similar to: Buffer size/rate woes

On Apr 21, 2007, at 3:53 PM, zmorris@mac.com wrote: > On Apr 21, 2007, at 12:36 PM, zmorris@mac.com wrote: > >> Hi, I tried both the stable and beta versions of the speex source >> code download on Mac OS 10.4.9. I just do: >> >> ... >> >> However, when I play the output file, I get the header and a >> second of audio, but the rest is just

Hi, I tried both the stable and beta versions of the speex source code download on Mac OS 10.4.9. I just do: ./configure make sudo make install Then I added libspeex.a from /usr/local/lib and the headers to my xcode project. My app compiles and I'm able to call all of the speex functions. I copied the example code from the website and tweaked it to include the first 10000 bytes of

Hey everyone, I just got Theora running on my Mac, and ran across several hurdles, that I was wondering if someone could help me with. I do a lot of tech support at work, and get the same questions over and over, so I tried skimming the archives but couldn't find the answers. Maybe these could go in a FAQ of some sort? These are fairly unavoidable issues that should probably be

Hi all, I am attempting to test AEC behavior at various sample rates. I ran a little experiment: I recorded a 10 seconds voice clip and the resampled at 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000. I have a small applications that plays a wave file, records whatever comes in from the microphone and applies the Speex AEC and preprocessor on the input. It then saves the raw

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

It is commonly said here that if I want to make AM radio-quality stuff at very low bitrates, a good way is to downsample. I downsampled a song to 11025Hz mono and encoded with -q 0, the result is about 18kbps and is at least radio quality. The downsampler I used is from Edinburgh speech tools, named ch_wave. `sox' performs terribly, so I didn't use it. However, I heard some unpleasant

As the subject implies, my question is: is it possible to use the -b (or -M) flag at non-44K sample rates? I'm working with an application that is trying to optimize for very small audio filesize. I found that downsampling to 11K and then using q0 gives high compression, but won't seem to drop below 64kbps or so. It seems like the combination of downsampling, then reducing to 30kps

I downloaded and built the Ogg and Vorbis projects and added the frameworks to my project. It just SIGTRAPs before reaching main, in dylib loading code. I then built libvorbis.a, libvorbisenc.a, libvorbisfile.a and added them to my project. Still SIGTRAPs. I built a libogg.a target from scratch in the Ogg project, and added libogg.a, still SIGTRAPs. So I have tried everything I know

Hi All, I've been doing a lot of cross platform work lately, getting Ogg Vorbis to run on Mac and PC, but haven't been able to find a readily available version of libogg and libvorbis for iPhone. I see a macosx target in the SVN trunk but not an iphone one. I found these links with some background: http://lists.xiph.org/pipermail/theora-dev/2008-October/003719.html

On Jul 2, 2007, at 7:34 PM, Jean-Marc Valin wrote: > Selon "Coffey, Michael" <mcoffey@avistar.com>: >> Believe me; I've "played with" priorities and buffering. > > Then either you haven't played well enough or you're using a > braindead OS. This is sort of what I was talking about with nibbling. Imagine you have a microphone sampling

Kirk, Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you don't use one of these sample rates, you'll be messing up important assumptions deep within the codec. Why these sample rates? It's telecommunications tradition, rather than PC audio tradition. If you want an efficient and high quality format for voice chat, try 16kHz with VBR quality 6. You should see

Hi All, I have tried using echo cancellation, and I think it doesn't work well, because of how I resample. I take the raw 44100 Hz on my mac and just do a loop averaging every 5.5 samples to 1 sample to get 8018.18 Hz, and feed that to speex in 160 sample buffers. I realize that this introduces some aliasing, and I may try the new speex resampler, but I find it hard to believe that

On Sun, 29 Aug 2004 17:53:29 -0700, Ralph Giles wrote: > On Mon, Aug 30, 2004 at 03:03:28AM +0100, Andy Baxter wrote: > >> Is there any way to bring the bitrate in ices2 down below 32 kbps? > > Generally the trick for this is to downsample the audio before encoding. > You can ask ices to do this with a resample stanza in the config file: > > <resample>

Hi, Maybe a bit off topic for this list, bt anyway. I have received several feature requests for DarkIce to support downsampling of the audio input before passing it to lame or ogg vorbis. For example the audio read from the soundcard would be 44.1kHz, and lame would get it at 22.05kHz. I figure two ways of doing this: 1. For lame, one can specify the input and the desired mp3 sampling rate,

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

There are a few flac players on OS X now, but none (at least none that i know of) seem to use Core Audio. So they all freak out if i try to play a 24/96 file. I was recently poking around with the Tiger X Code tools and there's a simple core audio player in there /Developer/Examples/CoreAudio/Services/AudioFileTools/ called afplay. how hard would it be to get this thing playing flac

Hi everyone, I'm still new at this audio development thing in OSX as all of my prior experience is in database applications. I'm also new to C++ and the Xcode environment, which makes things interesting as I learn. There is a CoreAudio wave player written in C++ to which I would like to add FLAC playback support. I have looked through the libFLAC++ API and think I have an idea of what

Hello all, Is anyone here potentially up to the task of adding support for CAF (the CoreAudio Format) into the flac command-line? This would present minimal difficulty under OSX, due to the presence of the CoreAudio API, but the real challenge would be to support CAF on Unix and Windows - everywhere that flac is now available. Although the format is rather unknown, there are some very

That's pretty bad. Both DirectSoundCapture and WinMM are capable of recording at 16kHz. I don't know why OpenAL would be incapable of handling it. It's not like it's at all rare or new. I would try 16000 and see if it works. Maybe the docs are wrong? Note that one option to retain high quality is to capture at a higher rate and then downsample using a resampling