This reverts commit c3b4eea262.
Since the recent firmware loader code supports caching at S3/S4 by
itself, we don't have to handle f/w caching in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Handle a potential dma_engine alloc error and fix the possible use of an
uninitialized status variable in dspxfr_one_seg(). Also correct the initial
sampling rate for Mic 1.
Update the module description.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the controls used for tuning the DSP effects.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the unsolicited response handler for incoming DSP responses and
jack detection reporting, and routines for reading the incoming DSP response.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the playback PCM open callback.
PCM stream setup and cleanup functions are added for use by PCM callbacks.
Delay stream cleanup if effects are on, to allow time for any effects tail to
finish.
Add the analog capture PCM callbacks.
Change the max channels of analog playback to 6.
Add two new PCMs: AMic2 and What-U-Hear.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the kcontrols for the DSP effects, playback and recording
source selection.
ca0132_is_vnode_effective() checks whether virtual node settings have
taken effect.
The control change helpers ca0132_pe_switch_set(), ca0132_voicefx_set()
and ca0132_cvoice_switch_set() are added to toggle playback / capture
DSP effects, ca0132_voicefx_info(), _get() and _put() are added for
input path DSP effect value access. The volume helpers are updated to
volume_info(), _get() and _set() to use the virtual nodes.
The redundant headphone and speaker switches and ct_extension function
are removed.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the framework to set effect parameters: ca0132_effects_set()
and ca0132_setup_defaults() are general functions for parameter setting and
initializing to default values. dspio_set_param() and dspio_set_uint_param()
are lower-level fns to simplify setting individual DSP parameters via an
SCP buffer transfer to the firmware.
The CA0132 chip parameter init code is added in ca0132_init_params().
In chipio_[write,read]_data(), the current chip address is auto-incremented
if no error has occurred.
ca0132_select_out() selects the current output. If autodetect is enabled,
use headphones (if jack detected) or speakers (if no jack).
ca0132_select_mic() selects the current mic in. If autodetect is enabled,
use exterior mic (if jack detected) or built-in mic (if no jack).
Init digital mic and switch between dmic and amic with ca0132_init_dmic(),
ca0132_set_dmic(). amic2 is initialized in ca0132_init_analog_mic2().
Finally, add verb tables for configuring DSP firmware.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds definitions and structs used for configuring DSP effects,
virtual nodes, effect tuning controls, and mixer control helpers.
The effect structs are also initialized.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tidy up and condense chipio_write_address|addx() functions.
Improve dspio_write_wait() to use jiffies for timeout calc.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add comments and descriptions to functions.
Bump dsp_free_ports() to below dsp_allocate_ports_format() to group the
alloc functions together for commenting.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the code needed to fetch the DSP binary image from the local
firmware install location and transfer it over to the chip using the new DSP
loader bus ops.
Actual DSP effect controls, parameters and mixers are to be included later.
- Add calls to new DSP loader system to transfer firmware to the hardware.
- Add chip read/write routines, DSP I/O, SCP packet format helper functions and
transfer DMA management.
- Add guard around DSP download to ensure loader config switch is enabled.
The general scheme for downloading the DSP is as follows:
1) If DSP firmware loader is enabled, ca0132_download_dsp() is called
to start the process.
2) Driver requests DSP image using request_firmware().
3) Driver sets up the streaming DMA for DSP image download
with dspload_image() and dspxfr_image(), which in turn calls the DSP loader
op snd_hda_codec_load_dsp_prepare() to ready the system.
4) DSP image will consist of 1 or more segments, each transferred in sequence
by a call to dspxfr_one_seg() and snd_hda_codec_load_dsp_trigger().
5) Once complete, the loader state is cleaned up with
snd_hda_codec_load_dsp_cleanup().
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pass DMA buffer pointers in calls to setup_bdle().
Add DSP loader callback routines to controller.
Add new DSP loader switch to Kconfig to turn off DSP firmware.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change for USB-audio disconnection race fixes introduced a
mutex deadlock again. There is a circular dependency between
chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a
device is opened during the disconnection operation:
A. snd_usb_audio_disconnect() ->
card.c::register_mutex ->
chip->shutdown_rwsem (write) ->
snd_card_disconnect() ->
pcm.c::register_mutex ->
pcm->open_mutex
B. snd_pcm_open() ->
pcm->open_mutex ->
snd_usb_pcm_open() ->
chip->shutdown_rwsem (read)
Since the chip->shutdown_rwsem protection in the case A is required
only for turning on the chip->shutdown flag and it doesn't have to be
taken for the whole operation, we can reduce its window in
snd_usb_audio_disconnect().
Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a precedence bug because | has higher precedence than ?:. This
code was cut and pasted and I fixed a similar bug a few days ago.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I don't think this works as intended. '|' higher precedence than ?: so
the bitwize OR "0 | (val & STR_MOST)" is a no-op.
I have re-written it to be more clear.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few small fixes plus a large but simple change for WM5102 which writes
out a bunch of register updates to the device when we enable the clock
as recommended following chip evaluation.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.7
A few small fixes plus a large but simple change for WM5102 which writes
out a bunch of register updates to the device when we enable the clock
as recommended following chip evaluation.
This is another variant of iMac 9,1 with a different codec SSID.
Reported-and-tested-by: Everaldo Canuto <everaldo.canuto@gmail.com>
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_put_volsw_sx function fails to update second control
if first control is updated by snd_soc_update_bits_locked.
Signed-off-by: Mukund Navada <navada@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
DAPM shutdown incorrectly uses "list" field of codec struct while
iterating over probed components (codec_dev_list). "list" field
refers to codecs registered in the system, "card_list" field is
used for probed components.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
There are uncovered cases whether the card refcount introduced by the
commit a0830dbd isn't properly increased or decreased:
- OSS PCM and mixer success paths
- When lookup function gets NULL
This patch fixes these places.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc269_toggle_power_output() was only use in ALC269VB. I rename it to
alc269vb_toggle_power_output().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback. It turned out that the problem is that we don't
wait until all URBs are killed.
This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181
Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The RayDAT reports the sync status of its inputs in consecutive bit
positions, so all we do in hdspm_s1_sync_check is to iterate over idx:
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
lock = (status & (0x1<<idx)) ? 1 : 0;
sync = (status & (0x100<<idx)) ? 1 : 0;
The index is given in kcontrol->private_value:
HDSPM_SYNC_CHECK("WC SyncCheck", 0),
HDSPM_SYNC_CHECK("AES SyncCheck", 1),
HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2),
HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3),
HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4),
HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5),
HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6),
HDSPM_SYNC_CHECK("TCO SyncCheck", 7),
HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8),
The patch corrects the indicated sync flags by passing the proper index
value to hdspm_s1_sync_check().
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the return value of cs42l52_set_fmt() when clock inversion is
not allowed and also remove the useless variable ret.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
[We had been assigning to ret but then ignoring the value we assgined
-- broonie]
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802
codec, correct the default configurations of speaker pins 0x24 and
0x33.
Reported-by: Massimo Del Fedele <max@veneto.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1802 codec provides the invalid connection lists of NID 0x24 and
0x33 containing the routes to a non-exist widget 0x3e. This confuses
the auto-parser. Fix it up in the driver by overriding these
connections.
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at
the point of the current line-out (i). When no valid path is found
for this output, this results in dac = 0, thus it creates a hole in
dac_nids[]. This confuses is_empty_dac() and trims the detected DAC
in later reference.
This patch fixes the bug by appending DAC properly to dac_nids[] in
via_auto_fill_adc_nids().
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When MCLK is supplied externally and BCLK and LRC are configured as outputs
(codec is master), the PLL values are only calculated correctly on the first
transmission. On subsequent transmissions, at differenct sample rates, the
wrong PLL values are used. Test for f_opclk instead of f_pllout to determine
if the PLL values are needed.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Several bug reports suggest that the forcibly resetting IEC958 status
bits is required for AD codecs to get the SPDIF output working
properly after changing streams.
Original fix credit to Javeed Shaikh.
BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361
Reported-by: Robin Kreis <r.kreis@uni-bremen.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add generic ESS vendor ID to pm_whitelist. This should fix suspend on
all Maestro-2 and Maestro-2E based PCI cards.
Tested on Terratec DMX and SF64-PCE2.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correctly enable the digital microphones with the right bits in the
right coeffecient registers on Cirrus CS4206/7 codecs. It also
prevents misconfiguring ADC1/2.
This fixes the digital mic on the Macbook Pro 10,1/Retina.
Based-on-patch-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The audio chipset used in Teradici's Tera2 host cards is the same as that in
the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards.
Signed-off-by: Lars R. Damerow <lars@pixar.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Playing 24-bit format file leads to channel swap on mx28 and the reason is that
the current driver performs one write/read to/from the SAIF_DATA register to
trigger the transfer.
This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register
and thus is capable of storing the 16-bit left and right channels, but for the
S24_LE case it can only store one channel, so in order to not lose the FIFO sync
an extra read/write is needed.
Reported-by: Dan Winner <DWinner@tc-helicon.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Dan Winner <DWinner@tc-helicon.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure we select the WM1250-EV1 as the current software system
configuration demands it.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai
Ian Minett
Dan Carpenter
Mark Brown
Mukund Navada
Misael Lopez Cruz
Kailang Yang
Adrian Knoth
Wei Yongjun
Charles Keepax
Eric Millbrandt
Ondrej Zary
Daniel J Blueman
Alexander Stein
Lars R. Damerow
Masanari Iida
Fabio Estevam
Dimitris Papastamos