This is only a very partial fix - the frequency-dependent envelope & LFO
register values aren't adjusted.
But I'm not sure they were even correct at 48 kHz to start with, as most
of them are precalculated by common code which assumes an EMU8K-specific
44.1 kHz word clock, and it seems somewhat unlikely that the hardware's
register interpretation was adjusted to compensate for the different
word clock.
In any case I'm not going to spend time on fixing that, as this code is
unlikely to be actually used by anyone today.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The value isn't used yet; the subsequent commits will do that.
This ignores the existence of rates above 48 kHz, which is fine, as the
hardware will just switch to the fallback clock source when fed with a
rate which is incompatible with the base clock multiplier, which
currently is always x1.
The sample rate display in /proc spdif-in is adjusted to reflect our
understanding of the input rates.
This is tested only with an 0404b card without sync card, so there is a
lot of room for improvement.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, we set the fallback as a side effect of setting the source. But
the fallback makes no sense at all when an internal clock is selected.
Defaulting to 48k for S/PDIF & ADAT makes sense, but as that is the
global default and we're not changing it automatically any more, it's
just fine to leave it entirely to the explicit setting.
This changes the name of the pre-existing control to something more
appropriate (regardless of the split), so users will need to adjust
their mixer settings.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the updated MIDI 2.0 spec has been published freshly, this is a
catch up to add the support for new specs, especially UMP v1.1
features, on Linux kernel.
The new UMP v1.1 introduced the concept of Function Blocks (FB), which
is a kind of superset of USB MIDI 2.0 Group Terminal Blocks (GTB).
The patch set adds the support for FB as the primary information
source while keeping the parse of GTB as fallback. Also UMP v1.1
supports the groupless messages, the protocol switch, static FBs, and
other new fundamental features, and those are supported as well.
Link: https://www.midi.org/midi-articles/details-about-midi-2-0-midi-ci-profiles-and-property-exchange
Link: https://lore.kernel.org/r/20230612081054.17200-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UMP v1.1 spec allows to inform whether the function blocks are static
and not dynamically updated. Add a new flag bit to
snd_ump_endpoint_info to reflect that attribute, too.
The flag is set when a USB MIDI device is still in the old MIDI 2.0
without UMP 1.1 support. Then the driver falls back to GTBs, and they
are supposed to be static-only.
Link: https://lore.kernel.org/r/20230612081054.17200-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UMP v1.1 supports the protocol switch via a UMP Stream message. When
it's received, we need to take care of the midi_version field in the
corresponding sequencer client, too.
This patch introduces a new ops to notify the protocol change to
snd_seq_ump_ops for handling it.
Link: https://lore.kernel.org/r/20230612081054.17200-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For allowing applications to track the FB active changes, this patch
adds the notification from the system port at each time a FB change is
handled and the active flag or re-grouping happens.
Link: https://lore.kernel.org/r/20230612081054.17200-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements the handling of the dynamic update of FB info.
When the FB info update is received after the initial parsing, it
means the dynamic FB info update. We compare the result, and if the
actual update is detected, it's notified via a new ops,
notify_fb_change, to the sequencer client, and the corresponding
sequencer ports are updated accordingly.
Link: https://lore.kernel.org/r/20230612081054.17200-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UMP Utility and Stream messages are "groupless", i.e. an incoming
groupless packet should be sent only to the UMP EP port, and the event
with the groupless message is sent to UMP EP as is without the group
translation per port.
Also, the former reserved bit 0 for the client group filter is now
used for groupless events. When the bit 0 is set, the groupless
events are filtered out and skipped.
Link: https://lore.kernel.org/r/20230612081054.17200-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Try to parse the UMP Endpoint and UMP Function Blocks for building the
topology at first. Only when those are missing (e.g. on an older USB
MIDI 2.0 spec or a unidirectional endpoint), the driver still creates
blocks based on USB group terminal block information as fallback.
Link: https://lore.kernel.org/r/20230612081054.17200-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the basic support for UMP Endpoint and UMP Function
Block parsing, which are extended in the new UMP v1.1 spec.
The patch provides a new helper function to perform the query of the
UMP Endpoint information and builds up the UMP blocks based on UMP
Function Block information. For the communication over the UMP
Endpoint, it opens the rawmidi device once internally, inquiries the
UMP Endpoint and Function Block info by sending new UMP Stream
messages, and waits for the response for each query.
The new UMP spec allows to update the FB info and change its
associated groups or its activeness on the fly, too. For catching it,
the UMP core keeps watching the incoming UMP messages, and
snd_ump_receive() handles the incoming UMP Stream messages to refresh
the FB info.
Link: https://lore.kernel.org/r/20230612081054.17200-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a few more fields to snd_ump_endpoint_info and snd_ump_block_info
that are added in the new v1.1 spec. Those are filled by the UMP Stream
messages.
The rawmidi protocol version is bumped to 2.0.4 to indicate those
updates.
Also, update the proc outputs to show the newly introduced fields.
Link: https://lore.kernel.org/r/20230612081054.17200-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA can only support single register read and write operations so does not
benefit from block writes. This means it gets no benefit from using the
rbtree register cache over the maple tree register cache so convert it to
use maple trees instead, it is more modern.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230609-alsa-hda-maple-v1-1-a2b725c8b8f5@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This test covers the new Virtual PCM Test Driver, including the capturing,
playback and ioctl redefinition functionalities for both interleaved and
non-interleaved access modes. This test is also helpful as an usage example
of the 'pcmtest' driver.
We have a lot of different virtual media drivers, which can be used for
testing of the userspace applications and media subsystem middle layer.
However, all of them are aimed at testing the video functionality and
simulating the video devices. For audio devices we have only snd-dummy
module, which is good in simulating the correct behavior of an ALSA device.
I decided to write a tool, which would help to test the userspace ALSA
programs (and the PCM middle layer as well) under unusual circumstances
to figure out how they would behave. So I came up with this Virtual PCM
Test Driver.
This new Virtual PCM Test Driver has several features which can be useful
during the userspace ALSA applications testing/fuzzing, or testing/fuzzing
of the PCM middle layer. Not all of them can be implemented using the
existing virtual drivers (like dummy or loopback). Here is what can this
driver do:
- Simulate both capture and playback processes
- Generate random or pattern-based capture data
- Check the playback stream for containing the looped pattern
- Inject delays into the playback and capturing processes
- Inject errors during the PCM callbacks
Also, this driver can check the playback stream for containing the
predefined pattern, which is used in the corresponding selftest to check
the PCM middle layer data transferring functionality. Additionally, this
driver redefines the default RESET ioctl, and the selftest covers this PCM
API functionality as well.
The driver supports both interleaved and non-interleaved access modes, and
have separate pattern buffers for each channel. The driver supports up to
4 channels and up to 8 substreams.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606193254.20791-3-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a lot of different virtual media drivers, which can be used for
testing of the userspace applications and media subsystem middle layer.
However, all of them are aimed at testing the video functionality and
simulating the video devices. For audio devices we have only snd-dummy
module, which is good in simulating the correct behavior of an ALSA device.
I decided to write a tool, which would help to test the userspace ALSA
programs (and the PCM middle layer as well) under unusual circumstances
to figure out how they would behave. So I came up with this Virtual PCM
Test Driver.
This new Virtual PCM Test Driver has several features which can be useful
during the userspace ALSA applications testing/fuzzing, or testing/fuzzing
of the PCM middle layer. Not all of them can be implemented using the
existing virtual drivers (like dummy or loopback). Here is what can this
driver do:
- Simulate both capture and playback processes
- Generate random or pattern-based capture data
- Inject delays into the playback and capturing processes
- Inject errors during the PCM callbacks
Also, this driver can check the playback stream for containing the
predefined pattern, which is used in the corresponding selftest to check
the PCM middle layer data transferring functionality. Additionally, this
driver redefines the default RESET ioctl, and the selftest covers this PCM
API functionality as well.
The driver supports both interleaved and non-interleaved access modes, and
have separate pattern buffers for each channel. The driver supports up to
4 channels and up to 8 substreams.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606193254.20791-2-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add documentation for the new Virtual PCM Test Driver. It covers all
possible usage cases: errors and delay injections, random and
pattern-based data generation, playback and ioctl redefinition
functionalities testing.
We have a lot of different virtual media drivers, which can be used for
testing of the userspace applications and media subsystem middle layer.
However, all of them are aimed at testing the video functionality and
simulating the video devices. For audio devices we have only snd-dummy
module, which is good in simulating the correct behavior of an ALSA device.
I decided to write a tool, which would help to test the userspace ALSA
programs (and the PCM middle layer as well) under unusual circumstances
to figure out how they would behave. So I came up with this Virtual PCM
Test Driver.
This new Virtual PCM Test Driver has several features which can be useful
during the userspace ALSA applications testing/fuzzing, or testing/fuzzing
of the PCM middle layer. Not all of them can be implemented using the
existing virtual drivers (like dummy or loopback). Here is what can this
driver do:
- Simulate both capture and playback processes
- Check the playback stream for containing the looped pattern
- Generate random or pattern-based capture data
- Inject delays into the playback and capturing processes
- Inject errors during the PCM callbacks
Also, this driver can check the playback stream for containing the
predefined pattern, which is used in the corresponding selftest to check
the PCM middle layer data transferring functionality. Additionally, this
driver redefines the default RESET ioctl, and the selftest covers this PCM
API functionality as well.
The driver supports both interleaved and non-interleaved access modes, and
have separate pattern buffers for each channel. The driver supports up to
4 channels and up to 8 substreams.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606193254.20791-1-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't need to change the numid at each time snd_ctl_rename_id() is
called, as the control element size itself doesn't change. Let's keep
the previous numid value.
Along with it, add a note about calling this function only in the
card init phase.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230606094035.14808-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the read event packet size in snd_seq_read() is defined by
client->midi_version value that is guaranteed to be zero if UMP isn't
enabled. But the static analyzer doesn't know of the fact, and it
still suspects as if it were leading to a potential overflow.
Add the more explicit check of CONFIG_SND_SEQ_UMP to determine the
aligned_size value for avoiding the confusion.
Fixes: 46397622a3 ("ALSA: seq: Add UMP support")
Reported-by: kernel test robot <lkp@intel.com>
Reported-by: Dan Carpenter <error27@gmail.com>
Closes: https://lore.kernel.org/r/202305261415.NY0vapZK-lkp@intel.com/
Link: https://lore.kernel.org/r/20230605144758.6677-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Include the FX bus map, without which the already present send routing
info would require looking up the documentation.
- Include the physical I/O channels as known to the driver
- Make the multi-channel capture map actually name the mapped input
channels rather than "FXBUS" (Audigy) or even "???" (SbLive)
- The latter two are omitted for E-MU cards, as their physical I/O is
routed through the FPGA
- While at it, make the "Card" field somewhat more useful
This includes de-duplicating the label tables between emuproc and emufx,
updating/improving the FX bus label table, and making the SB Live! 5.1
multi-track capture channel mapping hack data-driven.
Tested-by: Jonathan Dowland <jon@dow.land>
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The limits were appropriate only for the 2nd set.
FWIW, the channel count 4 for the 2nd set is suspicious as well - at
least P17V_PLAYBACK_FIFO_PTR actually has 8 channels, and comments on
HCFG2 hint at that as well. But all bitmasks are documented only for 4
channels. Anyway, rectifying that is out of scope for this patch.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On SB cards the number of captured channels is derived from the voice
mask mixer control. But for E-MU cards this wasn't actually "wired up",
so changing the mask would simply mess up the recording.
We could fix that, but the channel routing through the FPGA makes the
masking redundant. So instead we hide the control, and let the user
specify the PCM channel count the traditional way.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware can deal with primes up to 7 and power-of-two multiples
thereof; the limitation is reflected by the possible buffer sizes.
Note that setting the voice mask will not allow more than 16 channels
even on Sound Blaster Audigy anymore, as 32 seems a bit excessive (the
code overall appears to think so, just not in this case).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
