Merge tag 'sound-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've received a fairly wide range of changes at this time, including
  for ALSA and ASoC core, but all of them are rather small changes.

  Here are some highlights:

  ALSA / ASoC Core:
   - Fixes of inconsistent locking around control API helpers
   - A few new control API functions and cleanups
   - Workarounds for potential UAFs by delayed kobj releases
   - Unified PCM copy ops with iov_iter
   - Continued efforts for ASoC API cleanups

  ASoC:
   - An adaptor to allow use of IIO DACs and ADCs in ASoC which pulls in
     some IIO changes
   - Create a library function for intlog10() and use it in the NAU8825
     driver
   - Convert drivers to use the more modern maple tree register cache
   - Lots of work on the SOF framework, AMD and Intel drivers, including
     a lot of cleanup and new device support
   - Standardization of the presentation of jacks from drivers
   - Provision of some generic sound card DT properties
   - Support for AMD Van Gogh, AMD machines with MAX98388 and NAU8821,
     AWInic AW88261, Cirrus Logic CS35L36 and CS42L43, various Intel
     platforms including AVS machines with ES8336 and RT5663, Mediatek
     MT7986, NXP i.MX93, RealTek RT1017 and StarFive JH7110

  Others:
   - New test coverage including ASoC and topology tests in KUnit; this
     also involves enabling UML builds of ALSA since that's the default
     KUnit test environment which pulls in the addition of some stubs to
     the driver
   - More enhancement of pcmtest driver
   - A few fixes / enhancements of MIDI 2.0 UMP core
   - Using PCI definitions in allover HD-audio code
   - Support for Cirrus CS35L56 and TI TAS2781 HD-audio sub-codecs
   - CS35L41 HD-audio sub-codec improvements
   - Continued emu10k1 improvements"

* tag 'sound-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (693 commits)
  ALSA: pcm: Fix missing fixup call in compat hw_refine ioctl
  ASoC: dwc: i2s: Fix unused functions
  ALSA: usb-audio: Don't try to submit URBs after disconnection
  ALSA: emu10k1: add separate documentation for E-MU cards
  ALSA: emu10k1: more documentation updates
  ALSA: emu10k1: de-duplicate audigy-mixer.rst vs. sb-live-mixer.rst
  ALSA: ump: Fix -Wformat-truncation warnings
  ALSA: hda: Add missing dependency on CONFIG_EFI for Cirrus/TI sub-codecs
  ALSA: doc: Fix missing backquote in midi-2.0.rst
  ALSA: hda/realtek: Add quirk for mute LEDs on HP ENVY x360 15-eu0xxx
  ALSA: hda/tas2781: Switch back to use struct i2c_driver's .probe()
  ASoC: soc-core.c: Do not error if a DAI link component is not found
  ASoC: codecs: Fix error code in aw88261_i2c_probe()
  ASoC: audio-graph-card.c: move audio_graph_parse_of()
  ASoC: cs42l43: Use new-style PM runtime macros
  ALSA: documentation: Add description for USB MIDI 2.0 gadget driver
  ALSA: ump: Don't create unused substreams for static blocks
  ALSA: ump: Fill group names for legacy rawmidi substreams
  ALSA: usb-audio: Attach legacy rawmidi after probing all UMP EPs
  ALSA: ac97: Fix possible error value of *rac97
  ...

1 files changed, 226 insertions, 0 deletions

diff --git a/Documentation/sound/cards/emu-mixer.rst b/Documentation/sound/cards/emu-mixer.rst
new file mode 100644
index 000000000000..d87a6338d3d8
--- /dev/null
+++ b/Documentation/sound/cards/emu-mixer.rst
@@ -0,0 +1,226 @@
+==================================================
+E-MU Digital Audio System mixer / default DSP code
+==================================================
+
+This document covers the E-MU 0404/1010/1212/1616/1820 PCI/PCI-e/CardBus
+cards.
+
+These cards use regular EMU10K2 (SoundBlaster Audigy) chips, but with an
+alternative front-end geared towards semi-professional studio recording.
+
+This document is based on audigy-mixer.rst.
+
+
+Hardware compatibility
+======================
+
+The EMU10K2 chips have a very short capture FIFO, which makes recording
+unreliable if the card's PCI bus requests are not handled with the
+appropriate priority.
+This is the case on more modern motherboards, where the PCI bus is only a
+secondary peripheral, rather than the actual arbiter of device access.
+In particular, I got recording glitches during simultaneous playback on an
+Intel DP55 board (memory controller in the CPU), but had success with an
+Intel DP45 board (memory controller in the north bridge).
+
+The PCI Express variants of these cards (which have a PCI bridge on board,
+but are otherwise identical) may be less problematic.
+
+
+Driver capabilities
+===================
+
+This driver supports only 16-bit 44.1/48 kHz operation. The multi-channel
+device (see emu10k1-jack.rst) additionally supports 24-bit capture.
+
+A patchset to enhance the driver is available from `a GitHub repository
+<https://github.com/ossilator/linux/tree/ossis-emu10k1>`_.
+Its multi-channel device supports 24-bit for both playback and capture,
+and also supports full 88.2/96/176.4/192 kHz operation.
+It is not going to be upstreamed due to a fundamental disagreement about
+what constitutes a good user experience.
+
+
+Digital mixer controls
+======================
+
+Note that the controls work as attenuators: the maximum value is the neutral
+position leaving the signal unchanged. Note that if the same destination is
+mentioned in multiple controls, the signal is accumulated and can be clipped
+(set to maximal or minimal value without checking for overflow).
+
+Explanation of used abbreviations:
+
+DAC
+	digital to analog converter
+ADC
+	analog to digital converter
+LFE
+	low frequency effects (used as subwoofer signal)
+IEC958
+	S/PDIF
+FX-bus
+	the EMU10K2 chip has an effect bus containing 64 accumulators.
+	Each of the synthesizer voices can feed its output to these accumulators
+	and the DSP microcontroller can operate with the resulting sum.
+
+name='Clock Source',index=0
+---------------------------
+This control allows switching the word clock between interally generated
+44.1 or 48 kHz, or a number of external sources.
+
+Note: the sources for the 1616 CardBus card are unclear. Please report your
+findings.
+
+name='Clock Fallback',index=0
+-----------------------------
+This control determines the internal clock which the card switches to when
+the selected external clock source is/becomes invalid.
+
+name='DAC1 0202 14dB PAD',index=0, etc.
+---------------------------------------
+Output attenuation controls. Not available on 0404 cards.
+
+name='ADC1 14dB PAD 0202',index=0, etc.
+---------------------------------------
+Input attenuation controls. Not available on 0404 cards.
+
+name='Optical Output Mode',index=0
+----------------------------------
+Switches the TOSLINK output port between S/PDIF and ADAT.
+Not available on 0404 cards (fixed to S/PDIF).
+
+name='Optical Input Mode',index=0
+---------------------------------
+Switches the TOSLINK input port between S/PDIF and ADAT.
+Not available on 0404 cards (fixed to S/PDIF).
+
+name='PCM Front Playback Volume',index=0
+----------------------------------------
+This control is used to attenuate samples from left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM
+samples for 5.1 playback. The result samples are forwarded to the DSP 0 & 1
+playback channels.
+
+name='PCM Surround Playback Volume',index=0
+-------------------------------------------
+This control is used to attenuate samples from left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM
+samples for 5.1 playback. The result samples are forwarded to the DSP 2 & 3
+playback channels.
+
+name='PCM Side Playback Volume',index=0
+---------------------------------------
+This control is used to attenuate samples from left and right side PCM FX-bus
+accumulators. ALSA uses accumulators 14 and 15 for left and right side PCM
+samples for 7.1 playback. The result samples are forwarded to the DSP 6 & 7
+playback channels.
+
+name='PCM Center Playback Volume',index=0
+-----------------------------------------
+This control is used to attenuate samples from the center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM samples for 5.1 playback. The result samples
+are forwarded to the DSP 4 playback channel.
+
+name='PCM LFE Playback Volume',index=0
+--------------------------------------
+This control is used to attenuate samples from the LFE PCM FX-bus accumulator.
+ALSA uses accumulator 7 for LFE PCM samples for 5.1 playback. The result samples
+are forwarded to the DSP 5 playback channel.
+
+name='PCM Playback Volume',index=0
+----------------------------------
+This control is used to attenuate samples from left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
+stereo playback. The result samples are forwarded to the virtual stereo mixer.
+
+name='PCM Capture Volume',index=0
+---------------------------------
+This control is used to attenuate samples from left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the standard capture PCM device.
+
+name='Music Playback Volume',index=0
+------------------------------------
+This control is used to attenuate samples from left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the virtual stereo mixer.
+
+name='Music Capture Volume',index=0
+-----------------------------------
+These controls are used to attenuate samples from left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result is forwarded to the standard capture PCM device.
+
+name='Front Playback Volume',index=0
+------------------------------------
+This control is used to attenuate samples from the virtual stereo mixer.
+The result samples are forwarded to the DSP 0 & 1 playback channels.
+
+name='Surround Playback Volume',index=0
+---------------------------------------
+This control is used to attenuate samples from the virtual stereo mixer.
+The result samples are forwarded to the DSP 2 & 3 playback channels.
+
+name='Side Playback Volume',index=0
+-----------------------------------
+This control is used to attenuate samples from the virtual stereo mixer.
+The result samples are forwarded to the DSP 6 & 7 playback channels.
+
+name='Center Playback Volume',index=0
+-------------------------------------
+This control is used to attenuate samples from the virtual stereo mixer.
+The result samples are forwarded to the DSP 4 playback channel.
+
+name='LFE Playback Volume',index=0
+----------------------------------
+This control is used to attenuate samples from the virtual stereo mixer.
+The result samples are forwarded to the DSP 5 playback channel.
+
+name='Tone Control - Switch',index=0
+------------------------------------
+This control turns the tone control on or off. The samples forwarded to
+the DSP playback channels are affected.
+
+name='Tone Control - Bass',index=0
+----------------------------------
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+------------------------------------
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Master Playback Volume',index=0
+-------------------------------------
+This control is used to attenuate samples for all DSP playback channels.
+
+name='EMU Capture Volume',index=0
+----------------------------------
+This control is used to attenuate samples from the DSP 0 & 1 capture channels.
+The result is forwarded to the standard capture PCM device.
+
+name='DAC Left',index=0, etc.
+-----------------------------
+Select the source for the given physical audio output. These may be physical
+inputs, playback channels (DSP xx, specified as a decimal number), or silence.
+
+name='DSP x',index=0
+--------------------
+Select the source for the given capture channel (specified as a hexadecimal
+digit). Same options as for the physical audio outputs.
+
+
+PCM stream related controls
+===========================
+
+These controls are described in audigy-mixer.rst.
+
+
+MANUALS/PATENTS
+===============
+
+See sb-live-mixer.rst.