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Bug 550230 - A plug-in "openalsrc" for capture audio using OpenAL
A plug-in "openalsrc" for capture audio using OpenAL
Status: RESOLVED FIXED
Product: GStreamer
Classification: Platform
Component: gst-plugins-bad
git master
Other Windows
: Normal enhancement
: 0.10.23
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2008-09-01 14:32 UTC by Victor Lin
Modified: 2011-05-23 12:32 UTC
See Also:
GNOME target: ---
GNOME version: Unversioned Enhancement


Attachments
gstopenalsrc.c (15.78 KB, text/x-csrc)
2008-09-01 15:17 UTC, Victor Lin
Details
gstopenalsrc.h (1.52 KB, text/x-chdr)
2008-09-01 15:18 UTC, Victor Lin
Details
gstopenalsrc.c (15.78 KB, text/x-csrc)
2008-09-01 15:30 UTC, Victor Lin
Details

Description Victor Lin 2008-09-01 14:32:59 UTC
I just create a plug-in which use OpenAL to capture audio. I test it only under windows. It works correctly.
Comment 1 Victor Lin 2008-09-01 14:41:32 UTC
I have some bug to fix, I will upload it later.
Comment 2 Marc-Andre Lureau 2008-09-01 15:04:30 UTC
very nice
Comment 3 Victor Lin 2008-09-01 15:16:12 UTC
It work correctly with this option.

gst-launch --gst-plugin-path=..//bin/debug openalsrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=test.ogg

But I have no idea why it failed with this option:

gst-launch --gst-plugin-path=..//bin/debug openalsrc ! adder name=mix ! audioconvert ! vorbisenc ! oggmux ! filesink location=test.ogg filesrc location=c:/test.ogg ! oggdemux ! vorbisdec ! audioconvert ! mix.

Error message:
OIL: ERROR liboilcpu.c 135: (): No t
estamping function.  This is kinda bad.
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
ERROR: from element /pipeline0/openalsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2165): gst_base_src_loop (): /pipeline0/openalsrc0:
streaming task paused, reason not-negotiated (-4)
Execution ended after 125000000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
FREEING pipeline ...
Comment 4 Victor Lin 2008-09-01 15:17:35 UTC
Created attachment 117776 [details]
gstopenalsrc.c
Comment 5 Victor Lin 2008-09-01 15:18:09 UTC
Created attachment 117777 [details]
gstopenalsrc.h
Comment 6 Victor Lin 2008-09-01 15:30:41 UTC
Created attachment 117779 [details]
gstopenalsrc.c

fix a crash problem of DEBUG string format typo.

This typo cause a crash:
GST_DEBUG_OBJECT(osrc, "Read samples : %s", samples);
Comment 7 Victor Lin 2008-09-01 15:53:05 UTC
I think it fail because there are different format between audio file and openalsrc. I move audioconvert to front of adder, it works.

G:\Programming\radio\lib\gstopenalsrc\gst>gst-launch --gst-plugin-path=..//bin/debug openalsrc ! audioconvert ! adder name=mix ! vorbisenc ! oggmux ! filesink location=test.ogg filesrc location=c:/test.ogg ! oggdemux ! vorbisdec ! audioconvert ! mix.
Comment 8 Sebastian Dröge (slomo) 2011-05-19 08:40:27 UTC
Could you attach the openalsrc as a patch against gst-plugins-bad git please, including configure checks for openal?
Comment 9 Christian Fredrik Kalager Schaller 2011-05-20 13:19:15 UTC

*** This bug has been marked as a duplicate of bug 615615 ***
Comment 10 Victor Lin 2011-05-20 16:48:04 UTC
I'm sorry, it has been a long time I haven't touch GStream, also, I have no environment to test it for now. I will see is there any chance I have spare time to test it with latest code base and make the git patch.
Comment 11 Sebastian Dröge (slomo) 2011-05-23 12:32:01 UTC
commit 0e596670ef607044c09f28bb46174edeba90e7ed
Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
Date:   Mon May 23 14:14:09 2011 +0200

    openal: Add new openalsrc element
    
    Based on patches by Victor Lin <bornstub@gmail.com>
    
    Fixes bug #550230.