similar to: jspeex question

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

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

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 >

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

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 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

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. >

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

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

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.

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

Hi folks. The jspeex library has classes for converting speex to pcm and vice-versa. I also have other code that converts from G.711 to pcm (and vice-versa). I want to transcode G.711 to speex, using an input stream. Can I accomplish this in one step, or must I go G.711 -> PCM -> Speex? If possible in one step, is there some example code I could look at for reference? Thanks! -- Jeff

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

Flash player encodes speex at 16 kHz, mono, 16 bit. Fields in the format byte should be ignored if the format is speex. You can set the quality by Microphone.encodeQuality (default 6). You can also set the number of speex frames per tc message using Microphone.framesPerPacket. Flash player can only decode speex at 16 kHz, so make sure you have the proper sample rate. Jozsef > > Message:

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

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. >

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

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: > >