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

I am working on a project involving the conversion of a Mu-law sound format (Dialogic Mu-law wav format) file into a standard PCM encoded wav file format. Could somebody tell me if this feature is supported in the Vorbis software and if there is any source code available that performs this task. If not, does anybody know of any resources that might provide me with this tools or information.

Okay, before I submit my patch to make libao produce sorta-streamable wav files, I want to know what these partial wav files do to various players. I've posted two sample wav files on my webserver: http://volsung.dhcp.asu.edu/~stan/infinite.wav http://volsung.dhcp.asu.edu/~stan/zero.wav The first uses a riff and data length of 0xFFFFFFFF (approximately infinite) and second uses a riff and

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-----

Dude if it is a mail server that people from Getronics put together its no wonder its broken ;-)) Kind regards, Mark -- Mark de Bokx Internet Service Engineer ICT Communication and Media Services Planet Media Group the Netherlands "This is not an add banner, click here to find out why" -- seen on the Web -----Original Message----- From: volsung@asu.edu [mailto:volsung@asu.edu]

------- Forwarded Message Return-Path: owner-vorbis-dev@xiph.org Delivery-Date: Mon Jul 10 01:38:16 2000 Delivered-To: xiphmont@localhost.xiph.org Received: from localhost (localhost [127.0.0.1]) by bloopfish.xiph.org (Postfix) with ESMTP id 1A4A48882F for <xiphmont@localhost>; Mon, 10 Jul 2000 01:38:15 -0700 (PDT) Delivered-To: xiphmont@xiph.org Received: from xiph.org

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