#
923e0040 |
|
06-May-2024 |
Christos Margiolis <christos@FreeBSD.org> |
sound: Move chip.h contents to pcm/sound.h Sponsored by: The FreeBSD Foundation MFC after: 1 week Reviewed by: markj Differential Revision: https://reviews.freebsd.org/D44997
|
#
06a43763 |
|
16-Jan-2024 |
Christos Margiolis <christos@FreeBSD.org> |
snd_uaudio: prefix module declaration with "snd_" Although the module is compiled "snd_uaudio.ko", follow the rest of the sound modules' naming convention in the declaration as well. Sponsored by: The FreeBSD Foundation MFC after: 2 weeks Reviewed by: imp, emaste Differential Revision: https://reviews.freebsd.org/D43396
|
#
18d87fe4 |
|
16-Jan-2024 |
Christos Margiolis <christos@FreeBSD.org> |
snd_uaudio: provide information about the device name and attached driver Unlike the other sound drivers, snd_uaudio(4) doesn't provide information about the device's description and the driver it's attached to. A side-effect of this is that applications such as mixer(8), that fetch these strings through the OSS API's SNDCTL_CARDINFO ioctl will show a USB audio device as: pcm0:mixer: <USB Audio> at ? kld snd_uaudio This patch replaces the generic "USB Audio" description with the device's actual manufacturer and product strings, and the "at ?" string with the driver it's attached to: pcm0:mixer: <Focusrite Scarlett Solo USB> at uaudio0 kld snd_uaudio Sponsored by: The FreeBSD Foundation MFC after: 2 weeks Reviewed by: markj, emaste Differential Revision: https://reviews.freebsd.org/D43347
|
#
71625ec9 |
|
16-Aug-2023 |
Warner Losh <imp@FreeBSD.org> |
sys: Remove $FreeBSD$: one-line .c comment pattern Remove /^/[*/]\s*\$FreeBSD\$.*\n/
|
#
4d846d26 |
|
10-May-2023 |
Warner Losh <imp@FreeBSD.org> |
spdx: The BSD-2-Clause-FreeBSD identifier is obsolete, drop -FreeBSD The SPDX folks have obsoleted the BSD-2-Clause-FreeBSD identifier. Catch up to that fact and revert to their recommended match of BSD-2-Clause. Discussed with: pfg MFC After: 3 days Sponsored by: Netflix
|
#
2287364e |
|
06-May-2022 |
John Baldwin <jhb@FreeBSD.org> |
Remove pcm_devclass from DRIVER_MODULE invocations.
|
#
378503af |
|
01-Sep-2020 |
Mateusz Guzik <mjg@FreeBSD.org> |
sound: clean up empty lines in .c and .h files
|
#
86c9b3f3 |
|
30-Mar-2020 |
Hans Petter Selasky <hselasky@FreeBSD.org> |
Add support for multiple playback and recording devices per physical USB audio device. This requires some structural refactoring inside the driver, mostly about converting existing audio channel structures into arrays. The main audio mixer is provided by the first PCM instance. The non-first audio instances may only have a software mixer for PCM playback. Tested by: Horse Ma <Shichun.Ma@dell.com> MFC after: 1 week Sponsored by: Mellanox Technologies
|
#
718cf2cc |
|
27-Nov-2017 |
Pedro F. Giffuni <pfg@FreeBSD.org> |
sys/dev: further adoption of SPDX licensing ID tags. Mainly focus on files that use BSD 2-Clause license, however the tool I was using misidentified many licenses so this was mostly a manual - error prone - task. The Software Package Data Exchange (SPDX) group provides a specification to make it easier for automated tools to detect and summarize well known opensource licenses. We are gradually adopting the specification, noting that the tags are considered only advisory and do not, in any way, superceed or replace the license texts.
|
#
85bad582 |
|
21-Mar-2015 |
Hans Petter Selasky <hselasky@FreeBSD.org> |
The synchronisation value returned by the so-called feedback endpoint appears to be too inaccurate that it can be used to synchronize the playback data stream. If there is a recording endpoint associated with the playback endpoint, use that instead. That means if the isochronous OUT endpoint is asynchronus the USB audio driver will automatically start recording, if possible, to get exact information about the needed sample rate adjustments. In no recording endpoint is present, no rate adaption will be done. While at it fix an issue where the hardware buffer pointers don't get reset at the first device PCM trigger. Make some variables 32-bit to avoid problems with multithreading. MFC after: 3 days PR: 198444
|
#
61bfd867 |
|
30-Jan-2013 |
Sofian Brabez <sbz@FreeBSD.org> |
Use DEVMETHOD_END macro defined in sys/bus.h instead of {0, 0} sentinel on device_method_t arrays Reviewed by: cognet Approved by: cognet
|
#
a7d5f7eb |
|
19-Oct-2010 |
Jamie Gritton <jamie@FreeBSD.org> |
A new jail(8) with a configuration file, to replace the work currently done by /etc/rc.d/jail.
|
#
90da2b28 |
|
07-Jun-2009 |
Ariff Abdullah <ariff@FreeBSD.org> |
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.
|
#
3a3f90c6 |
|
23-Feb-2009 |
Andrew Thompson <thompsa@FreeBSD.org> |
Move the uaudio and ata-usb drivers into their correct locations.
|
#
d7f03759 |
|
19-Oct-2008 |
Ulf Lilleengen <lulf@FreeBSD.org> |
- Import the HEAD csup code which is the basis for the cvsmode work.
|
#
082f6383 |
|
17-Jun-2007 |
Ariff Abdullah <ariff@FreeBSD.org> |
Flush remaining malloc() cleanups (M_NOWAIT -> M_WAITOK).
|
#
bdfbdcec |
|
10-Jun-2007 |
Ariff Abdullah <ariff@FreeBSD.org> |
Filter/compress the amount of channel trigger. This should reduce much of lock/unlock contentions within the interrupt handler. Most of these drivers only need PCMTRIG_START or STOP (ABORT). Discussed with: scottl
|
#
e510f521 |
|
02-Jun-2007 |
Ariff Abdullah <ariff@FreeBSD.org> |
Use standard pcm_get/setflags() rather than dereferencing softc while enabling SD_F_SOFTPCMVOL or any flags.
|
#
7ab4fa8e |
|
16-Mar-2007 |
Ariff Abdullah <ariff@FreeBSD.org> |
[stage: 9/9] - SWAPLR quirk for (unknown, luckily it is mine) broken uaudio stick. Fixing by rewiring is impossible without damaging it. Luckily, we can fix it using "other" methods :) . - Add uaudio_get_vendor(), _product() and _release() in uaudio.c (currently used by uaudio_pcm quirk). - Implement CHANNEL_SETFRAGMENTS(). - Drop channel locking in few places where it is about to sleep somewhere. This should help eliminating illegal locking acquisition where the current thread is about to sleep, and also few deadlock cases. Dropping it right here is quite safe since it is already protected by CHN_F_BUSY flag and other threads won't bother to touch it. Solving other illegal locking issues are quite tricky without converting most usbd_do_request() calls to its equivalent _async() calls, which I intend to do it later after getting full test report from other people with different uaudio hardwares. - Fix memory leak issues during detach. This seems common to any drivers (notably emu10kx, csapcm?) with bridge functions.
|
#
b1d92216 |
|
26-Jan-2007 |
Ariff Abdullah <ariff@FreeBSD.org> |
Fix forever broken ua_chan_setblocksize() uninitialized return value which causing divide by zero panic in other places (notably chn_sync()).
|
#
7699548f |
|
28-Sep-2006 |
Ariff Abdullah <ariff@FreeBSD.org> |
Various fixups, especially for the upcomming High Definition Audio commit. 1) sys/dev/sound/pcm/sound.h sys/dev/sound/pcm/channel.c * Be more specific: SD_F_SOFTVOL -> SD_F_SOFTPCMVOL 2) sys/dev/sound/pcm/mixer.[ch] * Implement mix_setparentchild() mix_setrealdev() mix_getparent() mix_getchild() The purpose of these functions is implement relative volume adjustment, such as to tie two or more mixer device into a single logical device. Usefull for the upcoming HDA driver and few AC97 codec (such as AD1981B) where the master volume "vol" need to be implemented using this logical manner. 3) sys/dev/sound/pcm/ac97_patch.[ch] * Patch for AD1981B codec to enable (automuting) headphone jack sense. 4) sys/dev/sound/pcm/ac97.c * Implement proper logical master volume for AD9181B codec through various mix_set{parentchild,realdev}(). Tie both "ogain" (headphone volume) and "phone" (speaker/lineout) to a logical "vol". 5) sys/dev/sound/pcm/usb/uaudio_pcm.c * ditto, for "vol" -> { "pcm" }. MFC after: 1 month
|
#
f9dff1f9 |
|
13-Nov-2005 |
Alexander Leidinger <netchild@FreeBSD.org> |
Add support for 24/32 bit audio formats/conversion. It may be the case that you may hear some unwanted noise while playing back with 24/32 bit. This is a problem in the USB system. Explanation from Hans Petter Selasky: ---snip--- The current USB sound driver only uses one isochronous buffer, that is restarted when it is completed. This will lead to a short period of time, +1ms, where no sound data is sent to the external USB device. Depending on the load of your computer, this can be as much as 50ms. So the USB sound driver must use 2 isochronous transfers. At the beginning one will queue both. Then these are restarted on completion. This will result in a constant-rate data stream to the external sound device, a minimum sound buffer equal to the size of the isochronous buffer, and possibly the sound will reach your ears with less delay. Little delay is a result of constant data rate. Currently only my USB driver will support that. If one tries that with the USB driver in *BSD, then it will crash at the first moment one gets a buffer underrun. ---snip--- Submitted by: Kazuhito HONDA <kazuhito@ph.noda.tus.ac.jp> Mono-recording still not tested by: julian
|
#
f84e9487 |
|
02-Oct-2005 |
Alexander Leidinger <netchild@FreeBSD.org> |
Emulate pcm mixer controller for any uaudio device without it. Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
|
#
edffb4c8 |
|
18-Sep-2005 |
Alexander Leidinger <netchild@FreeBSD.org> |
Add the KLD to the sndstat info.
|
#
7f05203a |
|
27-Apr-2005 |
Julian Elischer <julian@FreeBSD.org> |
Add code from Kazuhito HONDA that allows the user to see the available modes in /dev/sndstat. e.g. pcm1: <USB Audio> at addr ? (0p/1r/0v channels duplex) mode 1:(input) 1ch, 16/16bit, pcm, 44100Hz mode 2:(input) 1ch, 16/16bit, pcm, 22050Hz mode 3:(input) 1ch, 16/16bit, pcm, 11025Hz mode 4:(input) 1ch, 16/16bit, pcm, 8000Hz
|
#
fe862a9b |
|
17-Apr-2005 |
Mathew Kanner <matk@FreeBSD.org> |
Fix a bug where we call pcm_getbuffersize twice. Pointed out by: Kazuhito HONDA <kazuhito at ph dot noda dot tus dot ac dot jp>
|
#
ece089c0 |
|
17-Apr-2005 |
Mathew Kanner <matk@FreeBSD.org> |
De-dma the uaudio <-> pcm bridge. We were not capable of doing DMA from this buffer anyway so the constraint that it had to be DMA capable only caused pain when devices failed to aquire the memory. Use a regular malloc instead with sndbuf_setup. Approved by: tanimura (mentor)
|
#
0224b85a |
|
17-Apr-2005 |
Julian Elischer <julian@FreeBSD.org> |
On record only devices, don't fail if we don't have a play channel. MFC after: 3 days
|
#
b7fd00d9 |
|
13-Apr-2005 |
Julian Elischer <julian@FreeBSD.org> |
The maximum allowable alloc is 16K not (16K-1). This whole section is actually overly restrictive and another patch is in the works.
|
#
d2b677bb |
|
01-Mar-2005 |
Warner Losh <imp@FreeBSD.org> |
Use BUS_PROBE_DEFAULT in preference to 0 and BUS_PROBE_LOW_PRIORITY in preference to some random negative number to allow other drivers a bite at the apple.
|
#
098ca2bd |
|
05-Jan-2005 |
Warner Losh <imp@FreeBSD.org> |
Start each of the license/copyright comments with /*-, minor shuffle of lines
|
#
587161d9 |
|
25-Dec-2004 |
Julian Elischer <julian@FreeBSD.org> |
Allow selection of a recording source on USB audio devices. PR: 75316 Submitted by: Kazuhito HONDA <kazuhito at ph dot noda dot tus dot ac dot jp> Obtained from: NetBSD plus changes MFC after: 2 weeks
|
#
65046f86 |
|
25-Dec-2004 |
Julian Elischer <julian@FreeBSD.org> |
Allow recording on at least some USB audio devices. PR: 75311 Submitted by: Kazuhito HONDA <kazuhito at ph dot noda dot tus dot ac dot jp> Obtained from: NetBSD plus changes MFC after: 2 weeks
|
#
d28a8145 |
|
24-Dec-2004 |
Julian Elischer <julian@FreeBSD.org> |
MFNetBSD: One of a set of patches submitted by Kazuhito HONDA to make the usb audio driver a lot more capable. PR: 75274 Submitted by: Kazuhito HONDA (kazuhito at ph dot noda dot tus dot ac dot jp) Obtained from: NetBSD (indirectly) MFC after: 2 weeks
|
#
0739ea1d |
|
15-Jul-2004 |
Seigo Tanimura <tanimura@FreeBSD.org> |
Rename the sound device drivers: - `sound' The generic sound driver, always required. - `snd_*' Device-dependent drivers, named after the sound module names. Configure accordingly to your hardware. In addition, rename the `snd_pcm' module to `sound' in order to sync with the driver names. Suggested by: cg
|
#
92fae6e7 |
|
31-May-2004 |
Takanori Watanabe <takawata@FreeBSD.org> |
Devclass have to be shared with same 'pcm' devclass, or unit management will corrupt.
|
#
f6b1c44d |
|
01-Jul-2003 |
Scott Long <scottl@FreeBSD.org> |
Mega busdma API commit. Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg. Lockfunc allows a driver to provide a function for managing its locking semantics while using busdma. At the moment, this is used for the asynchronous busdma_swi and callback mechanism. Two lockfunc implementations are provided: busdma_lock_mutex() performs standard mutex operations on the mutex that is specified from lockfuncarg. dftl_lock() is a panic implementation and is defaulted to when NULL, NULL are passed to bus_dma_tag_create(). The only time that NULL, NULL should ever be used is when the driver ensures that bus_dmamap_load() will not be deferred. Drivers that do not provide their own locking can pass busdma_lock_mutex,&Giant args in order to preserve the former behaviour. sparc64 and powerpc do not provide real busdma_swi functions, so this is largely a noop on those platforms. The busdma_swi on is64 is not properly locked yet, so warnings will be emitted on this platform when busdma callback deferrals happen. If anyone gets panics or warnings from dflt_lock() being called, please let me know right away. Reviewed by: tmm, gibbs
|
#
0586ff0d |
|
25-Nov-2002 |
Cameron Grant <cg@FreeBSD.org> |
if the list of supported formats is empty, fail the attach instead of panicing later. this is a band-aid pending further investigation. MFC After: 7 days Approved by: re
|
#
d807a231 |
|
21-Jul-2002 |
Nick Sayer <nsayer@FreeBSD.org> |
Add uaudio -- a USB audio device driver. This driver actually works slightly better on -stable than on -current (the system locks on detach on -current), so it should be MFC'd somewhat sooner. This driver currently points out a difficulty in the sound device framework. The PCM unregister routine is allowed to refuse the detach if the device is in use. In the case of a USB device, however, this unregistration is much more mandatory in nature, since the device is *actually* gone when this call is made. The sound subsystem really should not refuse an unregistration and should take its own steps to reject further I/O. As a result, if you detach a USB sound device while it is in use, you can expect a panic shortly thereafter. This device cannot currently record audio. Some routines are unwritten as of yet in uaudio.c to support recording. This device hangs my -current box on detach. I don't know why. This does not happen on my -stable machine. Obtained from: Hiroyuki Aizu MFC after: 2 weeks
|