similar to: Default endian for signed linear

Does anyone know if there is a "format-raw.c" routine available for Asterisk-0.9.0? Jim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041211/c7ffe360/attachment.htm

Hello Martin, > how a number of samples with no defined size is translated into 8, 16, or 32 bit values in my buffer Although I do not know in detail about the actual implementation of libvorbis, most of the decoders should internally produce single-precision (32bit) floating point values, or 32bit integers as the result of PCM decoding. They are downcast to PCM samples with 8bit, 16bit, or

Dear All, I am using a specific approach for my master thesis. In essence, a supervised reclassification is used as an intermediate step to find chemical parameters which are able to reclassify defined groups. These variables will be used in a next step where location and scale estimators of the groups are important. Traditionally linear discriminant analysis is used for reclassification which

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Hi, I'm doing a VoIP related project where at some point i need to decode speex data gathered from a flash client, and then encode it to ulaw or alaw pcm so i can send it in rtp packets. On the other way, i get rtp packets with ulaw and alaw pcm data, then i need to decode them to linear pcm and feed the speex encoder, so i can send the data to the flash player. I'm using JSpeex, i know

Hi folks, I'm doing a merge of my current branch onto the mainline (for testing) today. I believe it to be stable. Just a little more vorbisfile testing. After the merge, I have a few more patches to apply, then onto cascading/coupling. New stuff: Floor backend 1 and residue backend 1; both are present, but the mainline modes won't use either yet. Naturally, both are enabled for

If I use Speex, JSpeex actually, to compress an otherwise valid wave file with zero lengths in the header would it impact the compression at all? Here's what I'm doing during compression in Java: AudioFormat wavFormat = ais.getFormat(); AudioFormat speexFormat = new AudioFormat(SpeexEncoding.SPEEX_Q5, wavFormat.getSampleRate(),

Using Linux (Kubuntu) using the application "crip" ripping at -q 10 & using replaygain on for volume control -------------------------------------------- Now that I have 2000 .ogg files... first problem: There is NOT one Portable CD Player on the Market, that can play encoded .ogg files AND read the replaygain tags. via a cd-r second problem: Burning... if I am going to burn a

Flash player encodes speex at 16 kHz, mono, 16 bit. Fields in the format byte should be ignored if the format is speex. You can set the quality by Microphone.encodeQuality (default 6). You can also set the number of speex frames per tc message using Microphone.framesPerPacket. Flash player can only decode speex at 16 kHz, so make sure you have the proper sample rate. Jozsef > > Message:

Hello Mark, The resulting 8 bit file has a lot of squelching noise compared to the 16 bit output from OPUS decoder. During encode I am using popi16fmtBuffer[ui32Loop] = (opus_int16)pcRdBuf[ui32Loop]; And during decode since the data is in the lower 8 bit I use pc8bitSamples[ui32Loop] = ((unsigned short)pcop16OutBuf[ui32Loop] ^ 0x80); Regards Amit On Sat, Jan 9, 2016 at 2:39 PM, Amit Ashara

Hello Mark, Thanks. Let me try the proposed configuration first to make sure that linear 8-bit PCM is retrieved. Regards Amit On Sat, Jan 9, 2016 at 12:28 PM, Mark Harris <mark.hsj at gmail.com> wrote: > opus_decode() produces 16-bit signed linear PCM, and > opus_decode_float() produces 32-bit floating point PCM that is useful > when you want a higher bit depth. > > If you

Hello All, Turned out to be a coding error. The modified buffer was not being allocated to write back to the File System. After correction the mono 8 and 16 bit for 8K-48K works well now. Regards Amit On Mon, Jan 11, 2016 at 11:05 AM, Amit Ashara <ashara.amit at gmail.com> wrote: > Hello Mark, > > The resulting 8 bit file has a lot of squelching noise compared to the 16 > bit

opus_decode() produces 16-bit signed linear PCM, and opus_decode_float() produces 32-bit floating point PCM that is useful when you want a higher bit depth. If you need 8-bit linear PCM then a simple solution would be to use only the top 8 bits of each 16-bit sample from opus_decode(). Note that the WAV format uses unsigned rather than signed integers for 8-bit linear PCM. (It uses signed for

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,

Hello Benjamin, The original WAV file I have is linear 8-bit PCM. I want to ensure that original linear formats are kept as is. Later I will add support for ulaw. Regards Amit On Fri, Jan 8, 2016 at 5:34 PM, Benjamin Schwartz < benjamin.m.schwartz at gmail.com> wrote: > Do you really need linear 8-bit PCM or do you want ulaw? Linear 8-bit is > ... pretty rare. > > On Thu,

On Oct 21, 2008, at 2:27 AM, Jay Foad wrote: > Hi, > > I'd like to use LLVM to compile and optimise code when I don't know > whether the target CPU is big- or little-endian. This would allow me > to create a single optimised LLVM bitcode binary of an application, > and then run it through a JIT compiler on systems of differening > endianness. Ok. > I realise that

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, I'd like to use LLVM to compile and optimise code when I don't know whether the target CPU is big- or little-endian. This would allow me to create a single optimised LLVM bitcode binary of an application, and then run it through a JIT compiler on systems of differening endianness. I realise that in general the LLVM IR depends on various characteristics of the target; I'd just

Sorry -- I goofed on the sample rates! Apologies! Hello, everybody: I'm developing an application using Asterisk and a TDM-400 card. I understand the concept of the difference between GSM and WAV files when using Asterisk, but I'm not happy with the sound quality with the GSM compression. It's merely *acceptable* for a telephone call, but for anything else, it leaves something to be

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