Commit graph

1069 commits

Author SHA1 Message Date
Wim Taymans
b72eed8f03 audioadapter: don't emit result events for internal param enums
All of the _sync param enum calls are internal end should not result in
a result event externally.

See !2260
2025-01-25 16:34:35 +01:00
Wim Taymans
0c8f803d59 audioconvert: optimise f32d to/from s16s conversions
They are mostly used for sending and receiving RTP so it might be worth
to make them a bit faster.
2025-01-24 10:37:02 +01:00
Arun Raghavan
980d37fb80 audioadapter: Trivial whitespace fix
Makes an annoying clang warning about mismatched indents.
2025-01-21 10:47:14 -05:00
Arun Raghavan
e57a01594e audioconvert: Only use rate_scale when working at DSP rate
In convert mode, we will be resampling directly to the output rate, so
don't try to scale to driver rate.
2025-01-21 08:50:43 -05:00
Wim Taymans
d2c2276088 adapter: pass follower direction to converter
Partially revert 86af9de739

The PortParam does not give enough information to derive the direction
of the converter. If the converter is configured in convert/convert
there is just no way to know when to output a quantum or not.

Fix this by doing a quick probe of the follower and then pass the
direction to the converter.

See !2227
2025-01-21 11:43:11 +01:00
Pauli Virtanen
5f21ee8669 audioconvert: add delay_frac to spa_io_rate_match
Report the "fractional" part of the resampler delay in
spa_io_rate_match::delay_frac, in nanosamples (1/1e9 sample) at node
rate.

The delay values are best reported in units where it is clear what the
clock domain is, so report the value in fractional samples instead of
nanoseconds. Conversion to ns is also just dividision by the appropriate
rate.
2025-01-19 17:11:13 +02:00
Pauli Virtanen
f3a9ebd569 audioconvert: resampler: change resample_phase() unit to input samples
Report fractional delay in input samples instead of nsec at the nominal
input rate, as that is closer to what the value actually means.
2025-01-19 17:11:13 +02:00
Pauli Virtanen
eb91f097d9 audioconvert: resampler: fix off-by-one issues
Resampler without prefill was sometime outputting with different delay
than with prefill. Adjust initial history by 1 which seems to bring it
more in line.

The resampler phase also appears to depend on how many samples remain in
history which leads to possibly unexpected +-1 variation. Take this into
account in reported phase.

These changes make the resampler delay tests pass.

Both changes are sort of empirical --- I don't fully understand why
these would fix things but they seem to be needed to make the delay
calculations agree with what the resampler outputs.
2025-01-17 19:10:43 +02:00
Pauli Virtanen
79384530b5 audioconvert: resampler: add more resampler delay tests
Test also conversions between different rates, and with and without
prefill.
2025-01-17 19:10:43 +02:00
Pauli Virtanen
07f6dde3dd audioconvert: add function to get phase to resampler API
Support getting any fractional phase part of the resampler delay in its
API.
2025-01-17 19:10:43 +02:00
Wim Taymans
85c5d65c97 audioconvert: only consume from input what the resampler used
Keep track of the consumed samples from the input and use that to update
the in_offset. The resampler can tell us how much samples were used.
2025-01-17 12:20:06 +01:00
Wim Taymans
f914cf9327 audioconvert: the delay of the resampler is in input rate
The resampler needs ntaps/2-1 (delay) input samples to produce 1 output
sample so the delay is measured in input rate.
2025-01-15 17:02:39 +01:00
Wim Taymans
0913b3ef7b audioconvert: report delay in input rate
The delay is always expressed in samples at the output rate of the
resampler. For input streams we need to convert this to the expected
input rate.

Make the delay reporting in playback streams more accurate.
2025-01-15 15:25:49 +01:00
Pauli Virtanen
2c2bed2aeb audioconvert: fix resampler delay value and add test
Resampler delay for N taps is N/2-1 input samples.

Add test that checks this.

When input rate is varying, the resampler also accumulates additional
sub-sample delay. The resampler does not currently have API to report
the instantaneous sub-sample delay.  Add knownfail test for it.
2025-01-13 19:05:58 +02:00
Wim Taymans
049ab37a6d audioconvert: only deactive old filter when new filter loaded
When we fail to load the new filter, keep the old filter active or else
we will cause a crash when we clean up the old graph.
2025-01-13 16:58:28 +01:00
Wim Taymans
0868ff1ada math: improve denormal check
Use isnormal() when we can
2025-01-13 16:10:34 +01:00
Wim Taymans
73e11eea46 audioconvert: add wav stage only when needed 2025-01-13 15:39:32 +01:00
Pauli Virtanen
7d8657b7f4 audioconvert: recalculate stages on volume change
Volume changes may change mix passthrough status, so force stage
recalculation when they are done.
2025-01-11 16:13:11 +02:00
Wim Taymans
48416b32ad audioconvert: improve Buffer params
Make sure we only make the buffer for the follower larger when we
downsample because then we need to ask for more data from the follower
to fill up a quantum.

Never try to make the follower buffer smaller than the quantum limit.
The reason is that the graph rate could be decreased dynamically and
then we would end up with too small buffers.

See #4490
2025-01-09 11:30:02 +01:00
Wim Taymans
3a65472e9e audioconvert: add support for filter-graphs
Load multiple graphs with audioconvert.filter-graph.N where N is the
order where the graph is inserted/replaced. Run the graphs before the
channelmixer.

Graphs can be added and removed at runtime.
2025-01-07 13:20:31 +01:00
Wim Taymans
1f4e8b96c2 audioconvert: fix remap stages
Merge remap with convert to simplify some things.
2025-01-07 12:58:59 +01:00
Wim Taymans
c00df67c12 audioconvert: schedule work offline
Instead of recalculating what to do every cycle, we can prepare a
static schedule and just run that. We only need to reevaluate it when
something changes.
2025-01-07 12:58:55 +01:00
Wim Taymans
3fbf84f612 audioconvert: run resampler depending on direction
For input streams, first run the resampler and then the channelmix. This
ensures that the channelmix is run with the rate of the graph instead
of the rate of the input. This is nicer because rate and quantum align
with the graph and the sample accurate volume ramps will work as
intended.

For output streams, leave the resampler after the channelmix for the same
reasons.
2025-01-07 12:03:59 +01:00
Niklas Carlsson
7785f61677 audioconvert: align low-/highpass calculations
The current biquad calculations are based on RBJ's cookbook [1],
except for low-/highpass. Since the filter configuration is also
based on using the definition of Q, it makes sense to also align
the remaining calculations to use the same filter cookbook instead
of using resonance which doesn't result in the same coefficients
as when using Q.

[1] = https://www.w3.org/TR/audio-eq-cookbook/
2024-12-20 11:03:43 +01:00
Wim Taymans
0a0b1d594b audioconvert: use the same biquad as filter-chain 2024-11-27 16:47:53 +01:00
Wim Taymans
84bd4b7ea9 spa: use static inline for interfaces instead of macro
It gives better typechecking and a path to make a library of functions.
2024-11-26 11:35:49 +01:00
Wim Taymans
68877dcee7 audioconvert: only output when there is something to output
Or else we get 0 sized output buffers.
2024-11-06 09:28:09 +01:00
George Kiagiadakis
726234c82f audioconvert: always push out data when the out_offset reaches max_out
No matter if the input data (n_samples) is > 0, the code here should
flush out existing data if the output buffer is full
2024-11-04 20:36:12 +02:00
Wim Taymans
fb2b314660 audioconvert: interchange the resampler loops
Iterate the channels in the inner loop instead of the outer loop. This
makes it handle with 0 channels better but also does the more
complicated phase increment code only once for all channels. Also the
filters might stay in the cache for each channel now.
2024-10-30 13:31:24 +01:00
Wim Taymans
2c132be626 audioconvert: align some buffers
so that we can use aligned read and writes in SSE.
2024-10-23 12:54:23 +02:00
Wim Taymans
662bf68122 audioconvert: handle odd writes in delay
Add some padding to the delay buffer. If we wrap around, copy the
spilled samples to the front of the buffer. This makes it possible to
use the more optimized sse delay function in more cases.
2024-10-23 12:34:04 +02:00
Wim Taymans
3309e0b244 audioconvert: don't unroll when unaligned write pointer
We require the write pointer to be a multiple of 4 for our unrolled SSE
loop to work, so enforce this to avoid segfaults.
2024-10-23 11:40:34 +02:00
Wim Taymans
2c0ce6afc2 audioconvert: SSE optimize delay and convolver 2024-10-15 16:10:25 +02:00
Wim Taymans
33fb2f04c7 audioconvert: use MAX_TAPS constant 2024-10-15 12:49:16 +02:00
Wim Taymans
ad84c45c0c audioconvert: optimize lr4 handling
Make an SSE optimized LR4 function and one that handles 2 LR4s in
parallel.
2024-10-15 12:20:04 +02:00
Wim Taymans
fec3730489 audioconvert: avoid a memcpy when we can 2024-10-15 12:19:17 +02:00
Wim Taymans
a57f2f25b6 delay: improve delay performance
Use a wrap around delay ringbuffer. We can then avoid some modulo
arithmetic and read more efficiently.

Also handle the delay convolver case better by reversing the taps and
reading the taps and delay buffer without extra overhead.
2024-10-15 12:14:57 +02:00
Wim Taymans
8cd3fc6922 adapter: increase max-retry to 64
When the follower doesn't produce enough data for this many attempts,
bail and cause an xrun to avoid an infinite loop.

The limit of 8 cause real-life problems and should be larger. It should
probably depend on the expected size per cycle (node.latency) and the
current quantum but we don't always have this information.

See #4334
2024-10-07 09:28:39 +02:00
sunyuechi
245adda985 fmt-ops: add RVV optimizations for s32_to_f32d 2024-09-29 11:17:42 +08:00
sunyuechi
79d41e183e fmt-ops: add RVV optimizations for f32d_to_s32 2024-09-26 00:55:49 +08:00
sunyuechi
74832445ba fmt-ops: add RVV optimizations for s16_to_f32d 2024-09-25 10:50:05 +00:00
sunyuechi
588f2bcb69 RISCV: Improve scalar computation of f32d_to_s16 2024-09-25 10:50:05 +00:00
sunyuechi
8a8843ba20 fmt-ops: add RVV optimizations for f32d_s16 2024-09-23 08:10:43 +00:00
sunyuechi
852de6c35c fmt-ops: add RVV optimizations for f32d_s16d 2024-09-23 08:10:43 +00:00
sunyuechi
62ec61a3bb benchmark-fmt-ops: Fix test arg for f32_s16 2024-09-20 16:02:22 +08:00
sunyuechi
d932e52d5b fmt-ops: add R-V V optimizations for f32_s16 2024-09-18 10:40:48 +00:00
Wim Taymans
e2991f6398 json: add helper function to parse channel positions
Use the helper instead of duplicating the same code.

Also add some helpers to parse a json array of uint32_t

Move some functions to convert between type name and id.
2024-09-18 09:54:34 +02:00
Wim Taymans
563186eff6 adapter: also forward the RequestProcess events 2024-09-16 17:01:50 +02:00
Wim Taymans
ce390d5b22 spa: add spa_json_object_next
This gets the next key and value from an object. This function is better
because it will skip key/value pairs that don't fit in the array to hold
the key.

The previous code patter would stop parsing the object as soon as a key
larger than the available space was found.
2024-09-16 09:50:36 +02:00
Wim Taymans
cd81b5f39a spa: add spa_json_begin_array/object and relaxed versions
Add spa_json_begin_array/object to replace
spa_json_init+spa_json_begin_array/object

This function is better because it does not waste a useless spa_json
structure as an iterator. The relaxed versions also error out when the
container is mismatched because parsing a mismatched container is not
going to give any results anyway.
2024-09-16 09:50:33 +02:00