Wim Taymans
ecc0eecf0f
mixer-dsp: mix a quantum of data at a time
...
When there is no input, mix up to a quantum of data. Otherwise we might
send too much data to the next node and cause a delay if it does not
handle this.
2022-06-22 15:50:56 +02:00
Wim Taymans
a427e602bd
audioconvert: in merge mode, flush input
...
In merge mode we handle 1 quantum and then flush the input and
output. Fixes a case were samples were cut at the start.
2022-06-22 15:48:09 +02:00
Wim Taymans
a13f65f583
alsa-seq: disable the running status
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Running status is not something we want in the PipeWire graph.
2022-06-20 16:18:42 +02:00
Wim Taymans
25d9039f2e
alsa-seq: handle midi event decoder errors
2022-06-20 16:12:49 +02:00
Wim Taymans
40f50deae9
alsa: remove NoteOn 0-velocity fixup
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Pass MIDI events as they are.
JACK requires NoteOn 0-velocity midi events to be patched to NoteOff
events for compatibility with LV2 plugins. Let's do this patchup in
the JACK layer then and add an option to disable it.
It's best to pass the midi messages unmodified and then patch them up
wherever they need patching up.
2022-06-20 15:51:33 +02:00
Wim Taymans
d13a206866
audioconvert: fix resampler in_offset
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Only advance the in_offset with the number of samples that were consumed
by the resampler. In case when the resampler is filling up an old
buffer, this can be less than n_samples.
Fixes a2dp source and possibly others.
2022-06-20 12:33:52 +02:00
Lucas Holt
6a15a02ec2
Add support for MidnightBSD
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Fix build issue
Fix build issue
2022-06-19 18:22:47 +00:00
Barnabás Pőcze
492a328227
spa: audioconvert: free resampler
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resample::free() needs to be called in `impl_clear()` otherwise the
resampler's private data is leaked when the audioconvert instance is destroyed.
2022-06-16 19:49:50 +02:00
Barnabás Pőcze
96d0902dc8
spa: ffmpeg: set spa_handle_factory::version
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Initialize the version with the `SPA_VERSION_HANDLE_FACTORY` macro.
This way `spa-inspect` can properly inspect the factory.
2022-06-16 17:29:42 +02:00
Barnabás Pőcze
de5a85d808
spa: ffmpeg: implement spa_handle::clear
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Implement the spa_handle::clear method for both the encoder and decoder.
At the moment, they both do nothing.
2022-06-16 17:29:20 +02:00
Barnabás Pőcze
f2bd2ef6fb
spa: ffmpeg: implement spa_handle_factory::get_size
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This method is necessary for the creation of a handle,
so implement it for both the encoder and decoder.
2022-06-16 17:28:23 +02:00
Barnabás Pőcze
6cb8fb899d
spa: ffmpeg: move some function declarations to a header
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Move the declarations for `spa_ffmpeg_{dec,enc}_init` to the
"ffmpeg.h" header file. This way the implementation is checked
against the signature.
2022-06-16 17:27:36 +02:00
Wim Taymans
3cb5fab176
audioconvert: rework the control loop
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Use the offset to skip entries in the sequence array.
Use one loop to handle intermediate and trailing samples.
Fixes an issue where the last chunk of a sequence would be ignored.
2022-06-16 16:45:32 +02:00
Wim Taymans
de2e819fec
audioconvert2: rename to audioconvert
2022-06-16 09:19:05 +02:00
Wim Taymans
cf04bb573c
audioconvert: remove old plugins
2022-06-16 09:16:49 +02:00
Wim Taymans
c160cd0176
audioconvert2: implement control port
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Add a control port when requested in the PortConfig.
Fix the sequence loop.
Make a copy of the sample pointers before changing them.
2022-06-16 09:09:27 +02:00
Wim Taymans
5a60fd7041
audioconvert2: remove unused monitor field
2022-06-16 09:09:27 +02:00
Wim Taymans
194b8e2d97
audioconvert2: emit param change when volume changes
2022-06-16 09:09:27 +02:00
Wim Taymans
0ec41e60ef
audioconvert2: fix for quantum changes
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Use the offset as the buffer size.
Use the amount of requested - amount of used samples as queued samples.
2022-06-16 09:09:27 +02:00
Wim Taymans
fa9baa6488
examples: fix up adapter-control
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Negotiate format, then buffers.
Give quantum-limit as a property when making nodes.
Set clock and position io and fill in rate and duration.
2022-06-16 09:09:27 +02:00
Wim Taymans
db49021104
audioconvert2: add control port processing
2022-06-16 09:09:27 +02:00
Wim Taymans
d5c28149f1
audioconvert2: fix in/out queued samples
2022-06-16 09:09:27 +02:00
Wim Taymans
7fe15ee5db
audioconvert2: add beginnings of control ports
2022-06-16 09:09:27 +02:00
Wim Taymans
d3baa16559
audioconvert2: reset node in Flush/Suspend
2022-06-16 09:09:27 +02:00
Wim Taymans
8e6a8a705b
audioconvert2: clamp in and output samples
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Make sure we don't read or write more samples than available in the
buffers.
2022-06-16 09:09:27 +02:00
Wim Taymans
9e93fe3c36
audioconvert2: remap volumes to right channels
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We get the volumes with a channelmap of the input/output, remap this to
the channelmixer internal layout.
2022-06-16 09:09:27 +02:00
Wim Taymans
50c37cc801
audioconvert2: rearrange some code
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Move some code around so that we don't have to prepare the output
buffers when there is no input to process.
2022-06-16 09:09:27 +02:00
Wim Taymans
d5e333eea7
audioconvert2: in merge mode, always process quant samples
2022-06-16 09:09:27 +02:00
Wim Taymans
c4d77d421a
audioconver2: process monitor ports immediately
2022-06-16 09:09:27 +02:00
Wim Taymans
f45f1acd82
audioconvert2: use the position rate as resampler rate
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For the DSP inputs, use the position rate.
2022-06-16 09:09:27 +02:00
Wim Taymans
58abc1ec15
audioconvert2: Improvements
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Suggest 2 buffers by default.
Improve rate match flow.
monitor does not need the out_offset
Update rate match when we know the queued samples.
2022-06-16 09:09:27 +02:00
Wim Taymans
2352ae33c3
audioconvert2: add queueing
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Use in and out offset to read/write partial input/output.
2022-06-16 09:09:27 +02:00
Wim Taymans
d728076d33
audioconvert2: handle drain
2022-06-16 09:09:27 +02:00
Wim Taymans
1c4058b947
audioconvert2: clamp offset and size
2022-06-16 09:09:27 +02:00
Wim Taymans
229a7b6019
audioconvert2: fix default settings
2022-06-16 09:09:27 +02:00
Wim Taymans
f2c1fe106d
audioconvert2: more fixes
2022-06-16 09:09:27 +02:00
Wim Taymans
3806cdaa6f
audioconvert2: more improvements
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Use a wildcard rate for DSP ports.
Handle wildcards for rate and channels.
Calculate required in/out samples using quantum
Limit monitor and output number of samples.
2022-06-16 09:09:27 +02:00
Wim Taymans
c0f34e9d9d
audioconvert2: set blocks a stride for DSP ports
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They might not get connected and negotiated but we still need to
use empty buffers in that case.
2022-06-16 09:09:27 +02:00
Wim Taymans
2d6bf00acc
audioconvert2: add options
2022-06-16 09:09:27 +02:00
Wim Taymans
75ebdbdfa0
audioconvert2: start working on monitor ports
2022-06-16 09:09:27 +02:00
Wim Taymans
db2cbdf906
audioconvert2: DSP mode without format gives no ports
2022-06-16 09:09:27 +02:00
Wim Taymans
d53311099f
audioconvert2: format is optional for convert mode
2022-06-16 09:09:27 +02:00
Wim Taymans
e78a93d358
audioconvert2: fix audio rate property
2022-06-16 09:09:27 +02:00
Wim Taymans
9babdee6cd
audioconvert2: do input and output remapping
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We remap the input channels to our cannonical layout and then remap it
to the target layout.
2022-06-16 09:09:27 +02:00
Wim Taymans
922a088129
audioconvert2: handle volume
2022-06-16 09:09:27 +02:00
Wim Taymans
0099b80098
audioconvert2: add properties
2022-06-16 09:09:27 +02:00
Wim Taymans
8cb0d9fd7b
audioconvert: add channelmix and resample
2022-06-16 09:09:27 +02:00
Wim Taymans
399f638eb1
audioconvert: more work
2022-06-16 09:09:27 +02:00
Wim Taymans
7f9ddd0f15
Add audioconvert2
2022-06-16 09:09:27 +02:00
Ole Salscheider
3695611b20
channelmix: Assign slev in mixing matrix when generating side/rear from front
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This way, PSD upmixing sounds much more natural and is in line with the
original publication.
2022-06-14 10:58:25 +00:00