| Commit message (Collapse) | Author | Age | Files | Lines |
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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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Add missing .hgignore.
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Patch-level: Merged
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Change AC_PREREQ to 2.59.
Also, requires ALSA >= 1.0.11 (for bugfix of explug).
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Added PCM upmix and vdownmix plugins.
The upmix plugin is for upmixing to 4.0 or 5.1 surrounds
by simple copying (and delay line for rear).
The vdownmix plugin is a kind of "virtual surround", which
downmixes 4.0 or 5.1 input to 2.0 output with some effect.
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Fix a minor compile warning.
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Fix misc ompile warnings with ctl/pcm OSS plugins.
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Add a mutex to make sure that the Polypaudio plugin is thread safe.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
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The revent flags POLLIN/POLLOUT got switched in the Polypaudio plugin.
Most applications check both so it went unnoticed until now.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
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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.
Signed-off-by: Pierre Ossman <ossman@cendio.se>
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Fix a typo in file name.
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