similar to: 11025kHz and framesizes question

hi, i am using AVTransmit3.java for transmitting audio, how can i implement JSpeex or speex for compression and encoding. please give me some idea about this. shankar.s speex-dev-request@xiph.org wrote: Send Speex-dev mailing list submissions to speex-dev@xiph.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.xiph.org/mailman/listinfo/speex-dev or, via email, send a

Hi I am currently trying to port speex v1.1.6 to a microcontroller with very limited memory (<64Kbyte RAM). what I found when initialising the encoder, a chunk of 32Kb was attempted to be alloced, which failed: src/nb_celp.c: void *nb_encoder_init(const SpeexMode *m) { /* snip */ st = (EncState*)speex_alloc(sizeof(EncState)+8000*sizeof(spx_sig_t)); /* snip */ } same goes for the

Sorry. First error file is not "modes.c" but "modes_noglobals.c". ________________________________ From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of ¼Õ½Â¿ø Sent: Friday, January 20, 2006 11:40 AM To: speex-dev@xiph.org Subject: [Speex-dev] Compile error (svn 10743) Hi, I've received svn 10743 revision. But there are compile errors at

Hi, I've received svn 10743 revision. But there are compile errors at three files with VS2005. Please check these reports. Sohn, Platform system engineer. ============================================================================= 1. modes.c - nb_mode_new() SpeexNBMode * nb_mode; nb_mode = (SpeexNBMode *) speex_alloc (sizeof (SpeexNBMode)); if (nb_mode == NULL) return NULL;

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

You're asking the wrong question. The question is not "why does it would bad with Speex?", but "why does it sound good with LPC10 and MELP?". And the answer is that both are vocoders. Try dropping frames/subframes with anything else (Vorbis, MP3, G.729, u-law, ...) and it'll sound terrible as well. The only reason it sounds good with vocoders is because the

Here's one clue about whatever is causing the low quality speech. Speech sounds terrible at 1.01X faster, and it sounds excellent at normal speed (1.0X). So, the main problem is something that breaks with any change in frame size in the decoder. Any idea what that might be? Thanks, Bill On Tue, Oct 19, 2010 at 5:14 PM, Bill Cox <waywardgeek at gmail.com> wrote: > I was able to

Thanks Jean-Marc. I do want to change Speex frame size to 320. In this case, I've changed frame size in init and modes (include framesize and subframesize). However, the audio quality becomes bad. It seems something else needs to be changed as well. Could you give a clue? Thanks, Dennis -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent:

Hi. I'm Bill Cox, and I volunteer a bit for the Vinux project, which is Linux for people with vision impairments. Most blind users use a closed-source speech synthesis tool called voxin, as it's very easy to understand at high speed. I would like to make TTS synthesizers based on large recorded vocabularies of actual speech, but to make it useful for the blind, I need to be able to

This is going to take some explaining and I apologize in advance if any of this is found in the manual or sample code but I couldn't find it. I just graduated last may and this is my first experience with vocoders and dissecting a professional's code. I work for a company that is currently using a G729A vocoder from a 3rd party software company and is looking into speex so they no

Hi Alfred, First thing I'd like to ask is whether you're trying to achieve real-time performance, as I find it very unlikely that Speex could run in real-time on a micro-controller. Regarding bufSize, no you can't just make it the size you like. Now, it's possible to decrease the amount of memory allocated if e.g. you only use complexity 1. You can also assume that the stack is

Hi there. I am having a little trouble understanding the frame sizes chosen by the codec. testenc_uwb.c from the speex-1.0 source distribution has a framesize of 640 hardcoded and makes use of this value exclusively. However, a mode query on the actual codec returns 320 as a framesize for this mode. int tmp; speex_mode_query(&speex_uwb_mode, SPEEX_MODE_FRAME_SIZE, &tmp);

Hi, I am using speex 1.2beta2 on a 24bit DSP that has a severe program and data space limitations. I am only interested in the speex decoder for 11kbps narrowband implementation. I am using the following parameters and structures 160, /*frameSize*/ 40, /*subframeSize*/ 10, /*lpcSize*/ 17, /*pitchStart*/ 144, /*pitchEnd*/ /* 11 kbps medium bit-rate

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms

Hi, Steve. I tried your the time_scale_tests program, and it works well! Especially for low speed changes, it's the best I've heard so far. For high speed increases, there is what sounds like static added to the sound output. I've attached two sound samples of high speed speech, which is a 4X speed up of a popular TTS voice in the blind community (voxin/Eloquence). I've sped

To close debian bug #271052, http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=271052 I went ahead and did the following: $ diff -u ../original/speex-1.1.6/libspeex/nb_celp.c libspeex/nb_celp.c --- ../original/speex-1.1.6/libspeex/nb_celp.c 2004-07-15 01:16:52.000000000 -0400 +++ libspeex/nb_celp.c 2005-02-27 08:24:49.000000000 -0500 @@ -1746,6 +1746,21 @@ st->submodeSelect =

Hi all, I have a problem when I am encoding (or decoding) stereo audio. With mono data, things are fine and everything works without any problems. When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong? Below is the code, first, of the encoder and next that of the decoder. Any information or

Hi, I've been trying to encode a live audio input from the microphone on iOS device using opus. Uncompressed audio recording works fine with http://theamazingaudioengine.com/ Then, when I tried to do encoding, I'm stuck at figuring out why the buffer is empty: static int ec_write_byte(ec_enc *_this,unsigned _value){ if(_this->offs+_this->end_offs>=_this->storage)return

I didn't expect this, is it normal? When you use inbandfec, and have packet losses, you end up with more audio samples than you started with. With short frame (2.5), the FEC isn't supposed to do anything but it shouldn't do that should it? and 20mS frame, should do 'something' but? add audio? I'm using opus_demo for this. Here's the steps (I have my test audio in

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear