| Age | Commit message (Collapse) | Author |
|---|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Interrupt flag used for message handshake will be required for
stream interrupts, so conditionally compiled code without
HPI6205_NO_HSR_POLL defined can never be used; removing it.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Remove unused function.
Simplify hpi_alloc_control_cache.
Remove useless assignment to struct subsequently freed.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Create and use HPI_ERROR_DSP_COMMUNICATION _DSP_BOOTLOAD, rather than
backend-specific error codes (now returned as data with the error).
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
asihpi.c don't link playback and capture streams, there is too much
offset between them.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Some error codes had duplicate meanings. Just use one.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Reported samples_played from card may be inaccurate, so don't use it.
Update control names to be closer to alsa standard practice.
Also fixed some accidentally lowercased strings.
[Removed adriver.h inclusion for external module builds by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Remove many unused functions.
Update some message and cache structs.
Use pci info directly from pci_dev.
Allow control cache elements with variable size, and handle
large message/response from dsp.
hpi6000 and hpi6205: fix error path when adapter bootload fails.
hpimsgx.c get rid of code duplicated in hpicmn.c
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
All enum values numeric for easier finding, particularly error codes.
Remove many unused declarations.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Log HPI messages and responses in consistent numeric format,
which can be post-processed to get strings.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
rt->rate is an unsigned char so it's never equal to -1. It's not a huge
problem because the invalid rate is caught inside the call to
usb6fire_pcm_set_rate() which returns -EINVAL. But if we fix the test
then it prints out the correct error message so that's good.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
As requested by Takashi and Jaroslav, these arrays should not be in the
header file.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Inform users about the newly added support for RayDAT and AIO.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Incorporate changes by Florian Faber into hdspm.c. Code taken from
http://wiki.linuxproaudio.org/index.php/Driver:hdspe
Heavily reworked to mostly comply with the coding standard (whitespace
fixes, line width, C++ style comments)
The code was tested and confirmed to be working on RME RayDAT.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
What is working: Everything except SPDIF
- Hardware Master volume
- PCM 44-192kHz@24 bits, 6 channels out, 4 channels in (analog)
- MIDI in/out
- firmware loading after cold start
- phono/line switching
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
'cs4398_regs' in 'struct xonar_cs43xx' is an array of 'u8' with a size of
8. So, this code in sound/pci/oxygen/xonar_cs43xx.c::dump_d1_registers()
for (i = 2; i <= 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
will overrun the array when 'i == 8'.
I guess that what's needed to fix it is the trivial patch below, but I
must admit that I have no idea about this code, so I may very well be
wrong. Additionally, I have no way to actually test this, so all I know is
that the below compiles. Someone who actually knows this code should take
a look before anything is comitted - consider the below (not much more
than) a bug report.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
BugLink: http://bugs.launchpad.net/bugs/705323
Thinkpad Edge 14 has one more SSID that suffers from disabled auto-mute.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
sound/pci/hda/patch_realtek.c: In function ‘alc_apply_fixup’:
sound/pci/hda/patch_realtek.c:1724:14: warning: unused variable ‘modelname’
snd_printdd() is evaluated only when CONFIG_SND_DEBUG_VERBOSE=y.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
It seems working well with model=hp-bpc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This reverts commit 03b7a1ab557efe34e8f79b78660e514bd7374248.
This commit was mistakenly re-introduced. While the change is harmless
(as ALC887 uses patch_alc888() now), we should get rid of any wrong code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
WM8750 address is 0x1b, not 0x1a. Without this fix ALSA detects no sound
cards on Zipit
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Fix jack detection on Zipit Z2, otherwise it
disables headphones output when jack is connected
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
While I2S/TDM/AC97 DAI is built-in, others are compiled as modules,
SND_BF5XX_SOC_SPORT will be module, then DAI can't get some symbols.
Except that, SND_BF5XX_AC97 depends on SND_BF5XX_SOC_AC97 too.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
We don't want to use internal frame syncs otherwise we sometimes
get out of sync, so don't enable them when setting up the SPORT.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
We need to tweak how we query the active capture/playback state after
the recent overhauls of common code.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
|
|
One spot was missed in this driver when converting from
snd_soc_dai.private_data to snd_soc_dai_get_drvdata.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix initialization for HP 2011 notebooks
ALSA: hda - Add support for VMware controller
ALSA: hda - consitify string arrays
ALSA: hda - Add add multi-streaming playback for AD1988
ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
ASoC: WM8990: msleep() takes milliseconds not jiffies
ALSA : au88x0 - Limit number of channels to fix Oops via OSS emu
ALSA: constify functions in ac97
ASoC: WL1273 FM radio: Fix breakage with MFD API changes
ALSA: hda - More coverage for odd-number channels elimination for HDMI
ALSA: hda - Store PCM parameters properly in HDMI open callback
ALSA: hda - Rearrange fixup struct in patch_realtek.c
ALSA: oxygen: Xonar DG: fix CS4245 register writes
ALSA: hda - Suppress the odd number of channels for HDMI
ALSA: hda - Add fixup-call in init callback
ALSA: hda - Reorganize fixup structure for Realtek
ALSA: hda - Apply Sony VAIO hweq fixup only once
ALSA: hda - Apply mario fixup only once
ALSA: hda - Remove unused fixup entry for ALC262
|
|
The driver was using an initial value for the clock on the SPI bus
which was read from ICE1712 EEPROM,
ice->eeprom.data[ICE_EEP1_GPIO_STATE] & ICE1712_DELTA_AP_CCLK (0x02)
It appears some cards have it default high, some cards
have it default low. On my Delta 66 rev. E:
$ cat /proc/asound/M66/ice1712 | grep 'GPIO state'
GPIO state : 0x70 /* ICE1712_DELTA_AP_CCLK bit is zero */
On my Audiophile 2496:
$ cat /proc/asound/M2496/ice1712 | grep 'GPIO state'
GPIO state : 0xfe /* ICE1712_DELTA_AP_CCLK bit is one */
It must be raised before the first SPI write happens, or the write will
fail, leading to:
[ 23.248721] invalid CS8427 signature 0x0: let me try again...
I theorize that 4eb4550ab37d351ab0973ccec921a5a2d8560ec7
is no longer needed, it was a different way to workaround
the problem.
[fixed variable decleration by tiwai]
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
Fixes for HP 2011 notebooks: enable dock ports and disable BTL
initialization in the driver.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add the new PCI ID 0x15ad and device ID 0x1977 for VMware HDAudio
Controller.
[changed to use AZX_DRIVER_GENERIC by tiwai]
Signed-off-by: Bankim Bhavsar <bbhavsar@vmware.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Attached a patch which add a new model to support multi-streaming
playback for ad1988.
playback another stereo stream through the front panel headphone on
device 2 while playback through the speakers connected to rear panel
on device 0 at the same time.
Tested with ad1988a rev2 codec on asus P5B-V motherboard.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Changelog:
1. I2S module of EP93xx should be feed by 32bit DMA transfers. This is
hardware limitation and that's the way original Cirrus's driver worked.
This will fix distorted sound playback and make capture actually work in
present ep93xx drivers.
I've found, that author of code, on which modern ep93xx-i2s.c and
ep93xx-pcm.c are based, had faced this problem also in 2007:
http://blog.gmane.org/gmane.linux.ports.arm.cirrus/month=20070101/page=3
Now SoC code uses his developments, but not overcomes the hardware
issues. Some details from EP93xx users guide:
Both I2S transmitter and receiver have similar 16x32bit FIFO, where they
store 8 samples for both left and right channels. The FIFO is always
32bit wide and should be properly aligned if you use samples of other
width. Transmitter and receiver have configuration registers for
selection of I2S word length (16, 24, 32). They are I2STXWrdLen and
I2SRXWrdLen.
Yes, EP93xx DMA can do byte, word and quad-word transfers. The width for
transfers to and from peripherals is selected by particular module
configuration. Lucky AC97 module has such configuration: AC97RXCRx
registers, bit CM (Compact mode enable) switches between 16 and 32 bit
samples. AC97TXCRx registers have the same bits for transmitters.
ep93xx-ac97.c enables this compact mode and so has all the rights to use
S16_LE format.
No one has found such a configuration in I2S module until now in any
Cirrus manuals. I2S module always feeds it's 32bit wide FIFO with 32bit
samples consecutively for left and right channels. You cannot use 32-bit
DMA transfers to transfer two 16-bit samples.
So we can use two formats for AC97, but should remove all but S32_LE for
I2S. Always using 32 bit chunks is not a problem for I2S, the codec I
use uses less bits too (24), it's permitted by I2S standard.
In proposed patch formats list shortened to just S32_LE, this makes all
the DMA transactions right, while ALSA will do all sample format
translation for us.
2. Incorrect setting of LRCLK (2 times slower) in original ep93xx-i2s.c
masks the first problem.
DMA takes two 16 bit samples instead of one, overall sound speed seems
to be normal, but you get actually 4000 sampling rate instead of
requested 8000 and therefore some noise... This is also the reason why
the capture function not worked at all in this driver...
If we take a look into I2S specification, we will figure that LRCLK MUST
be equal to sample rate, if we are talking about stereo (in mono too,
but it's not our case at all).
In proposed patch SCLK and LRCLK rates are corrected, assuming we always
send 32 bits * 2 channels to codec.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
