similar to: Speex - Real time??

Thanks for the advice. With complexity=1, bit rates 5950,8000 and 15000 (CBR only), I'm still getting 30-50 seconds encoding time for a 10-second file. To anyone who has made this work in ARMv4, or knows how to, can I get some advice on the settings? FIXED_POINT is already defined in all relevant files. My source code is patterned after sampleenc and I haven't tried using Ogg.. my source

Hi, I'm using speex 1.1.10 and writing a program similar to sampleenc. I need to use the fixed point library as I'm eventually porting to ARM processor but I encountered a problem with speex_encode_int(), which I used in place of speex_encode(). I gave it a short instead of a float (manual says it takes a short for the audio - 2nd parameter) but the compiler says it could not convert the

Hi,I'm having performance issues encoding with speex on an iphone4 + iOS5. The same code yields excellent results on iPad2 + iOS5, Mac OS X 10.6 as well as Windows 7. The call to speex_encode_int is takes 0.0065 to complete. Because encoding is so latent the receiver incurs underruns. Here is my speex setup: ? ? unsigned val = 0; ? ? /**? ? ?* Initialize the encoder in narrow band mode.? ?

hy, here is my speex encoder/decoder .. the sampleRate i use is 16000 and quality,complexity are at 5. can someone take a look in to the code and see if there is something that is making that robot voice here is a link to colored and numbered code, same as below http://barvanjekode.gama.us/temp/1257361243.html thanx. code ----------------------- #include "codec.h"

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

hy, recording and playback is working perfectly without speex. i have try to set samplerat from 6000 to 441000 and quality from 1 to 10 sam with complexy, but the best i can get is with 16000 samplerate, 5quality and 3complexy .. but still, the voice that came out is annoying, artificial, robot ,... Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

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, I'm using the latest version of libspeex (checked out today) and have some problems with checking the target bitrate in the encoder. Calling: peex_encoder_ctl(globals->enc_state,SPEEX_GET_BITRATE, &bitrate); Seems to set bitrate to 24600 no matter what the qualitysetting is.. The functions used previous to this call is: peex_bits_init(&globals->bits);

Hi, I'm trying to use Speex for real time encoding in an iPhone project but I'm having latency problems. I appreciate any tips you may provide for speeding things up. The recording callback is providing 1024 samples (mono, frequency: 44100 Hz, of type short) every 23 ms (1024.0 / 44100.0). I'm saving the samples in a buffer and use them in the encoding thread (set for mode: wide

Jonathan Lennox wrote: > -# if defined(FIXED_POINT) > +# if defined(FIXED_POINT) && \ > + ((defined(OPUS_ARM_MAY_HAVE_NEON) && !defined(OPUS_ARM_PRESUME_NEON)) || \ > + (defined(OPUS_ARM_MAY_HAVE_MEDIA) && !defined(OPUS_ARM_PRESUME_MEDIA)) || \ > + (defined(OPUS_ARM_MAY_HAVE_EDSP) && !defined(OPUS_ARM_PRESUME_EDSP))) > opus_val32 (*const

Please compare samples: http://www.ivo.pl/ivo2/sound/krwawa_org.wav http://www.ivo.pl/ivo2/sound/krwawa_fixed.wav http://www.ivo.pl/ivo2/sound/krwawa_float.wav krwawa_org.wav is original file. Both output files was created using same program by compressing from wav to speex, then decompressing from speex to wav (using speex_encode_int and speex_decode_int). Only difference was used speex

I bought a ADSP-BF533-EZkit-LITE V2.1 to develop a speech codec application using SPEEX. I have tested SpeexEcho project from both BlackfinSDK-Rel201 and BlackfinSDK-Rel310. When disabling SPEEX processing, i.e. using PASS-THROUGH mode, the output voice is ok, which is exactly as the input voice. However, when enabling SPEEX processing, the output is bad, no matter how to set the parameters in

Hi, I encoded file in a ipaq 1940, pocket pc 2003, processor Samsung s3c241(similar to ARM), 266 Mhz. I would like to know your opinion for the quality output if this is right to entered parameters. File original: http://padawan.ing.puc.cl/chavezorig.wav File encoded: http://padawan.ing.puc.cl/chavezq1.wav The parameters for encoded are: state = speex_encoder_init(&speex_nb_mode); int tmp=0;

--- celt/arm/arm_celt_map.c | 24 +++++++++++- celt/arm/pitch_arm.h | 97 +++++++++++++++++++++++++++++++++---------------- 2 files changed, 88 insertions(+), 33 deletions(-) diff --git a/celt/arm/arm_celt_map.c b/celt/arm/arm_celt_map.c index 0c9acff..cc6b706 100644 --- a/celt/arm/arm_celt_map.c +++ b/celt/arm/arm_celt_map.c @@ -94,9 +94,14 @@ void (*const

this is in reply to a different message i posted a couple days ago , im not sure how to specifically reply to it sorry :( heres the message: thank you very much vinod this does give me a nicer idea :) but i still have a couple questions well firstly, speex_bits_write is still blocking forever, i can never seem to get past that line (i am using the armv4 build on ppc2002, but i havent found

Uses NEON optimized fixed point fft routines in NE10 library Signed-off-by: Viswanath Puttagunta <viswanath.puttagunta at linaro.org> Signed-off-by: Jonathan Lennox <jonathan at vidyo.com> --- Makefile.am | 12 +- celt/arm/arm_celt_map.c | 46 ++-- celt/arm/celt_ne10_fft.c | 98 +++++---- celt/arm/fft_arm.h |

Hi, I have a module sampling raw PCM data on Win CE as 10ms time slice (160 bytes), mono, 8000HZ, 16 bits per sample. Does anyone know what is the mflops for using fixed point on a Win CE compared to using floating point? Looking at the manual, "In practice, frame_size will correspond to 20 ms when using 8, 16, or 32 kHz sampling rate." for a 8 kHz sampling, the framesize should be

Hi, I am getting this: ----- armv4-unknown-linux-gnu-ld -o tests/fcntl.shared -e main interp.o tests/fcntl.o -R libc.so /usr/lib/gcc-lib/armv4-unknown-linux-gnu/3.3.5/libgcc.a armv4-unknown-linux-gnu-ld: internal error: aborting at /space/var/tmp/portage/binutils-2.15.90.0.3-r4/work/binutils-2.15.90.0.3/ld/ldlang.c line 3835 in lang_place_orphans armv4-unknown-linux-gnu-ld: please report this