This is V2 of the patch, after feedback from Clemens and Daniel.
This patch adds SuperSpeed support to the USB drivers under sound/. It adds
tests for USB_SPEED_SUPER to the appropriate places that check for the USB
speed.
This patch has been tested with our SS USB3 device emulating a set of Yamaha
speakers and a Logitech microphone, but with the descriptors modified to add
USB3 support. It has also been tested with the real speakers and microphone,
to make sure that USB2 devices still work.
Signed-off-by: Paul Zimmerman <paulz@synopsys.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Its hardware is handled more fully by the new azt1605/azt2316 drivers.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a new driver for Aztech Sound Galaxy ISA soundcards based on the
AZT1605 and AZT2316 chipsets. It's constructed as two seperate drivers
for either chipset generated from the same source file, with (very)
minimal ifdeffery.
The drivers check the SB DSP version to decide if they are being loaded
for the right chip. AZT1605 returns 2.1 by default and AZT2316 3.1.
This isn't full-proof as the DSP version can actually be set through
software but it's close enough -- as far as I've been able to see, the
DSP version can not be stored in the EEPROM and the cards will therefore
startup with the defaults.
This distinction could (with the same success rate) also be used to
decide which chip we're looking at at runtime meaning a single, merged
driver is also an option but I feel it's actually nicer this way. A
merged driver would have to postpone translating the passed in resource
values to the card configuration until it knew which one it was looking
at and would need to postpone erring out on mpu_irq=10 for azt1605 and
mpu_irq=3 for azt2316.
The drivers have been tested on various cards. For snd-azt1605:
FCC-ID I38-MMSN811: Aztech Sound Galaxy Nova 16 Extra
FCC-ID I38-MMSN822: Aztech Sound Galaxy Pro 16 II
and for snd-azt2316:
FCC-ID I38-MMSN824: Aztech Sound Galaxy Pro 16 AB
FCC-ID I38-MMSN826: Trust Sound Expert DeLuxe Wave 32 (05201)
FCC-ID I38-MMSN830: Trust Sound Expert DeLuxe 16+ (05202)
FCC-ID I38-MMSN837: Packard Bell ISA Soundcard 030069
FCC-ID I38-MMSN846: Trust Sound Expert DeLuxe 16-3D (06300)
FCC-ID I38-MMSN847: Trust Sound Expert DeLuxe Wave 32-3D (06301)
FCC-ID I38-MMSN852: Aztech Sound Galaxy Waverider Pro 32-3D
826 and 846 were also marketed directly by Aztech and then known as:
FCC-ID I38-MMSN826: Aztech Sound Galaxy Waverider 32+
FCC-ID I38-MMSN846: Aztech Sound Galaxy Nova 16 Extra II-3D
Together, these cover the AZT1605 and AT2316A, AZT2316R and AZT2316-S
chipsets. All cards work fully -- full-duplex PCM, MIDI and FM. Full
duplex is a little flaky on some.
I38-MSN811 tends to not work in full-duplex but sometimes does with the
highest success rate being achieved when you first start the capture and
then a playback instead of the other way around (it's a CS4231-KL
codec).
The cards with an AD1845XP codec (my I38-MMSN826 and one of my
I38-MMSN830s) are also somewhat duplex-challenged. Sometimes full-duplex
works, sometimes not and this varies from try to try. This seems likely
to be a timing problem somewhere inside wss-lib.
I38-MMSN826 has an additional "ICS2115 WaveFront" wavetable synth
onboard that isn't supported yet. The wavetable synths on I38-MMSN847
and I38-MMSN852 are wired directly to the standard MPU-401 UART and the
AUX1 input on the codec and work without problem.
CD-ROM audio on the cards is routed to the codec "Line" input, Line-In
to its Aux input, and FM/Wavetable to its AUX1 input. I did not rename
the controls due to the capture source enumeration: I see that
capture-source overrides are hardcoded in wss-lib and this is just too
ugly to live.
Versus the old snd-sgalaxy driver these drivers add support for the
models without a configuration EEPROM (which are common), full-duplex,
MPU-401 UART and OPL3. In the future they might grow support for that
ICS2115 WaveFront synth on 826 and an hwdep interface to write to the
EEPROM on the models that have one.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the entries for 92HD87B1/3 and 92HD87B2/4 codecs.
These are compatible with existing 83xxx codecs.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Add missing codec IDs.
* Modify some existing codec names for discrete GPUs to match newly
added IDs. Note: existing names were a mixture of marketing and
engineering GPU names. Equally, there's no reason that codec IDs
have to be specific to a particular GPU or board, so identify
codecs in a less marketing-oriented fashion.
* Reformat codec ID table so it's easier to read, for me at least.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd-aloop module allows redirecting of the PCM playback in the
kernel back to the user space using the standard ALSA PCM capture API.
The module also allows time synchronization with another timing source
and notifications of playback stream parameter changes.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The Conexant CX20584 with 141f:5068 seems compatible with other
cxt5066 code. Just add the missing id.
Tested-by: Cristopher Camacho Leandro <ccamacho@linuxmail.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This provides a new model and pin config for the snd-hda-intel
92HD83XXX codec for hp laptop model dv7-4000, enabling the subwoofer.
Signed-off-by: Steven Eastland <seastland at gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I discovered tonight that ALSA no longer sets up a stream for the second ADC
provided by the Realtek ALC260 HDA codec. At some point alc_build_pcms()
started using stream_analog_alt_capture when constructing the second ADC
stream, but patch_alc260() was never updated accordingly. I have no idea
when this regression occurred. The trivial patch to patch_alc260() given
below fixes the problem as far as I can tell. The patch is against 2.6.35.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
Call the gpio reset platform function instead of using the flawed
ac97 functionality of the MPC5200(b)
From MPC5200B User's Manual:
"Some AC97 devices goes to a test mode, if the Sync line is high
during the Res line is low (reset phase). To avoid this behavior the
Sync line must be also forced to zero during the reset phase. To do
that, the pin muxing should switch to GPIO mode and the GPIO control
register should be used to control the output lines."
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6:
pcmcia: avoid buffer overflow in pcmcia_setup_isa_irq
pcmcia: do not request windows if you don't need to
pcmcia: insert PCMCIA device resources into resource tree
pcmcia: export resource information to sysfs
pcmcia: use struct resource for PCMCIA devices, part 2
pcmcia: remove memreq_t
pcmcia: move local definitions out of include/pcmcia/cs.h
pcmcia: do not use io_req_t when calling pcmcia_request_io()
pcmcia: do not use io_req_t after call to pcmcia_request_io()
pcmcia: use struct resource for PCMCIA devices
pcmcia: clean up cs.h
pcmcia: use pcmica_{read,write}_config_byte
pcmcia: remove cs_types.h
pcmcia: remove unused flag, simplify headers
pcmcia: remove obsolete CS_EVENT_ definitions
pcmcia: split up central event handler
pcmcia: simplify event callback
pcmcia: remove obsolete ioctl
Conflicts in:
- drivers/staging/comedi/drivers/*
- drivers/staging/wlags49_h2/wl_cs.c
due to dev_info_t and whitespace changes
So far, we reset the converter setups like the stream-tag, the
channel-id and format-id in prepare callbacks, and clear them in
cleanup callbacks. This often causes a silence of the digital
receiver for a couple of seconds.
This patch tries to delay the converter setup changes as much as
possible. The converter setups are cached and aren't reset as long
as the same values are used. At suspend/resume, they are cleared
to be recovered properly, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Indent the branch of an if.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@r disable braces4@
position p1,p2;
statement S1,S2;
@@
(
if (...) { ... }
|
if (...) S1@p1 S2@p2
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
if (p1[0].column == p2[0].column):
cocci.print_main("branch",p1)
cocci.print_secs("after",p2)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>