| Commit message (Collapse) | Author | Age | Files | Lines |
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Patch-level: Merged
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Patch-level: Merged
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Patch-level: Merged
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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)
Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
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This patch file adds a REAME file for alsa-dsp plugin.
Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
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This patch file changes the build configuration files to add alsa-dsp
plugin to communicate
with n770 system.
Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
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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"]
}
Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
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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"]
}
The plugin supports the following:
* Playback:
o 16-bit PCM formats:
+ S16_LE
+ S16_BE
+ U16_LE
+ U16_BE
o 8-bit PCM formats:
+ A_LAW
+ MU_LAW
+ U8
+ S8
o Rates:
+ 8 KHz
+ 11.025 KHz
+ 12 KHz
+ 16 KHz
+ 22.050 KHz
+ 24 KHz
+ 32 KHz
+ 44.1 KHz
+ 48 KHz
o Channels:
+ Mono
+ Stereo
* Recording:
o 16-bit PCM formats:
+ S16_LE
o 8-bit PCM formats:
+ A_LAW
+ MU_LAW
o Rates:
+ 8 KHz
o Channels
+ Mono
Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
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This patch file adds communication protocol with maemo SDK audio system.
Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
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This patch file adds header files needed by alsa-dsp plugin.
Signed-off-by: Eduardo Valentin <eduardo.valentin@indt.org.br>
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Patch-level: Merged
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Patch-level: Merged
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Some stray mentions of the old Polypaudio name was still present in the
PulseAudio plug-in.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
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Patch-level: Merged
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Patch-level: Merged
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The name in configure.in used a different capitalisation than the name
in the corresponding Makefile.am. Change it so that both use just lowercase.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
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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.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
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Added missing a52.txt in EXTRA_DIST.
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Patch-level: Merged
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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.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
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- 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.
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Added hgcompile script.
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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.
From: Maarten Maathuis <madman2003@gmail.com>
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Patch-level: Merged
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Accept integer value for card option, as found in the example
in a52.txt.
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the jack plugin closes stdin if the pcm interfaces is opened but jack
isn't running. Initializing the file descriptors to -1 fixes the problem.
From: Mikael Magnusson <mikma264@gmail.com>
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Patch-level: Merged
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Patch-level: Merged
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- Fix iec958 header frames for S16-BE
- Add more comments in the code
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Check XRUN in the write function and pointer callback of a52 plugin.
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Fix .hgignore to use glob patterns.
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Added (experimental) a52 output plugin.
The plugin requires libavcodec as the audio encoding engine.
See doc/a52.txt for the usage.
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Check the malloc error properly.
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- Add channel option to specify the output channels explicitly
- Fix 6-channel input
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Added a rate converter pluging using libsamplerate.
The plugin is built only when libsamplerate is detected by configure.
See doc/samplerate.txt for usage.
This plugin is released under GPL (to follow the license of
libsamplerate), not LGPL.
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