similar to: transcoding G.711 (u-law) to Speex

Just an update, and a follow-up question: I'm making progress on this issue, and will likely have something working very soon, now that I understand how the jspeex transcoding classes work. Unfortunately, I will need to make a couple small changes to the jspeex source code. My question is - is anybody using jspeex for streaming speex-encoded audio? It works great for static audio, but seems

Have you tried typing "speex rtp" into google code search? It gives lots of examples of real applications which do exactly that. http://www.google.com/codesearch?as_q=speex+rtp -SteveK On 11/1/10 1:13 PM, "Jeff Ramin" <jeff.ramin at singlewire.com> wrote: > >Thanks again Steve. I'll search for the term you mention below. > >What I really want is to

Yes, I have made that search, but I'm restricted to Java. On 11/01/2010 12:21 PM, Steve Kann wrote: > Have you tried typing "speex rtp" into google code search? It gives lots > of examples of real applications which do exactly that. > > http://www.google.com/codesearch?as_q=speex+rtp > > > -SteveK > > > On 11/1/10 1:13 PM, "Jeff

Thanks again Steve. I'll search for the term you mention below. What I really want is to take the output of the speex encoder and spit it out on the network via RTP. I haven't been able to find a library or code example that does that. On 11/01/2010 12:03 PM, Steve Kann wrote: > Jeff, > > It's in the manual: > >

Jeff, It's in the manual: http://www.speex.org/docs/manual/speex-manual/node10.html (table 3 and 4). However, if you're asking this, you're probably trying to do something wrong, or the hard way. You probably shouldn't be taking speex output, and trying to "count bytes". If you are using the API, then you will just get the bits out, and then you'll know how

Is Jspeex still being worked on? Is there somewhere I can send the code changes I've made to facilitate smooth streaming? On 11/09/2010 11:33 AM, Jeff Ramin wrote: > > Just an update, and a follow-up question: > > I'm making progress on this issue, and will likely have something working > very soon, now that I understand how the jspeex transcoding classes work. >

Thanks Steve. Is there a document anywhere that shows how many bytes/bits of data are produced by the speex encoding process for a given amount of time sampling rate and quality setting? On 11/01/2010 09:41 AM, Steve Kann wrote: > Jeff, > > RFC-5574 is standards-track: http://tools.ietf.org/html/rfc5574 so, > while it's not an approved standard, it's more standardized

Jeff, RFC-5574 is standards-track: http://tools.ietf.org/html/rfc5574 so, while it's not an approved standard, it's more standardized than a lot of interoperable traffic on the internets these days. The RFC specifies packetization guidelines, which is basically that you put one or more frames in a packet, and then pad the rest with 0 bits until you have a while number of octets.

On 11/05/2010 02:13 PM, Pascal Pochol wrote: > Jeff, > >> I found the problem I was having (described below), and I'm now able to >> hear the audio at the destination. >> >> However, the audio does not play smoothly. It has hiccups and jumps and >> stops. I can't guarantee this is the problem, but it may be related to >> delays >> caused when

I need to stream speex-encoded audio over RTP, which doesn't seem to be standardized yet, so I'm gonna roll my own code. I control both the sending and receiving sides, so I can pretty much do what I want. I want each packet to contain 20ms worth of audio (sampled at 8KHz), and I'm encoding using a constant bit rate and quality:6. Q: how do I determine how many bytes of data go into

I have a couple apps running on my machine; one takes a PCM audio stream, converts it to Speex, and sends it over the network using RTP. The other receives the RTP packets, and then converts the Speex data back to PCM. The PCM is then played out the audio system. I'm currently ending up w/ static. Anybody have any pointers as to how to debug the situation? Thanks. -- Jeff Ramin Software

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

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

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

Jeff, > I found the problem I was having (described below), and I'm now able to > hear the audio at the destination. > > However, the audio does not play smoothly. It has hiccups and jumps and > stops. I can't guarantee this is the problem, but it may be related to > delays > caused when the Speex2PcmAudioInputStream needs to process an Ogg > page header. >

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 >

Max Lapshin <max.lapshin at gmail.com> writes: > On Wed, Apr 14, 2010 at 3:11 AM, Randy Yates <yates at ieee.org> wrote: >> Can someone please point me to such an example file? Note that I prefer >> speex encoded as a sound stream and NOT a sound event in order to reduce >> latency on playback at the client. However, an example of speex encoded >> as an

Hi, Vipin, What you're going to want to do here is convert your 8-bit stereo samples into 16-bit mono samples and back again when you're done(*). What you are actually doing here, it sounds like, is feeding 2 8-bit mono samples into Jspeex, which expects 16-bit mono. So, the most-significant-byte is in the right channel, and more or less gets encoded/decoded properly, but the

hello, i switched to use the encoder.processData() and encoder.getProcessedData() of jspeex. however it looks to me like a memory leak ... memory usage is increasing very fast and there is no visible stop ... after about five minutes java.lang.OutOfMemory occurs. I think it must be the jspeex component, because before i added jspeex to my app usage was constant at about 5mb. is it possible

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL: