similar to: JSPEEX ?

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

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

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

Since two people have asked me to open the source code on my work-in-progress Integer Speex project, I have put it up onto my site www.openacm.org as a download in the project tree. the suggested directory structure is as follows, i.e. the speex source folder should be on the same directory level as the sphinx filder. peex-1.0.1 (ideally the folder from within the SpeexACM source archive)

Hello, I am unable to get the backgroundrb server to run. Initially I got: /usr/local/lib/ruby/gems/1.8/gems/activesupport-2.0.1/lib/active_support/dependencies.rb:263:in `load_missing_constant'': uninitialized constant BackgrounDRb::MasterProxy (NameError) from /usr/local/lib/ruby/gems/1.8/gems/activesupport-2.0.1/lib/active_support/dependencies.rb:453:in

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

I noticed a little while back that Andrew@teledesign.co.uk posted he was going to port the speex onto a TI DSP. Any news on this or similar developements? --- >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' containing only the word 'unsubscribe' in

Hello all, I''m new to Ruby and have come across this project that seems to be the most mature scheduling project for Ruby. I''ve had some brief experience with Quartz for Java and was wondering if there were any plans to provide an option to save scheduled jobs into some sort of data store? If not, is that something the project would be interested in? Has someone

We are bellow 30MIPS for our 16bit single MAC fixed point DSP. 8khz/8kbps I aggree with Jean-Marc that MIPS is dependent from the target DSP, but it's a very good indication when you compare same architecture DSPs. It is very difficult (impossible) to use effecient a DSP using only c. George ----- Original Message ----- From: "Andre Borrelly" <andre@myavalaunch.com> To:

Le mer 31/03/2004 à 02:51, George Bratis a écrit : > I think that if you want to use a low cost dsp you must rewrite ti in > assebmly. > In our tests you will need 4-5 times more MIPS just compiling C code. You may need to rewrite 3-4 compute-intensive, but definitely not the whole thing. Jean-Marc -- Jean-Marc Valin http://www.xiph.org/~jm/ LABORIUS Université de Sherbrooke,

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, 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

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., 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

Hi, Is real-time possible with the current java-based JSpeex for real-time full duplex voice communication? If yes, can you please also provide some details of the Wintel (Windows on Intel) specifications that you have used to achieve that, and perhaps the JSpeex encoding/decoding specs used? Thanks. Cheers, Jo Yew Everyone should have a big dream and live a meaningful life to its

how up to date is the jspeex? --------------------------------- Yahoo!7 Mail has just got even bigger and better with unlimited storage on all webmail accounts. Find out more. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070616/75aec5d2/attachment.html

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:

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

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