similar to: HW-Speex: Fileformat and encoding process

Hello, I would like to run a script over a speex file which determines, how long the speech contained in the speex file is. So I run through the ogg file, skip the first two ogg _pages_ (speex header and comment header, both contain IMHO only 1 ogg _packet_), and then count in any ogg page the number of the ogg _frames_ (which, unless I used -nframes while encoding, which I did not), and multiply

On Tue, May 16, 2006 at 03:29:47PM +0200, Bj?rn Thalheim wrote: > > I would like to run a script over a speex file which determines, how > long the speech contained in the speex file is. > So I run through the ogg file, skip the first two ogg _pages_ (speex > header and comment header, both contain IMHO only 1 ogg _packet_), and > then count in any ogg page the number of the ogg

Dear Jean-Marc and Peter: Thank you both very much for your time and advice. I did not realize that Lame MP3 code has a -r option (without reading its code). I have tried the suggested command lines verbatim with the added -r option, along with other combination of option settings. I could not figure out how to eliminate the distortion in the result, like voice turning either high or low

On Fri, Apr 4, 2008 at 12:19 AM, Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > Jahn, Ray (R.) a ?crit : > > > Dear Speex codec community: > > > > I am working on conversion of voice files. I could not figure out how to use speexdec.exe 1.2.3 in piped mode in order to avoid the creation of the potentially large intermediate *.wav or *.pcm files. Any

Hi, Sorry if my question not related to speex. I have created a voip application using speex. My internet line is 64Kbps. I`m using narrow band mode, 8000 bitrate, 20ms of 8000Khz sampling rate, no buffering, no preprocessing, just set bitrate and go. When i send single frame inside rtp the sound was not good, choppy and noisy. But when i pack around 8 frame inside rtp packet, the sound was

hi, I noticed the build-nsdzones.pl only converts the first zone of my named.conf to nsd.zones. It also did not include either a master or notify keyword, even though my bind configuration for that zone had an also-notify keyword. Even not listing an also-notify line did not cause build-nsdzones.pl to process more then one zone. Second, I had a question about the nsd.zones format. Does the

> There are a usually lot more data packets than header packets, so > having an internal length there hurts your bitrate a lot more. Of > course, it may not be significant for an uncompressed text codec. If muxed with a video, it's insignificant (well, for my test cases). > With codecs using the 'count of decodable samples' rule to calculate > their granulepos, the

Hi Bob! I could not find llvm file for ARM target in llvm-gcc 4.2 front end source code. $llvm-gcc-src/gcc/config.gcc file alpha*-*-*) cpu_type=alpha need_64bit_hwint=yes # LLVM LOCAL begin out_cxx_file=alpha/llvm-alpha.cpp # LLVM LOCAL end ;; ... arm*-*-*) cpu_type=arm extra_headers="mmintrin.h" ;; ... i[34567]86-*-*) cpu_type=i386 # LLVM LOCAL begin

Okay, I feel really stupid here but I've been unable to figure out how to easily seek within a speex file so that playback starts from the beginning. I've scoured the mailing lists for solutions and the best that I could come up with was to see what libvorbis was doing when it performed a seek inside of an ogg stream but it looked rather complicated for what I want to do. Does

On 14/02/2008, ogg.k.ogg.k@googlemail.com <ogg.k.ogg.k@googlemail.com> wrote: > Ralph wrote: > > Conrad, does this work for the other codecs as well? I vaguely > > remember libvorbis putting -1 on the header packets instead of 0. > > AFAIK, it's 0, -1 being for partial pages, from the ogg docs. -1 (or, I > suppose all negative values, but I'm not sure) mean

There is a bug in "speex_resampler_process_interleaved_int" and "speex_resampler_process_interleaved_float" in the resampler. It seems that when "speex_resampler_process_int" is called for the first channel, "in_len" is set to the actual number of samples used (when output limited), which you would think would be OK to use for all the subsequent calls

From: Jyri Sarha <jsarha at ti.com> This bug happens quite often when resampling from a low to a high sample-rate with big enough factor. Also the resampling call has to be limited by the output buffer size and some unused samples needs be left in the input buffer. Sometimes when up-sampling with a big factor the resampling function wants to peek one more sample from the input buffer to

Hi Gurus, I''m learning the Rails framework, and I have this exact block of html/javascript in two places in an if/else construct in application.html.erb: <body onload="Modalbox.show(''<div class=\''warning\''> <p> <center> <%= flash[ :notice ] %> </center> </p> <p> <center> <input

That is from 2.5, and just because there is nothing special listed in config.gcc does not mean it doesn't work. For 2.5, the ARM port of llvm-gcc did not require a separate llvm-arm.cpp source file, so nothing needed to be added to config.gcc. It worked fine as far as I know. For 2.6, you will see that there are some ARM-related changes to config.gcc in llvm-gcc. On Sep 30, 2009,

.. And a version without the API changes. -------------- next part -------------- Index: libspeex/resample_sse.h =================================================================== --- libspeex/resample_sse.h (revision 0) +++ libspeex/resample_sse.h (revision 0) @@ -0,0 +1,128 @@ +/* Copyright (C) 2002-2008 Jean-Marc Valin + * Copyright (C) 2008 Thorvald Natvig + */ +/** + @file resample_sse.h +

Hi, Here's the (hopefully) final version of the resampler, now always using st->mem as the buffer area. It only allocates buffers on the stack when it's necesarry to convert the output between int and float. -------------- next part -------------- Index: include/speex/speex_resampler.h =================================================================== ---

Fixes issue #95 on GitHub. --- src/opus_multistream_encoder.c | 57 ++++------------------------------ 1 file changed, 6 insertions(+), 51 deletions(-) diff --git a/src/opus_multistream_encoder.c b/src/opus_multistream_encoder.c index 6cc1f432..9cb9bf34 100644 --- a/src/opus_multistream_encoder.c +++ b/src/opus_multistream_encoder.c @@ -742,20 +742,9 @@ static void ambisonics_rate_allocation( {

Hi List, Now I will send to you more specific what I am trying to do. I have one Asterisk Channel where receives Midia Frames in the codecs format: Speex UltraWideBand and Speex NarrowBand. When I use Speex NarrowBand the Asterisk is able to convert this frame to G711. But when I use Speex UltraWideBand the Asterisk don't convert it. Then I need in my Asterisk Channel Source include the Speex

Hi, I've attached a patch against SVN r14645 which adds GCC visibility information to all symbols exported from libspeex.so and libspeexdsp.so. It includes a configure.ac change to test that both the compiler flags and __attribute__((visibility)) works, and if so will #define EXPORT __attribute__((visibility("default"))) and if not #define EXPORT I've attached a diff output

Hi, I've attached a patch against SVN r14645 which adds GCC visibility information to all symbols exported from libspeex.so and libspeexdsp.so. It includes a configure.ac change to test that both the compiler flags and __attribute__((visibility)) works, and if so will #define EXPORT __attribute__((visibility("default"))) and if not #define EXPORT I've attached a diff output