similar to: Compression for audiobooks

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

Hi every one, I know that speex suports Symbian now in the unstable release but I wounder if any one can compile a decoder for Symbian MMF? I have googled for one but ending up with only "yes, speex suports Symbina" and no binary. I would use it to finaly listen on my audiobooks but I think it would be a good demonstration aplication. Best Regards P?r Moberg

Hi, Up until now, I've been providing narrowband streams at 24kbps MP3. We've reluctantly been using MP3 instead of say Ogg Vorbis due to wider support. But it's getting to the point where low bitrate MP3 doesn't really cut it anymore. With technologies like AACPlus out there in common usage, people have come to expect better and one can hardly blame them. So I thought

Hi, I am developing a voip phone that uses speex v1.2b1 narrowband mode. Libspeex decodes the 21-byte packets I recieve into 160 bytes and I can play them fine, but when I encode, the result is 38-bytes long. I have read that 38 bytes means that it is encoding at 15KHz, but I have set it to nb-mode and when stepping through the libspeex encoding sourcecode, I see that st->sampling_rate is

just to check that I've got everything right. the encoder allows as input - mono only - 8 or 16 bits as floating point numbers (without scaling to 1.0). floating-point wavs (IEEE) will also work, but it's better to scale them to something like 8000. - any sample rate (should be set with speex_encoder_ctl), but prefered are 8/16/32 kHz. but what modes should I use for a given rate? also -

Hello, thank you for responding. I'm sure I'm confusing something, but I did not explicitly set either kHz or kbps, I just set the mode to narrowband, and traced through libspeex's code to see that sample_rate was 8000 something, I'm guessing Hz. My assumption was: if you set PA to nb-mode, then you should encode 160 bytes to 21. Is that mistaken? ys On 6/13/07, Jean-Marc Valin

Hi all, One question that I hope someone on the list just knows the answer to without having to delve too deeply into the code: How does UWB mode divvy up the bandwidth and pack it in the bitstream? I know from the documentation that WB mode codes the first 0-4K kHz band as a Narrowband packet, and then adds on the 4-8 kHz band coded separately (so that a NB decoder can decode a WB bitstream

Also, you would be sattisfied with the narrowband decoding because the GBA sound quality isn't very good, however, that depends greatly on what you're going to do with speex on a GBA.... (an RPG with audio dialogs? ;-)) also, using a DSP in a GBA Cartridge sounds like overkill to me, however, that depends greatly on what are you developing and what's your target audience....

Hi, I need to get an overview over the memory/speed requirements for speex. We want to investigate whether we can implement on a target processor with limited resources. The target processor is a 16 bit fixed pint DSP with a single ALU and MAC. It runs at 60 MHz and has a total of 16 Kbytes RAM from which other processing also will eat. What we would like to implement is a speex encoder and

Le mar 18/11/2003 à 17:39, Allen Drennan a écrit : > Speaking of video conferencing in conjunction with Speex, we are > currently beta testing a solution we developed that offers multi-point, > multi-party video and audio using the Speex engine for voice. > > http://www.wiredred.com/downloads/ecsetup.exe > > The fair and good audio settings are Speex narrowband, high quality

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

Hello Trying to setup dovecot installation with sieve plugin, can't configure how to make redirect working. My setup 1 Exim + dovecot lda begin transports dovecot_virtual_delivery: driver = pipe command = /usr/local/dovecot/libexec/dovecot/deliver -d $local_part message_prefix = message_suffix = delivery_date_add envelope_to_add return_path_add log_output user = exim 2

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

Hello, congratulations to the Ogg Vorbis team - RC2 sounds good. But... RC2 in 128 kbps mode seems to clip off all frequencys beyond 16 KHz. On the tracks I tested Beta 4 gave response even beyond 18 KHz. Some testings on a randomly chosen track: (other tracks gave similar results) Artist: Judas Priest Album: Jugulator Title: Bullet Train Beta4: 127 kbps, ~ 18 KHz (!) RC2: 132 kbps (!), ~ 16

FYI, the Asterisk software PBX <http://www.asterisk.org/> has now incorporated my recent patches to support dynamic RTP payload types. As a consequence, its SIP implementation now supports Speex, so if you have a Speex-compatible SIP client, you can use it to make calls using Asterisk. Some caveats: - Only narrowband (8 kHz) Speex is currently supported; not wideband. (Unfortunately,

Hi, Just to let you know that Speex 1.0 RC3 is out. It should be the last RC until 1.0 is released (tentative release date is March 16th). Please give RC3 a try and report bugs so they can be fixed before 1.0. As usual, download from http://www.speex.org/download.html Jean-Marc -- Jean-Marc Valin, M.Sc.A. LABORIUS (http://www.gel.usherb.ca/laborius) Université de Sherbrooke, Québec,

First let me introduce myself: Sr Computer Scientist 20+ yrs SW/HW development Audio Rookie < 60 days Intel based computer Soundblaster 128 PCI Linux Fedora Core 2 ALSA - Advanced Linux Sound Architecture ALSA provides the audio and MIDI functionality to the Linux operating system as of Fedora Core 2 and is the future of Linux (so I have been told) http://www.alsa-project.org/ I have read

Hi I've been using Samba from pkgsrc successfully on Illumos, SmartOS specifically until I moved to 4.17.11. When I try to access my home folder on Linux I get, chdir_current_service: vfs_ChDir(/home/fukr/kev) failed: Permission denied. Current token: uid=10000, gid=10001, 3 groups: 10001 10000 10002 My home folder permissions are 700. If I change to 755 it works fine. My the uid, gid are,

> Page 3: > > To be compliant with this specification, implementations MUST support > 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. > The sampling rate MUST be 8, 16 or 32 kHz. > > There is a type above after (narrowband), there is a " extra character. > > I don't understand what is the motivation to specify "SHOULD

Narrowband (8 kHz), wideband (16 kHz), and ultra-wideband (32 kHz) compression in the same bitstream What is the meaning of this? can i find more information about this somewhere? And what is packel loss concealment, an you enable this somewhere? thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: