similar to: question about coding

Thanks for the answer. So data are stored in little-endian format on the WAVE file. But have I to encode in big endian format or little endian is ok?! Thanks ---------- Original Message ---------------------------------- From: "John Miles" <jmiles@pop.net> Date: Tue, 23 May 2006 12:15:05 -0700 >You are probably reading little-endian samples on a big-endian platform, and

I'm sorry, but I've found in the paper "Speex: A Free Codec For Free Speech" at page 5 the phrase "If more than one channel is used, then an instance is required for each channel (Speex is not stateless)". What does it mean? Could it be that is the problem that I've met? Thanks Andrea ---------- Original Message ---------------------------------- From:

Hi All, I'm very new to speex and in fact handling audio at all, it seems I have run in to a problem I seem unable to fix. I'm trying to take audio from a microphone using alsa, then encode it as speex and save to disk. I have been wondering if it has something to do with endian type, but speexenc and speexdec works fine. Currently I have the following setup: Platform:

> I had a similar question ... is the endian-ness of the encoded > speex file, system dependent? or is it always little endian? If it's > always little endian (like the header seems to be) then big endian > machines (or java) will need to map everything to bigendian before > decoding ... Well, wav's are considered little endian but for raw files, there's a

Hi all, I've built Speex 1.03 on Mac OS X. I encoded a raw PCM file successfully using: /usr/local/bin/speexenc raw_samples.raw --rate 44100 --be --16bit raw_enc.spx Warning: Speex is only optimized for 8, 16 and 32 kHz. It will still work at 44100 Hz but your mileage may vary Encoding 44100 Hz audio using ultra-wideband (sub-band CELP) mode (mono) If I decode to .wav, it works fine.

Hi, I have writen a simple JNI wrapper for speex so i could use it from Java, so far i managed to record from a raw pcm file and from the javasound inputLine and to decode the output in narrowband CBR mode and play it back still using javasound. And it sounds quite good. I haven't tried anything else because my project has no use for wideband and ultra wideband. VBR would be nice but i have

Hi Jean-Marc, Thanks. I've got the code for sampleenc and sampledec. But I didn't know how to verify whether the decoded file is correct. So I tried to analyse the decoded output and compile with my input. Test case: PCM samples, 8 bit, 8000 samples per sec. When I pulled all my input to 0s, after encode and decode, the output fluctuates around -0.12 to 0.2 (in float), so when finally

John Miles wrote: > You'll definitely have to encode in little-endian form. PCM audio is always > little-endian, to the best of my knowledge. I have never seen any > big-endian audio data files. Try AIFF, AU and IFF just for starters. They are definietly not the only ones. Erik -- +-----------------------------------------------------------+ Erik de Castro Lopo

Hi John, I noticed the values after decode is very much different from the values of the original PCM samples. For cases when the PCM sample values are constant (because no one is speaking), the decoded values fluctuate too. These fluatuations are causing the background noise. Is my observation expected? Thanks, Carine ----- Original Message ---- From: John Miles <jmiles@pop.net> To:

Hi, I am capturing 16 bit mono sound samples using some Win32 API function calls. These function calls return the sound samples in an array of characters. My assumption is that this array of characters represents pairs of bytes that make up a short integer. I'm using the following code to convert the samples: LPTSTR *lpSaveBuffer; ...... for(j=0;j<nBytes;++J) { BYTE a =

Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not

> Unfortunately the ALSA API defines a number of formats which are > in practice extremely rare. In particular, any unsigned int format > larger than 8 bits. For instance, the only unsigned int type that > libsndfile supports is unsigned 8 bit. I expected this, it just seemed like a good starting point to get more than 7 formats on the table. Specifically I wanted to the logarithmic

Hello Jean-Marc, I encoded a 143 kb file on both the desktop as well as the Pocket PC using the default settings/ Here are the results. Size : Desktop - 17.6 Kb, Pocket PC -17.1 kb (not much difference in size) Now when I try to decode the file encoded on the Pocket PC, I get a message "This doesn't look like a Speex file". However, when I decode the Pocket PC encoded file on

Hello. I'm running the swfdec-extract tool built from the 0.6.0 source tarball under gentoo and I'm finding that it's not able to rip the MP3s from the SWF file. I'm getting this: SWFDEC: ERROR: swfdec_codec_gst.c(269): swfdec_gst_decoder_init: failed to create decoder It manages to save the file as a WAV file, but I'd prefer the MP3 to get at the ID3 info in the file.

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Hi, all, I have a question about encoding/decoding in big/little endian machines. I have a client running in an old PowerBook G4 with big-endian CPU, the audio stream is encoded then transmitted to a Mac Mini with little-endian CPU, and decoded in this Mac Mini. I am wondering the decoded stream is in big-endian or little-endian? What kind of necessary steps I need to process when I build the

Hi, I got my hands on the LCA 2003 CD today and I tried to listen to it on my powerbook running Linux. Instead of getting speech, I got an earfull of noise! Attempting the same experiment on an i386 with the same CD produced understandable speech. It seems that the output routine in speexdec converts the output data to little endian short format, which is incorrect on the ppc architecture. The

Hi all, Siliva contacted me about this OggPCM proposal and asked me to join in. For those who don't know me, I am the main author and maintainer of libsndfile and therefore know quite a bit about how uncompressed audio is stored in sound files. However even I would not consider myself an expert; there are areas to do with channel assignments that I know I am ignorant of. I am also quite

List, A recent discussion over on XiphWiki is trying to decide if OggPCM should support only little-endianness or the usual combo of big and little. It started with the following statement by an user (Qqq): "Portable players are usually ARM, which is usually little-endian. The Macintosh is now little-endian. Obviously the PC is little-endian. Clearly there is a winner. It's long past

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling