Commit Graph

1263 Commits

Author SHA1 Message Date
mav
0c43daaa5e Add two Cirrus Logic codec IDs.
Add GPIO setting quirk for Apple MacBookPro5,5.

Submitted by:	ed
2009-11-26 20:25:57 +00:00
thompsa
49624dd6ce add support for MIDI devices without audio control stream.
Submitted by:	Hans Petter Selasky
2009-11-22 21:26:27 +00:00
thompsa
5e7f05ee90 remove volume alignment (was previously not correctly implemented)
Submitted by:	HPS
Reported by:	Jaakko Heinonen
2009-11-20 09:00:38 +00:00
mav
54fc684345 Add more codec IDs. 2009-11-13 21:06:33 +00:00
thompsa
267005bd55 Improve support for High-speed USB audio devices.
- fix issues regarding the mixer, where the interface number was not set in
  time.
- fix wrong use of resolution parameter.

Submitted by:	Hans Petter Selasky
2009-11-08 21:00:50 +00:00
mav
36fe7b55be Fix typo in previous commit.
Add Realtek ALC887 codec ID.
2009-09-30 11:05:12 +00:00
mav
c71d3f7c6e Add some bits of HDMI/DisplayPort support from later specification updates.
It may be not enough to make them work, but at least should give some
information about these beasts.
2009-09-29 09:36:38 +00:00
joel
1b7bf8e6f4 Move es137x.c and es137x.h to a 2-clause BSD license. Also move a few
comments.

Submitted by:	Joachim Kuebart
Approved by:	core, Russell Cattelan <cattelan@thebarn.com>
2009-09-22 13:23:59 +00:00
marius
0110876eec - According to Linux, the ALi M5451 can do 31-bit DMA instead of just
30-bit like the reset of the controllers supported by this driver.
  Actually ALi M5451 can be setup up to generate 32-bit addresses by
  setting the 31st bit via the accompanying ISA bridge, which allows
  it to work in sparc64 machines whose IOMMU require at least 32-bit
  DMA. Even though other architectures would also benefit from 32-bit
  DMA, enabling this bit is limited to sparc64 as bus_dma(9) doesn't
  generally guarantee that a low address of BUS_SPACE_MAXADDR_32BIT
  results in a buffer in the 32-bit range.
- According to Tatsuo YOKOGAWA's ali(4), the the DMA transfer size of
  ALi M5451 is fixed to 64k and in fact using the default size of 4k
- The 4DWAVE DX and NX require the recording buffer to be 8-byte
  aligned so adjust the bus_dma_tag_create(9) accordingly.
- Unlike the rest of the controllers supported by this driver, the
  ALi M5451 only has 32 hardware channels instead of 64 so limit the
  loop in tr_intr() accordingly. [1]

Submitted by:	yongari [1]
Reviewed by:	yongari (superset of what is committed)
MFC after:	3 days
2009-09-22 11:38:45 +00:00
mav
65e0b4852a Add NVidia MCP89 HDA controller IDs. 2009-09-09 04:48:41 +00:00
mav
01bd20ffac Add Intel 82801JD (one more ICH10) HDA controller ID.
Submitted by:	yongari
2009-09-09 04:36:56 +00:00
mav
487be1f13e Improve HDA controller capabilities logging. 2009-09-02 11:39:19 +00:00
alfred
04c45eb735 Remove redundant Giant reference. Giant will be dropped
automatically when the mutex argument is NULL.

Reported by: Various people
Submitted by: hps
2009-08-24 04:57:48 +00:00
jhb
9b0755de9f Temporarily revert the new-bus locking for 8.0 release. It will be
reintroduced after HEAD is reopened for commits by re@.

Approved by:	re (kib), attilio
2009-08-20 19:17:53 +00:00
attilio
7f42e47a67 Make the newbus subsystem Giant free by adding the new newbus sxlock.
The newbus lock is responsible for protecting newbus internIal structures,
device states and devclass flags. It is necessary to hold it when all
such datas are accessed. For the other operations, softc locking should
ensure enough protection to avoid races.

Newbus lock is automatically held when virtual operations on the device
and bus are invoked when loading the driver or when the suspend/resume
take place. For other 'spourious' operations trying to access/modify
the newbus topology, newbus lock needs to be automatically acquired and
dropped.

For the moment Giant is also acquired in some key point (modules subsystem)
in order to avoid problems before the 8.0 release as module handlers could
make assumptions about it. This Giant locking should go just after
the release happens.

Please keep in mind that the public interface can be expanded in order
to provide more support, if there are really necessities at some point
and also some bugs could arise as long as the patch needs a bit of
further testing.

Bump __FreeBSD_version in order to reflect the newbus lock introduction.

Reviewed by:    ed, hps, jhb, imp, mav, scottl
No answer by:   ariff, thompsa, yongari
Tested by:      pho,
                G. Trematerra <giovanni dot trematerra at gmail dot com>,
                Brandon Gooch <jamesbrandongooch at gmail dot com>
Sponsored by:   Yahoo! Incorporated
Approved by:	re (ksmith)
2009-08-02 14:28:40 +00:00
alfred
10ba0f5068 USB audio:
- code factoring patch from "Eygene Ryabinkin"
- P4 ID: 166149

Submitted by:	hps
Approved by:	re
2009-07-30 00:14:56 +00:00
mav
7218ddf669 Disable MSI by default for nVidia MCP55 chipset.
It is reported to be broken in the same way as MCP51.

PR:		kern/136429
Approved by:	re (kib)
2009-07-14 19:18:31 +00:00
ariff
7fc9bbac6d - Do aggresive saturation on various polynomial interpolators.
This dramatically pushing 99.9% interpolations and quantizations
  error _below_ -180dB on 32bit dynamic range, resulting extremely
  high quality conversion.
- Use BSPLINE interpolator for filter oversampling factor greater or
  equal than 64 (log2 6).

Approved by:	re (kib)
2009-07-14 18:53:34 +00:00
marcel
8233306b4d Isochronous transfers only have 1 frame buffer, but multiple
frame lengths. The frame buffer is at index 0.

Approved by:	re (kensmith)
Obtained from:	HPS
2009-07-12 16:50:32 +00:00
ariff
c6b231f0ca Rearrange shift operation to increase interpolation accuracy,
further reducing conversion artifacts and better worst case SNR.

Approved by:	re (kib)
2009-07-09 22:21:18 +00:00
ariff
488931f6d9 - Increase dynamic range of filter coefficients from 28bit to 30bit.
This cause dramatic effect in overall precision and conversion quality
  by pushing down most aliasing artifacts around -180 dB.

  Spectrogram analysis/comparison:

  	http://people.freebsd.org/~ariff/z_comparison/z_28vs30/

- Guard against possible 64bit overflow during accumulation process by
  slightly normalize and saturate sample and coefficient multiplication,
  possible during extreme 32bit downsampling (eg. 380KHz -> 8KHz) with
  custom preset that require more than ~7000 taps filter (which is
  overkill).

- Add knobs through FEEDER_RATE_PRESETS to set dynamic range of filter
  coefficients/accumulator and prefered polynomial interpolator:

  	COEFFICIENT_BIT:X
	(where 1 <= X <= 30, default: 30)

	ACCUMULATOR_BIT:X
	(where 32 <= X <=64, default: 58)

	INTERPOLATOR:I
	(where I = ZOH, LINEAR, QUADRATIC, HERMITE, BSPLINE,
 	           OPT32X, OPT16X, OPT8X, OPT4X, OPT2X)

Approved by:	re (kib)
2009-07-05 18:15:06 +00:00
thompsa
6a523f03ea Use the correct mutex in umidi_open()
Submitted by:	Hans Petter Selasky
Approved by:	re (kib)
2009-06-27 21:21:11 +00:00
kib
a7a5954511 Change the type of uio_resid member of struct uio from int to ssize_t.
Note that this does not actually enable full-range i/o requests for
64 architectures, and is done now to update KBI only.

Tested by:	pho
Reviewed by:	jhb, bde (as part of the review of the bigger patch)
2009-06-25 18:46:30 +00:00
mav
5acb1b62b2 Some DMA related changes:
- honor parent DMA tag limitations, as man page requires,
 - allow data buffer to be allocated within full 64bit address range, when
   support is announced by hardware,
 - add quirk, disabling 64bit addresses for broken chips, use it for MCP78.
2009-06-24 17:03:06 +00:00
ariff
98f843fb56 Slight comment fix. 2009-06-24 02:01:16 +00:00
thompsa
30004d4d8e Fix a typeo in the frame len function to unbreak the build, make it shorter
while I am here.
2009-06-23 06:00:31 +00:00
thompsa
74c6c20b93 - Make struct usb_xfer opaque so that drivers can not access the internals
- Reduce the number of headers needed for a usb driver, the common case is just   usb.h and usbdi.h
2009-06-23 02:19:59 +00:00
ariff
cf1afc1774 - Add a way to change filter oversampling factor through
FEEDER_RATE_PRESET "OVERSAMPLING_FACTOR:X .. .." where
  X = log2(oversampling factor).

- Lower down default filter oversampling factor from 128
  (log2 = 7) to 32 (log2 = 5), saving worth of 80 Kb.
  The use of better polynomial interpolator will raise
  its conversion quality/accuracy to match (or slightly
  better) with previous settings.

- Bump driver version.
2009-06-15 04:31:34 +00:00
ariff
548135de8c Remap type of polynomial interpolators for better polyphase
coefficients quality:
- Linear interpolator for oversampling factor larger and equal
  than 4096 (log2 = 12).
- Quadratic interpolator for oversampling factor larger and equal
  than 256 (log2 = 8).

Default oversampling factor (128 ~ log2 = 7) will use OPT32X, which
provides better accuracy.
2009-06-15 04:05:38 +00:00
thompsa
06303d491a s/usb2_/usb_|usbd_/ on all function names for the USB stack. 2009-06-15 01:02:43 +00:00
mav
553f130d15 Fix type of lowaddr variable. 2009-06-14 07:34:21 +00:00
ariff
2432cecd07 Remove custom KOBJMETHOD(), CHANNEL_DECLARE() and MIXER_DECLARE()
(enabled with SND_DEBUG) that was intended to provoke build failure
due to inconsistencies.
2009-06-11 09:06:09 +00:00
ariff
6a36125c72 Move machine dependant AFMT_* definition from sound.h
to global soundcard.h .
2009-06-10 03:56:24 +00:00
ariff
7517f12400 Fix compile time warning on sparc64, thanks to strict kobj signatures checking.
Noticed by:	bz
2009-06-08 23:24:01 +00:00
ariff
ad9205923a Fix powerpc build failure due to strict kobj signatures checking. 2009-06-08 08:10:52 +00:00
ariff
40420f94ac Fix build on sparc64.
Pointy hat:	ariff@
2009-06-07 23:38:16 +00:00
ariff
46d6a0d647 Bump driver revision (should have bumped it earlier). 2009-06-07 19:36:25 +00:00
ariff
e3faadaafe Sound Mega-commit. Expect further cleanup until code freeze.
For a slightly thorough explaination, please refer to
	[1] http://people.freebsd.org/~ariff/SOUND_4.TXT.html .

Summary of changes includes:

1 Volume Per-Channel (vpc).  Provides private / standalone volume control
  unique per-stream pcm channel without touching master volume / pcm.
  Applications can directly use SNDCTL_DSP_[GET|SET][PLAY|REC]VOL, or for
  backwards compatibility, SOUND_MIXER_PCM through the opened dsp device
  instead of /dev/mixer.  Special "bypass" mode is enabled through
  /dev/mixer which will automatically detect if the adjustment is made
  through /dev/mixer and forward its request to this private volume
  controller.  Changes to this volume object will not interfere with
  other channels.

  Requirements:
    - SNDCTL_DSP_[GET|SET][PLAY|REC]_VOL are newer ioctls (OSSv4) which
      require specific application modifications (preferred).
    - No modifications required for using bypass mode, so applications
      like mplayer or xmms should work out of the box.

  Kernel hints:
    - hint.pcm.%d.vpc (0 = disable vpc).

  Kernel sysctls:
    - hw.snd.vpc_mixer_bypass (default: 1).  Enable or disable /dev/mixer
      bypass mode.
    - hw.snd.vpc_autoreset (default: 1).  By default, closing/opening
      /dev/dsp will reset the volume back to 0 db gain/attenuation.
      Setting this to 0 will preserve its settings across device
      closing/opening.
    - hw.snd.vpc_reset (default: 0).  Panic/reset button to reset all
      volume settings back to 0 db.
    - hw.snd.vpc_0db (default: 45).  0 db relative to linear mixer value.

2 High quality fixed-point Bandlimited SINC sampling rate converter,
  based on Julius O'Smith's Digital Audio Resampling -
  http://ccrma.stanford.edu/~jos/resample/.  It includes a filter design
  script written in awk (the clumsiest joke I've ever written)
    - 100% 32bit fixed-point, 64bit accumulator.
    - Possibly among the fastest (if not fastest) of its kind.
    - Resampling quality is tunable, either runtime or during kernel
      compilation (FEEDER_RATE_PRESETS).
    - Quality can be further customized during kernel compilation by
      defining FEEDER_RATE_PRESETS in /etc/make.conf.

  Kernel sysctls:
    - hw.snd.feeder_rate_quality.
      0 - Zero-order Hold (ZOH).  Fastest, bad quality.
      1 - Linear Interpolation (LINEAR).  Slightly slower than ZOH,
          better quality but still does not eliminate aliasing.
      2 - (and above) - Sinc Interpolation(SINC).  Best quality.  SINC
          quality always start from 2 and above.

  Rough quality comparisons:
    - http://people.freebsd.org/~ariff/z_comparison/

3 Bit-perfect mode.  Bypasses all feeder/dsp effects.  Pure sound will be
  directly fed into the hardware.

4 Parametric (compile time) Software Equalizer (Bass/Treble mixer). Can
  be customized by defining FEEDER_EQ_PRESETS in /etc/make.conf.

5 Transparent/Adaptive Virtual Channel. Now you don't have to disable
  vchans in order to make digital format pass through.  It also makes
  vchans more dynamic by choosing a better format/rate among all the
  concurrent streams, which means that dev.pcm.X.play.vchanformat/rate
  becomes sort of optional.

6 Exclusive Stream, with special open() mode O_EXCL.  This will "mute"
  other concurrent vchan streams and only allow a single channel with
  O_EXCL set to keep producing sound.

Other Changes:
    * most feeder_* stuffs are compilable in userland. Let's not
      speculate whether we should go all out for it (save that for
      FreeBSD 16.0-RELEASE).
    * kobj signature fixups, thanks to Andriy Gapon <avg@freebsd.org>
    * pull out channel mixing logic out of vchan.c and create its own
      feeder_mixer for world justice.
    * various refactoring here and there, for good or bad.
    * activation of few more OSSv4 ioctls() (see [1] above).
    * opt_snd.h for possible compile time configuration:
      (mostly for debugging purposes, don't try these at home)
        SND_DEBUG
        SND_DIAGNOSTIC
        SND_FEEDER_MULTIFORMAT
        SND_FEEDER_FULL_MULTIFORMAT
        SND_FEEDER_RATE_HP
        SND_PCM_64
        SND_OLDSTEREO

Manual page updates are on the way.

Tested by:	joel, Olivier SMEDTS <olivier at gid0 d org>, too many
          	unsung / unnamed heroes.
2009-06-07 19:12:08 +00:00
thompsa
c670960df0 revert r162516. We only support 1 or 2 channels per stream
which reflects mono and stereo.

Submitted by:	Hans Petter Selasky
2009-06-04 21:59:28 +00:00
mav
960d9e52b8 Comment out old Realtek ALC883 quirk, that was disabling phantop power on
mic inputs. I have no idea what for it was made that time, but now I have
several reports that it should be removed to make microphones work. If
this quirk is still required for some systems then they should be identified
and specified explicitly.
2009-06-01 13:13:47 +00:00
thompsa
44c17bdf07 s/usb2_/usb_/ on all typedefs for the USB stack. 2009-05-29 18:46:57 +00:00
thompsa
af6fb4f3d2 s/usb2_/usb_/ on all C structs for the USB stack. 2009-05-28 17:36:36 +00:00
thompsa
b36d17c973 Provide a workaround for USB devices that do not support mono or stereo
operation by overriding the channel count.

Submitted by:	Hans Petter Selasky
Reported by:	MIHIRA Sanpei Yoshiro
2009-05-27 19:45:04 +00:00
joel
3fb43873ed Slightly adjust copyright text.
Approved by:	Hannu Savolainen <hannu@opensound.com>
2009-05-27 18:17:58 +00:00
joel
0bcd1e27c7 Slightly adjust copyright text.
Approved by:	luigi
2009-05-27 18:16:53 +00:00
joel
9150148ab8 Separate comments from the license text. 2009-05-27 18:13:15 +00:00
thompsa
406855c9ff Fix a few variable renames of usb2_mode outside dev/usb. 2009-05-21 02:09:12 +00:00
joel
6cadac237d Slightly adjust copyright text.
Approved by:	matk
2009-05-20 18:38:43 +00:00
joel
465a3a97f7 Remove license clauses 3 and 4 as per rev. 1.65 of midi.c in NetBSD.
Approved by:	matk
2009-05-20 18:34:26 +00:00
joel
ecf58c5644 Remove license clauses 3 and 4 as per rev. 1.65 of auvia.c in NetBSD. 2009-05-20 18:31:11 +00:00