it but really couldn't do much with it, especially