New audio filter layer
Hi, I have finally come to a state in the development of the new audio filter layer where I can show my work. The patch is attached to this mail. I still have some problems with the buffering. If you want to help me have a look at line 179 in dec_audio.c. Sometimes I have to throw away data in order to avoid buffer over flow in buf this only happens when forward/backward but is still annoying and makes the aflayer unsuitable to use with mencoder. The new and fun stuff is: 1. Resampling from any to any sample rate. 2. Auto configuration 3. Changing the number of channels to any number of channels. Anyway it as you can see the patch is huge. Have a look and give me feedback. I hope to apply it to CVS soon. Cheers, //Anders
Hi, I'm sending you an audio filter to the Anders' new layer and a program to see this data. My audio filter does not filter anything, just do some analysis and output it via FIFO. The other program is a SDL window that captures data from FIFO and displays it. It's based on plugins, but I've done just a basic "bar plot" yet, Some cool ones to come. You'll need the fftw package (http://www.fftw.org) and have to modify some files: main/Makefile: add "-lfftw -lrfftw" to the end of COMMON_LIBS main/libaf/af.c: replace it with the attached copy (Anders version doens't support choosing plugins in the CMD line yet, so you must compile with your option) main/libaf/Makefile: add af_analysis.c to SRCS And create this fifo file: ~/.mplayer/af_analysis.fifo. Just mkfifo ~/.mplayer/af_analysis.fifo Just for your information, after aplying these things and recompiling, You'll notice that mplayer becomes blocked. It's because no one is listen to the fifo it's outputing. So you launch the "audio_visualization" program (also attached) and it'll display an ugly bar plot I count on you to improve this program, so give it a try and send me your feedback! Gustavo --- Anders Johansson <ajh@watri.uwa.edu.au> escreveu:
Hi,
I have finally come to a state in the development of the new audio filter layer where I can show my work. The patch is attached to this mail. I still have some problems with the buffering. If you want to help me have a look at line 179 in dec_audio.c. Sometimes I have to throw away data in order to avoid buffer over flow in buf this only happens when forward/backward but is still annoying and makes the aflayer unsuitable to use with mencoder.
The new and fun stuff is:
1. Resampling from any to any sample rate. 2. Auto configuration 3. Changing the number of channels to any number of channels.
Anyway it as you can see the patch is huge. Have a look and give me feedback. I hope to apply it to CVS soon.
Cheers, //Anders
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participants (2)
-
Anders Johansson -
Gustavo Sverzut Barbieri