Takashi Iwai [Thu, 24 May 2007 13:28:22 +0000 (15:28 +0200)]
Enable link with external libspeex
- Add --with-speex configure option to specify the build of speex rate
plugin. As default, it's linked to external library. If not available,
defaults to builtin code.
- Show build conditions at the end of configure script
- Use AS_HELP_TEXT()
I'm attaching an updated version of my resampler plugin. It fixes a few
minor issues and it adds support for fixed-point processing (just add
-DFIXED_POINT to the build). Let me know if there's any problem.
Pierre Ossman [Wed, 7 Mar 2007 14:52:05 +0000 (15:52 +0100)]
Improve parameter constraints in alsa-pulse
Remove parameter constraints where we actually have none. Also, restrict
total buffer size to 4 MB as current versions of the PulseAudio server
will refuse streams larger than that.
Takashi Iwai [Tue, 20 Feb 2007 11:43:58 +0000 (12:43 +0100)]
Add livavcodec resampler plugin
Add a new pcm_rate plugin, "lavcrate", that uses the resampling
filter from libavcodec. It should provide high performance and
good output quality.
Add a documentation file for lavcrate.
Update autoconf and automake to build lavcrate.
Alsa support for Maemo SDK (n770) - Remove compile warnings
Here is a patch to remove compile warnings. It must be applied
after last patch set I sent.
This patch simply changes signedness of some point from the code
to match the correct sign used by dsp-protocol structures. All must
use unsigned variables.
It also changes the way the pthread_mutex is initialized. The
warning about pthreads is also removed.
I tested the compilation with:
gcc (GCC) 4.1.2 20061028 (prerelease) (Debian 4.1.1-19)
and
sbox-arm-linux-gcc (GCC) 3.4.4 (release) (CodeSourcery ARM 2005q3-2)
Alsa support for Maemo SDK (n770): External Control plugin
This patch file adds an ALSA External Control plugin. This source uses
the dsp-protocol
implementation.
The plugin probes for all communication channel at the start time. It
will handle only
channels specified into alsa configuration file. An configuration example is:
# Mixer
ctl.!default {
type dsp_ctl
playback_devices ["/dev/dsptask/pcm2"]
recording_devices ["/dev/dsptask/pcm_rec"]
}
Alsa support for Maemo SDK (n770): External PCM IO plugin
This patch file adds an ALSA External PCM I/O plugin. This source uses
the dsp-protocol
implementation.
The plugin probes for a free communication channel at the start time.
It will probe only
for channels specified into the configuration file for the plugin. An
configuration example is:
# PCM
pcm.!default {
type alsa_dsp
playback_device_file ["/dev/dsptask/pcm2"]
recording_device_file ["/dev/dsptask/pcm_rec"]
}
Pierre Ossman [Wed, 12 Jul 2006 14:47:20 +0000 (16:47 +0200)]
Follow Polypaudio/PulseAudio name change
Polypaudio recently changed its name to PulseAudio which affects the
names of libraries of header files. Update the polyp, now pulse, plug-in
to follow this name change.
Pierre Ossman [Mon, 29 May 2006 10:19:46 +0000 (12:19 +0200)]
Update Polypaudio plug-in to the 0.9.0 API
The new version of Polypaudio includes a threading abstraction that
allows application of a more synchronous nature to use the API more
easily. Using this, the complexity of the Polypaudio plug-in is greatly
reduced and also removes the risk of stalling the communications layer.
Takashi Iwai [Wed, 24 May 2006 10:02:00 +0000 (12:02 +0200)]
a52 - Add slavepcm option
- Added slavepcm option to specify the slave PCM string explicitly
- Don't use plug but linear plugin for default slave.
We need only the linear format conversion, and the channel/rate
conversion should be avoided.
Takashi Iwai [Tue, 2 May 2006 11:50:39 +0000 (13:50 +0200)]
Fix port names in jack plugin
There is a flaw in alsa-jack, the channel name (out_001, etc) is
ended with a newline.
This causes problems when using jack_connect and jack_disconnect.
Pierre Ossman [Wed, 8 Mar 2006 12:16:51 +0000 (12:16 +0000)]
Properly terminate stream in polypaudio plugin
Some applications like to call prepare over and over again, recreating
the stream each time. Previously we just cleaned up the local end each
time, but this makes sure the server also releases its resources.
Takashi Iwai [Thu, 23 Feb 2006 14:31:13 +0000 (14:31 +0000)]
Update polyp plugin
From: Pierre Ossman <ossman@cendio.se>
Here is a final update to the plugin and the documentation. This release
adds capture volume and mute switches. This should also be the last API
change in polypaudio before the next release, so this plugin should be
stable for now.
Takashi Iwai [Wed, 22 Feb 2006 15:38:52 +0000 (15:38 +0000)]
Fixes for polyp plugin
From: Pierre Ossman <ossman@cendio.se>
There shouldn't be any more API changes until polypaudio 0.8 that affect
this plugin. I am working on supporting capture volume and mute
switches, but that should be a strict addition (i.e. it can wait until
the next release if things get tight).
git://git.alsa-project.org / alsa-plugins.git / log