search for: 16000hz

Hi all, I'm attempting to encode raw audio data using libFLAC++. My audio data is 16 bit, mono, 16000Hz. I set all the appropriate parameters on the encoder and then call init(). Everything appears to be ok. I don't know how to properly convert from char *data to the FLAC__int32 *[] requested by the process function. I think this is where my problem is. If I call process() like this: FLAC__i...

I knew I left something out. It would be MP3 encoded at 16kbps/16000Hz -- Phil / w2lie On Sep 5, 2010, at 5:15 PM, Geoff Shang <geoff at QuiteLikely.com> wrote: > Hi, > > I guess the real question is what audio format and quality are you looking at here? > > You will need to have a CPU capable of encoding in real-time and you may not be able...

Hi, I'm no DSP or audio expert by any means, but I can share what works for me. People in the know, I would appreciate tips on whether this stuff is ok. You could sample at 32000Hz (or 48000Hz, any AC97 card will support this), run a 8000Hz lowpass filter over the data (16000Hz sample rate can only represent frequencies up to 8000Hz) and then drop every second (or 2 out of 3 for 48000->16000) sample. The result being, 16000Hz sampled audio. If you omit the filter the result will contain artifacts. At the other end for playback you can convert it back to 48000 (or what...

...m no DSP or audio expert by any means, but I can share what > works for me. People in the know, I would appreciate tips on > whether this stuff is ok. > > You could sample at 32000Hz (or 48000Hz, any AC97 card will > support this), run a 8000Hz lowpass filter over the data > (16000Hz sample rate can only represent frequencies up to > 8000Hz) and then drop every second (or 2 out of 3 for > 48000->16000) sample. The result being, 16000Hz sampled > audio. If you omit the filter the result will contain artifacts. > > At the other end for playback you can conve...

Hi! I would really appreciate if someone could try to help me out with this... i've been struggling with it for quite a while now. :( Anyway, I was trying to use ices to broadcast mp3s with icecast2 (a recent cvs snapshot of all 3 components from brendan's site). I have it working fine for some of my mp3s (encoded with lame -b 32) but with the other mp3s which are actually the ones that

...t 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: PowerPC (powerbook) running Linux (2.6) Input stream: Signed 16bit Little endian, mono, PCM @ 16000Hz if saved out 'as is' can be played with "aplay" using the following command: "aplay -c1 -r16000 -f S16_LE -t raw <fileame>" Bellow is part of some test code that I have be...

I compared quicktime aac and vorbis,i think quicktime aac is better than vorbis at 80Kbps. please tell me if i'm wrong. using command: qtaacenc.exe --tvbr 31 --highest --samplerate keep test.wav qt.m4a oggenc2.exe --raw -q 1.6 test.wav -o vorbis.ogg the version: qtaacenc version 20100725 with QuickTime 7.6.7 OggEnc v2.87 (libvorbis 1.3.1) links:

...l and after doing *CLI> show cepstral female voices Available Cepstral voices: Name Gender Age Language Sample Rate License ========== ========= ===== ==================== ============= ============ Amy female 30 US English 16000Hz unlicensed I see that Amy's voice shows up but when I tried to get cepstral to speak from extensions.conf I get the failed to load library messages shown below. Google searches didn't turn up anything useful yet. Has anyone had this problem before or does someone know what is wrong? This i...

...pageout always returns 0, so, insufficent data has been accumulated to write a page. Can anyone help me and guess, where the fault may be. After all I've read this seems to be the right encoding procedure. So is it the python-wrapper? Is it the data I've been using (16Bit-Little Endian, 16000Hz, Mono wav-file) wrong? Thank you very much! Matthias

...et.au> wrote: > Hi: > > My guess is that ices can't handle 16000 or 32000hz > sampling rates. The > 32kbps file is 22050hz and the 80kbps file is > 32000hz. I couldn't get > mpg123 to play test2 remotely, but lame defaults to > 32000hz with 40kbps > mono and 16000hz with 40kbps stereo, so I'm guessing > this is the problem. > You can either use --resample to force a different > sampling rate or pick a > different bitrate. > > Geoff. > > > -- > Geoff Shang <gshang@uq.net.au> > ICQ number 43634701 > > Make su...

...t 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: PowerPC (powerbook) running Linux (2.6) Input stream: Signed 16bit Little endian, mono, PCM @ 16000Hz if saved out 'as is' can be played with "aplay" using the following command: "aplay -c1 -r16000 -f S16_LE -t raw <fileame>" Bellow is part of some test code that I have be...

I'm in the process of trying to build a project, and I'm hoping that someone on the list can help me out. I have an Icecast server already running, and some feeds coming in from Simplecast, Edcast, and DarkIce. These encoders are obviously on computers owned by users who can setup feeds are send them to my Icecast server remotely. What I am looking for, is a small PC that is capable

...been > wondering if it has something to do with endian type, but speexenc and > speexdec works fine. > > Currently I have the following setup: > > Platform: PowerPC (powerbook) running Linux (2.6) > Input stream: Signed 16bit Little endian, mono, PCM @ 16000Hz > if saved out 'as is' can be played with > "aplay" > using the following command: > "aplay -c1 -r16000 -f S16_LE -t raw > <fileame>" > > Bellow is pa...

Well, I'm not a professional in AEC theory, but what I've mentioned is: speex_echo_state_init(20, 320*10) - frame size should correspond to 20ms. At your sampling rate (16000hz) is should be 16K*0.02 = 320. The same I can notice about echo tail. 100ms: 16000*0.3 = 4800, not 3200 as you has. But that's not crucial I think. Just wanna you get the point. "Internally, speex_echo_playback() simply buffers the playback frame so it can be used by speex_echo_capture()...

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

Hello, In one of my recent assignments, I was asked to develop a ogg player. I am not able to find the right repository of the source code. There are few repositories on vorbis but I am not clear which one is the right one for Windows environment. If somebody has successfully compiled any ogg player ever, his ideas will be helpful to me. In one of the code set, I was able to compile the