similar to: Conversion API for computer telephony systems (Dialogic Mu-law wa v format to PCM encoded wav format)

Thanks. sox was exactly what I was looking for. -----Original Message----- From: volsung@asu.edu [mailto:volsung@asu.edu] Sent: Thursday, February 08, 2001 2:06 PM To: vorbis-dev@xiph.org Subject: Re: [vorbis-dev] Conversion API for computer telephony systems (Dialogic Mu-law wa v format to PCM encoded wav format) On Thu, 8 Feb 2001, Giovanni Sanfelici wrote: > I am working on a project

Because u-law and a-law are non-linear algorithms and the mathematics of FLAC (and AFAIK all lossless encoders) are build for linear PCM. Adding these formats would change the FLAC format altogether, decoders are not made to work with it (they can only output plain PCM) Best option would be converting these files to 16-bit plain WAV and compressing, but I guess files won't be much smaller

On Nov 19, 2011, at 16:42, Giulio Paci wrote: > So the problem would be suboptimal compression due to suboptimal > assumption about the input signal, right? The problem is more that FLAC should not be a collection of code to read every possible file format in existence. That would be a duplication of at least two other audio file format conversion utilities, and quite bug ridden,

Thank you all for your answers. They were all useful. Il 21/11/2011 07:37, Erik de Castro Lopo ha scritto: > Giulio Paci wrote: > >> thank you for your answer. So the problem would be suboptimal >> compression due to suboptimal assumption about the input signal, right? >> What I do not understand is how the format of a FLAC format would be >> affected by supporting

Hi to all! I have a database of audio files that I want to losslessly compress. Unfortunately I have several 8bit A-law and mu-law files in the database and I see from here http://flac.sourceforge.net/documentation_tools_flac.html that they are not supported by the reference flac encoder/decoder. Is there a reason for this? Would it be possible to add support for these files in the reference

Hi, sndfile-convert already converts from all these formats to FLAC, but the flac tool itself has more flac-specific options. Is it possible to use sndfile-convert to provide the input data? In any case Erik is maintaining both libsndfile and libflac, and it's unlikely he'd want to duplicate the code. If anything it'd make more sense to remove code for reading other formats from the

Hi Martijn, thank you for your answer. So the problem would be suboptimal compression due to suboptimal assumption about the input signal, right? What I do not understand is how the format of a FLAC format would be affected by supporting A-law and mu-law files as input (and thus output). Despite of suboptimal performance, is it possible to treat 8bit *-law samples as 8bit linear PCM files and

Speex 1.0.3 uses floats, but the unstable version 1.1.x uses short. Jean-Marc Le ven 16/04/2004 16:04, Kenji Chan a crit : > Im confused about the format of the PCM/WAV data for speex_encode & > speex_decode > > speex_encode(enc_state, input_frame, &bits); > > In the manual, it says input_frame is a (float *) pointing to the > beginning of a speech

Le ven 16/04/2004 à 16:39, Kenji Chan a écrit : > Ok, I'm using 1.1.4, so I should pass signed short(wav data) directly to > speex_encode()? First, I'd like to ak why you want to use 1.1.4. If you don't need the new features (fixed-point or preprocessor), you should probably stick to 1.0.3, which has received more testing. If you choose to use 1.1.4, you need to send shorts to

> You said, it's unstable, yes, I got compiling errors with MsVC6 and MsVC.net > And I modified it a bit to get it compiled. Should I post the errors > here? YES, kindly post the errors that you encountered. - Abhishek <p><p><p>---------- Original Message ----------- From: "Kenji Chan" <adslbqmr@tpg.com.au> To: <speex-dev@xiph.org> Sent:

I am having trouble converting a .flv file uploaded from Adobe Flex / Flash Player 10 to a Red5 server using the speex coder: http://livedocs.adobe.com/flex/3/langref/flash/media/Microphone.html http://jira.red5.org/confluence/display/codecs/Speex+Codec Questions: 1. How do I extract the audio track out of such a .flv file? 2. How do I convert it from Speex to .wav PCM? Thanks.

Hello everyone, I tried to compress audio with opus-1.3.1/src/opus_demo.c recently, which works fine on mono and stereo data . Now I want to compress interleaved 7 channels pcm/wav ( recorded by Microphone array :6mic+ 1reference signal ) with opus, But I have not found an interface that compress multi-channels pcm/wav. 1、Is there a multi-channel compression interface can be used in my case? If

Ok, I'm using 1.1.4, so I should pass signed short(wav data) directly to speex_encode()? But the samplecode that comes with 1.1.4 shows me to put short in float array, and pass the float array to speex_encode() <p>-----Original Message----- From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org] On Behalf Of Jean-Marc Valin Sent: Saturday, April 17, 2004 6:25 AM To: speex

I'm confused about the format of the PCM/WAV data for speex_encode & speex_decode speex_encode(enc_state, input_frame, &bits); In the manual, it says "input_frame is a (float *) pointing to the beginning of a speech frame" (for encode) But I've found that in speexenc.c and testenc.c, short* is used instead of float* So, isn't it signed 16 bit samples(if

I thought it would be more update/better/faster, any thing wrong? In the future, would you use short, and forget about float? If so, I use 1.1.4, I wouldn't need to change my code in the future, right? You said, it's unstable, yes, I got compiling errors with MsVC6 and MsVC.net And I modified it a bit to get it compiled. Should I post the errors here? <p>-----Original Message-----

Hi All, I've had a great success these last couple of days on bringing up Asterisk with a load of Cisco 7940 phones and a couple of soft phones as well. I'd really like to try and link this in to my companies PBX so I can call in from the PSTN as well. I understand there is a charge for the module that would allow me to use my old Dialogic board, can anyway provide some more information

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

This is resolved: apt-get remove libspeex-dev cd ~/src/speex-1.2rc1/ ./configure --prefix=/usr make; make install cd ../ffmpeg ./configure --enable-libspeex make; make install worked; then I was able to decode a Speex .flv file: ~/flvs$ ffmpeg -i SpeexQ6R16Efalse.flv foo.wav FFmpeg version SVN-r17174, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-libspeex

Hi, How can I create a web page allowing people to listen (with their own PC) a couple of .wav/a-law files stored on a Linux server ? Chances are users would access this web page from Internet Explorer but if I could make it available to other browsers, that would be better. I googled a bit and couldn't find a tag such as media://myaudiofile.wav that would fulfill this spec. As much as

Spanish/Espa?ol: Hola a Tod at s, Estoy en la implementaci?n de una soluci?n a medida en la cual requerimos conectarnos v?a SIP a un IMG de DIalogic y salir hacia otras centrales (MSC) v?a SS7, la conexi?n entre Asterisk y el IMG es SIP, pero como requerimiento necesito enviar un NOA (Nature of Address) con valor 8 por requerimientos normativos, hasta ahora no he podido encontrar como puedo