similar to: Decoding

It's not playing at half speed... It's playing at 2x speed. -Dragon Jean-Marc Valin wrote: > If it plays at half the speed, it means that either you're setting the > sampling rate wrong or (if you're using Linux) the machine at the other > end is using the broken i810 OSS audio driver which can't set itself to > mono. About writing code properly, just look at

What do you mean by write my code properly? With the way it's written right now it is encoding writing out to a stream and decoding from that stream but i'd like to use VBR but when i try it my app fails. and also on a different note I've written some test apps where it recordes from the mic encodes decodes then playes but when i try streaming it over the internet it has long

Well I've comverted to UDP but still have the tick and it can't be the diff soundcards cause it does it on the same computer. Also the wave out buffer keeps growing as if extra data is comming from somewhere eg: the wave out plays just as fast as the wave in so the wave out buffer should not "grow". -Dragon (dragon@dazoe.net) (www.dazoe.net) Jean-Marc Valin wrote:

Le samedi 04 juin 2005 ? 23:38 -0500, Dragon a ?crit : > It's not playing at half speed... It's playing at 2x speed. That exactly what I mean, if you play a mono stream to a soundcard set to stereo, it'll play twice as fast. > -Dragon > > Jean-Marc Valin wrote: > > If it plays at half the speed, it means that either you're setting the > > sampling rate

If it plays at half the speed, it means that either you're setting the sampling rate wrong or (if you're using Linux) the machine at the other end is using the broken i810 OSS audio driver which can't set itself to mono. About writing code properly, just look at speexenc/speexdec if you want to have an example of Ogg transport. Jean-Marc Le samedi 04 juin 2005 ? 23:29 -0500, Dragon

Hello, i working on a VoIP app and i'm useing TCP to keep packets in order that's not my issue tho, I've solved my cpu useage problem but now i have a tick sound every time i fill a buffer there is a tick does the decoder add a small amount of sicelence to each frame or what else could it be? -- -Dragon (dragon@dazoe.net) (www.dazoe.net)

Hello, I have been running Shorewall for quite some time at an ISP client of mine to protect his LAN. We have just upgraded to 2.2.4 and he now wants to put his servers in a DMZ. The servers have public IPs in two classes xxx.xxx.79.0 and xxx.xxx.242.0. The public IP on the router for each class is xxx.xxx.79.126 and xxx.xxx.242.126. I am using masq and 192.168.1.0 on eth0 LAN I have tried

Hi evrybody, I wonder witch hardware requirements it is for speex decoding (especially 16kHz mono sound). Could it be possible to use the Gameboy advance hardware if you added a FPU? or dose it need to be more powerfull? ----------------------------- Reclaim Your Inbox! http://www.mozilla.org/products/thunderbird/ Rediscover The Web! http://www.mozilla.org/products/firefox/

Compared to todays PDAs the GameBoy Advance (GBA) is low spec. it has a 16MHz ARM7tdmi RISC processor. 32-bit but it can handle 16-bit instructions in a mode that resembles "bytecode" (a 32-bit instruction is executed using a 16-bit instruction). I got a feeling that the GBA would need a FPU. For more information on the GBA visit this info page:

Thanks, it makes more sense to me now. Does speex_bit_read_from also append in that manner? Jean-Marc Valin wrote: > speex_bits_reset is before encoding, not decoding. If you don't call it, > then everything you encode will be appended so it will grow without end. > As for the point of the SpeexBits struct, it's there as a sort of "array > of bits" to it's easy

Greetings, It seems that the last manual and the last reference document available from speex.org web site are seriously out of date (from 2003). This makes it hard and error prone to try using any more recent features of speex in new developments that might benefit from the undocumented new features. Are there more up-to-date documents at all? And if so, could someone put them to the web site?

Hi there, here is a little bug report: # ./configure --with-gnu-ld --enable-sse # make [...] vq.c:99: error: conflicting types for `vq_nbest' vq.h:44: error: previous declaration of `vq_nbest' vq.c:133: error: conflicting types for `vq_nbest_sign' vq.h:46: error: previous declaration of `vq_nbest_sign' The --enable-sse option took this bug to the surface. The header file is

Le lundi 06 juin 2005 ? 20:19 -0500, Dragon a ?crit : > Well I've comverted to UDP but still have the tick and it can't be the > diff soundcards cause it does it on the same computer. Also the wave out > buffer keeps growing as if extra data is comming from somewhere eg: the > wave out plays just as fast as the wave in so the wave out buffer should > not

Hi to all, I developed a VoIP application using Speex. Now i want to record the conversation to a file. As i have two indipendent thread transmitting e receiving voice i whish to mix both audio streams in one stream before saving the conversation to a file. I'm using Speex in narrow band mode, 8 bits per sample, mono. i found at this link http://www.vttoth.com/digimix.htm a simple way to

Hi folks, I am new with Speex, while looking for Voice Chat in MSN/Yahoo, i cam across Speex. In case of Yahoo, i found Speex with variable bit rate. and in this case, the speex_decode function is not producing any output. Does in case of variable bit rate, something else is to be done with decoding process. Please guide me, what to do. Thanx in advance Lokesh

Hi Jean-Marc, Thanks for the quick response. Here are the numbers for our little ARM systems: CPU: ARM920T, 200Mhz Speex specs: Complexity 2, 5.95Kbps Encoding: ~30%, Decoding: ~4%, with Echo Cancellation / Pre-processor: > 85% Compilation: arm-linux-gcc -pipe -Wall -W -O2 arm-linux-g++ -pipe -fno-exceptions -fno-rtti -Wall -W -O2 Configuration: --host=arm-linux

The API reference says it is "+-2^16 range." Which is right? I've yet to successfully encode or decode anything. Jean-Marc Valin wrote: >It is documented and the range is +-32767 > > Jean-Marc > >Le mardi 07 juin 2005 ? 11:59 -0500, Matt DeMoss a ?crit : > > >>/input_frame/ is a /(float *)/ pointing to the beginning of a speech frame >>

Because a wideband signal has both a narrowband part and a wideband part the first part of the wideband decode is to call the narrowband decode. In the fixed point version the signal generated from the narrowband decode is downshifted at the end. In this same version the wideband then upshifts the signal by the same amount. I would like to do away with the downshift on the narrowband side

Hello ! I am developing a kind of phone using TI's OMAP 5912 and speex. The soft works great on ARM, and it is time to move speex and voice processing stuff to DSP. Could some one point me how to do it ? Should I install CCS ? Or it possible to compile speex using just ti_dsptools ? Does anybody ran speex and dsp gateway ? It would be great, if some one share makefile, cmd etc. for

Hi there is a small bug in the sample code 'sampledec.c' here: http://www.speex.org/manual2/node12.html#SECTION000121000000000000000 it is trying to destroy the encoder of the "decoder" state, this patch should fix it: Index: doc/sampledec.c =================================================================== --- doc/sampledec.c (revision 11555) +++ doc/sampledec.c