Provide channel maps for individual stereo streams of ENS1370 and
ENS1371. Note that the configuration of ENS1370 uses the secondary
PCM as the front unlike ENS1371.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is better style as we acquire resources we will need before we go into
the ASoC card probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Better style as we get all the resources we need prior to starting the
ASoC level probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
MacBook Pro 10,1 needs a few adjustments to make it working:
- more COEF verbs
- some pin config overrides to disable the unused pin (0x0d, 0x12),
and fix the internal mic (0x0e)
In addition, it uses GPIO 1 and 3 like other MacBooks.
The internal digital mic on the machine is still problematic: it seems
that only the right channel is used and the left is always static.
This looks like a hardware design, so we need to cope in the software
side somehow...
The primary information and test were brought from Daniel J Blueman.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_pick_fixup() didn't check the case where the device mask bits
are set, typically used for SND_PCI_QUIRK_VENDOR() entries. Fix this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally the bogus period at BDL head was introduced as a workaround
for the mismatching position update at the period boundary, typically
seen on dmix. However, for applications like PulseAudio that don't
require period wake ups, this workaround is just superfluous. Thus
better to disable it when no_period_wakeup is given in hw_params.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit c20c5a841c changed some chipsets to
default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of
POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including
repeated sounds on my Asus laptop.
My hardware is Cougar Point which the commit log of
c20c5a841c mentions as tested so POS_FIX_COMBO
probably works in general but apparently it doesn't on Asus K53E therefore the
need for the quirk.
Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always
false and it will never do compress capture. The test for O_WRONLY is
also slightly off. The original test would consider "->flags =
(O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid.
I've also removed the pr_err() because that could flood dmesg.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In general, mono streams have no dedicated speaker assignment, thus
they should be rather marked as UNKNOWN position.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap)
, it is not necessary to provide codec->control_data anymore.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap)
, it is not necessary to provide codec->control_data anymore.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap)
, it is not necessary to provide codec->control_data anymore.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
cpu_dai is not in use in this function and just generates warning at
compile time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is better style since it has us obtaining all resources before we
try the ASoC probe. This change also fixes a potential issue where we
don't enable the regulators before trying to confirm the device ID which
could cause a failure during probe in some system configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
The core_intercon is added two times, remove the redundant one
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
struct fsi_master *master became member of struct fsi_priv from
71f6e0645b
(ASoC: sh_fsi: avoid using global variable)
So, master = NULL is not necessary on fsi_probe() now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pm_runtime_disable() error handling timing on fsi_probe() was wrong.
This patch fixes it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Long term all drivers should be using regmap directly. This is more
idiomatic and moves us towards the removal of the ASoC level cache
code.
The initialiasation of reserved register bits in probe() is slightly odd
as the defaults being written don't appear to match the silicon defaults
but the new code should have the same effect as the old code.
The watchdog code will now unconditionally do a mute and unmute when
resyncing but since we only sync when we are very sure there is something
to sync this should have no impact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Johannes Stezenbach <js@sig21.net>
This is better style as it ensures we don't try to do the ASoC probe
without required resources. Also convert to devm_ while we're at it,
saving a bit of code, and fix a leak of enable on error.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Johannes Stezenbach <js@sig21.net>
Ensure that we have confirmed that we've got the device in place before
we register with ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is better style since we acquire all needed resources before we try
to do the ASoC card probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_hda_codec_reset() calls restore_pincfgs() where the codec is
powered up again, which eventually tries to resume and initialize via
the callbacks of the codec. However, it's the place just after codec
free callback, thus no codec callbacks should be called after that.
On a codec like CS4206, it results in Oops due to the access in init
callback.
This patch fixes the issue by clearing the codec callbacks properly
after freeing codec.
Reported-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be used to enable additional control of the regulators.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The purpose of this flag is unclear. If the problem is that some machines
have broken misc/NO_PRESENCE bits, they should be fixed by pin fixups.
In addition, this causes jack detection functionality to be flawed on
the M31EI, where there are two jacks without jack detection (which is
properly marked as NO_PRESENCE), but due to ignore_misc_bit, these
jacks are instead being reported as being present but always unplugged.
BugLink: https://bugs.launchpad.net/bugs/939161
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The I2S controllers are programmed with an "attention" level of 4 DWORDs.
This must match the configuration passed to the DMA driver, so that when
they burst in data, they don't overflow the available FIFO space. Also,
the burst size is relevant to the destination for playback, and source
for capture, not vice-versa as originally written.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The VOLATILE flag was added to control elements by
snd_pcm_add_chmap_ctls() just because I didn't want to have a
side-effect of "alsactl restore". But now the set operation doesn't
allow to change the value unless the PCM stream is in PREAPRED state,
there is no reason to keep this flag. Let's rip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC650 has a channel swap option between surround and CLFE channels,
so we need to tweak the channel maps dynamically depending on the
register bit.
Now struct snd_ac97 can contain chmap pointers for playback and
capture. The driver may store these and let ac97 driver changing the
channel mapping dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... instead of the standard fixed channel maps.
The generic HDMI is based on the audio infoframe, and its configuration
can be selected via CA bits. Thus we need a translation between the
CA index and the verbose channel map list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although HD-audio allows pair-wise channel configurations, only the
fixed channel positions are used in this version. In future, this can
be changed and allow user to modify the channel positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SNDRV_CTL_ELEM_ACCESS_VOLATILE bit flag wasn't properly inherited
at creating control elements via snd_ctl_new1().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements the basic data types for the standard channel
mapping API handling.
- The definitions of the channel positions and the new TLV types are
added in sound/asound.h and sound/tlv.h, so that they can be
referred from user-space.
- Introduced a new helper function snd_pcm_add_chmap_ctls() to create
control elements representing the channel maps for each PCM
(sub)stream.
- Some standard pre-defined channel maps are provided for
convenience.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently the check for non-PCM stream state was added to the generic
HDMI driver code. But this check should be done rather to each pin
instead of each converter. Otherwise when a different converter is
assigned at the next open, the audio infoframe can be inconsistent
with the setup using the previous converter.
For fixing this issue, this patch moves the state of the current
non-PCM status from per_cvt to per_pin. (In addition an unused
argument cvt_nid is stripped from hdmi_setup_channel_mapping())
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the missing fine delayed time adjustment gives
strange error messages at each start of the playback stream, such as
delay: estimated 0, actual 352
delay: estimated 353, actual 705
These come from the sanity check in retire_playback_urb(). Before the
stream is activated via start_endpoints(), a few silent packets have
been already sent. And at this point the delay account is still in
the state as if the new packets are just queued, so the driver gets
confused and spews the bogus error messages.
For fixing the issue, we just need to check whether the received
packet is valid, whether it's zero sized or not.
Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pause and resume operations indicate that the stream can be
un-paused/resumed from the exact location they were paused/suspended.
This is not true for this driver, the pause and suspend triggers share
the same code path with stop, they flush all pending DMA transfers.
This drops all pending samples. The pause_release/resume triggers are
the same as start, except that prepare won't be called beforehand,
nothing will be enqueued to the DMA engine and nothing will happen (no
audio). Removing the pause flag will let apps know that it isn't
supported. Removing the resume flag will cause user space to call
prepare and start instead of resume, so audio will continue playing when
the system wakes up.
Before removing the pause and resume flags, I tested this on an exynos
5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a
write error. Suspend/resume testing led to the same result. Removing
the two flags fixes suspend/resume (since snd_pcm_prepare is called
again). And leads to a proper reporting of pause not supported.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
This uses already defined name of registers and makes code more readable.
Signed-off-by: Gaëtan Carlier <gcembed@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For multiple speaker outs, the names were previously
"Speaker,0", "Speaker,1", "Center"/"LFE", "Speaker,3". This is
inconsistent, confusing, and is not picked up correctly by PulseAudio.
Instead use "Front", "Surround", "Center"/"LFE", "Side" which
is more standard.
BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1046734
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with
vmaster hook in patch_sigmatel.c], the former Master volume control
was converted to PCM. This was supposed to be covered by the vmaster
control. But due to the lack of "PCM" slave definition, this didn't
happen properly. The patch fixes the missing entry.
Reported-by: Andrew Shadura <bugzilla@tut.by>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For HBR stream test, use straight channel mapping way.
when switched back to "speaker-test -c8", even the audio
infoframe is up-to-date, there should be correct channel mapping setup.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDMI channel remapping apparently effects HBR packets on Intel's chips.
For compressed non-PCM audio, use "straight-through" channel mapping.
For uncompressed multi-channel pcm audio, use normal channel mapping.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The array channel_allocations[] is an ordered list, add function to get
correct order by ca_index.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert 0865a75(ASoC: imx-ssi: Remove mono support).
The bug this patch is meant to solve doesn't occur in Visstrim_M10 boards.
Furthermore, after applying this patch sound in Visstrim_M10 is played
at slower rates.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since bypass paths aren't part of DAPM streams and we may not have any
DAPM streams there may not be anything that triggers a DAPM sync for
them. Mark all input and output widgets as dirty and then sync to do so
at the end of suspend and resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The OMAP2+ variant of McASP is different from Davinci variant w.r.to
some register offset.
Changes
- Add new MCASP_VERSION_3 to identify new variant. New DT compatible
"ti,omap2-mcasp-audio" to identify version 3 controller.
- The register offsets are handled depending on the version.
Note:
DMA parameters (dma fifo offset) are not updated and will be done later.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Playing a mono track on a mc13783 codec results in incorrect playback rate.
Remove mono support so that a mono track can be played correctly.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Gaëtan Carlier <gcembed@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the Tegra+WM8903 ASoC platform data header out of
arch/arm/mach-tegra, as a pre-requisite of single zImage.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix warning by using format specifier %zu for type size_t
Sparse warning:
sound/soc/codecs/wm0010.c:411:2: warning:
format ‘%d’ expects argument of type ‘int’,
but argument 4 has type ‘size_t’ [-Wformat]
Signed-off-by: Emil Goode <emilgoode@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're holding the wm0010->lock mutex when we goto err_core.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Part of commit (which patches sound/soc/omap/mcbsp.c file):
8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code
since the tree where it has been applied did not had the earlier patch:
d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration
which changed code around omap_mcbsp_6pin_src_mux().
Because of the missing part from 8fef626 the sound/soc/omap/mcbsp.c does
not compile in linux-next.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the main ALSA version number from the kernel ALSA driver.
The ALSA driver package release diverges from the upstream. This may
confuse users to see the same ALSA version for many kernel releases
and this version lost it's original purpose and connection.
The "ioctl" APIs have own version numbers, so the user space may check
for specific API changes only.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Playback Designs' USB devices have some hardware limitations on their
USB interface. In particular:
- They need a 20ms delay after each class compliant request as the
hardware ACKs the USB packets before the device is actually ready
for the next command. Sending data immediately will result in buffer
overflows in the hardware.
- The devices send bogus feedback data at the start of each stream
which confuse the feedback format auto-detection.
This patch introduces a new quirks hook that is called after each
control packet and which adds a delay for all devices that match
Playback Designs' USB VID for now.
In addition, it adds a counter to snd_usb_endpoint to drop received
packets on the floor. Another new quirks function that is called once
an endpoint is started initializes that counter for these devices on
their sync endpoint.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com>
Supported-by: Demian Martin <demianm_1@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
uinfo has been allocated in this function and should be
freed before leaving from the error handling cases.
spatch with a semantic match is used to found this problem.
(http://coccinelle.lip6.fr/)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SNDRV_MAIN_OBJECT_FILE hasn't done anything since the pre-git days, and
the only remaining reference occurs as a #define in sound/last.c. Drop
that last mention of it.
Signed-off-by: Josh Triplett <josh@joshtriplett.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit d56b9b9c46 ("The scheduled removal
of some OSS drivers") removed all traces of maui_boot.h from the tree.
Remove its entries in dontdiff and oss's .gitignore file.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Fix typo caused by recent commit (cf53756 - ASoC: davinci: davinci-pcm
does not need to be a plaform_driver)
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The if condition
if (!buf && !buf->area)
checks if the buf pointer is NULL and then dereferences it again to
check if the buffer area is NULL, resulting in possible NULL
dereference.
Signed-off-by: Prasad Joshi <prasadjoshi.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface")
saved us some unnecessary calls to snd_usb_set_interface() but ignored
the fact that there is at least one device out there which operates on
two endpoint in different interfaces simultaniously.
Take care for this by catching the case where data and sync endpoints
are located on different interfaces and calling snd_usb_set_interface()
between the start of the two endpoints.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Robert M. Albrecht <linux@romal.de>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to support devices with implicit feedback streaming models,
packet sizes are now stored with each individual urb, and the PCM
handling code which fills the buffers purely relies on the size fields
now.
However, calling snd_usb_audio_next_packet_size() for all possible
packets in an URB at once, prior to letting the PCM code do its job
does in fact not lead to the same behaviour than what the old code did:
The PCM code will break its loop once a period boundary is reached,
consequently using up less packets that it really could.
As snd_usb_audio_next_packet_size() implements a feedback mechanism to
the endpoints phase accumulator, the number of calls to that function
matters, and when called too often, the data rate runs out of bounds.
Fix this by making the next_packet function public, and call it from the
PCM code as before if the packet data sizes are not defined.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB
frame counter") were unfortunately lost during the refactoring of the
snd-usb driver in 3.5.
This patch adds them back, restoring the correct delay information
behaviour.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_endpoint_free() frees the structure that contains its argument.
Signed-off-by: Pavel Roskin <proski@gnu.org>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of pm_notify callback in snd_hda_codec_free() should be with
the check of the current state whether pm_notify(false) is called or
not, instead of codec->power_on check.
For improving the code readability and fixing this inconsistency,
codec->d3_stop_clk_ok is renamed to codec->pm_down_notified, and this
flag is set only when runtime PM down is called. The new name reflects
to a more direct purpose of the flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
k1212MinADCSens and k1212MaxADCSens are defined wrongly.
The max must be greater than the min by obvious reason.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46561
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_HDA_POWER_SAVE is no longer an experimental feature and its
behavior can be well controlled via the default value and module
parameter. Let's just replace it with the standard CONFIG_PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a codec provides its own set_power_state op, the D3-clock-stop
isn't checked correctly. And the recent changes for repeating the
state-setting operation isn't applied to such a codec, too.
This patch fixes these issues by moving the call of codec's own op to
the place where the generic power-set operation is done, and move the
power-state synchronization code out of
snd_hda_set_power_state_to_all() so that it can be called always at
the end of power-up/down sequence, and updates the D3 clock-stop flag
properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the HD-audio is removed, it leaves the refcounts when codecs are
powered up (usually yes) in the destructor. For fixing the unbalance,
and cleaning up the code mess, this patch changes the following:
- change pm_notify callback to take the explicit power on/off state,
- check of D3 stop-clock and keep_link_on flags is moved to the caller
side,
- call pm_notify callback in snd_hda_codec_new() and snd_hda_codec_free()
so that the refcounts are proprely updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in
PCM capture stream") fixed a scheduling-while-atomic bug that happened
when snd_usb_endpoint_start was called from the trigger callback, which
is an atmic context. However, the patch breaks the idea of the endpoints
reference counting, which is the reason why the driver has been
refactored lately.
Revert that commit and let snd_usb_endpoint_start() take care of the URB
cancellation again. As this function is called from both atomic and
non-atomic context, add a flag to denote whether the function may sleep.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
v2: Fixed result still wrong in the case of 512 KiB DRAM. Oops.
Applicable to 3.5.3 mainline.
In emu8000.c, size_dram determines the amount of memory on the sound card by
doing write/readback tests starting at 512 KiB and incrementing by 512 KiB.
On success, detected_size is updated to the successful address and testing
continues. On failure, the loop is immediately exited. The resulting
detected_size is 512 KiB too small except in two special cases:
1. If there is no memory, the initial 0 value of detected_size is used, which
is correct.
2. If the address space wraps around, detected_size is updated before the
bailout, so the result is correct.
The patch corrects all cases and was tested with an AWE64 Gold. Before:
EMU8000 [0x620]: 3584 Kb on-board memory detected
asfxload 4GMGSMT.SF2 (4174814 B) fails.
After:
EMU8000 [0x620]: 4096 Kb on-board memory detected
asfxload 4GMGSMT.SF2 succeeds.
I do not have a card with 512 KiB to test with, but by forcibly enabling the
added conditional I verified on the AWE64 Gold that it detects 512 KiB
(successfully reading from the first memory location) and does not hang the
card.
C.f. Bug 46451 https://bugzilla.kernel.org/show_bug.cgi?id=46451
Signed-off-by: David Flater <dave@flaterco.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is to rename the directory "ep93xx" in "cirrus".
Name more accurately reflects the manufacturer and allows to add
drivers not only for architecture ep93xx in this directory.
Patch not contain any functional changes.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All SAMSUNG ASoC needs SND_SOC_SAMSUNG configuration.
This patch change Kconfig to support all SAMSUNG ASoC.
Signed-off-by: Sangsu Park <sangsu4u.park@samsung.com>
Acked-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Both the schematics and practical testing show that the HP detect GPIO
is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio
should not specify to invert the signal.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <stable@vger.kernel.org> # v3.4 v3.5
These codecs seem reporting EPSS but require longer delay for the
proper D3 transition. For example, D3_STOP_CLOCK_OK bit won't be set
correctly even after D3.
In this patch, codec->epss flag is overridden for avoid the
misbehavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EPSS parameter should be static, so we can read it once and remember.
This also allows more easily to override the wrong EPSS capability
reported from a codec by changing the flag in the codec
initialization step.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use unsigned int to make clear that the codes required only for
modules will be reduced by the compiler optimization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add device tree probe for McASP driver.
Note:
DMA parameters are not populated from DT and will be done later.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Davinci McASP header & driver are shared by few OMAP platforms (like
TI81xx, AM335x). Splitting asp header into Davinci platform specific
header and Audio specific header helps to share them across platforms.
Audio specific defines is moved to to common
<linux/platform_data/davinci_asp.h> so that the header can be
accessed by all related platforms.
While here, correct the header usage (remove multiple header
re-definitions and unused headers) and remove platform names from
structures comments and enum. Also some some coding style errors.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Same as the commit 518de86 (ASoC: tegra: register 'platform' from DAIs,
get rid of pdev). It makes davinci-pcm not a platform_driver but helper
to register "platform", so that the platform_device for davinci-pcm can
be saved completely.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Device tree support for tlv320aic3x CODEC driver.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_spi_driver makes the code simpler by eliminating
module_init and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_spi_driver makes the code simpler by eliminating
module_init and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_spi_driver makes the code simpler by eliminating
module_init and module_exit calls.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we always need to have set and get callbacks for McBSP sidetone it
makes sense to combine the two macro to create the two callbacks.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To remove duplicated code from the driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When asked to add the ST controls warn only if the st_data is missing.
In this way we do not block the otherwise functional card to probe.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In ddevice tree booted kernel all device have unique name and their device
id is set to 0.
Use the mcbsp->id for checking to decide which control set we should add
for McBSP sidetone handling.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is more idimatic for modern drivers. Also fix a couple of return
codes while we're at it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Doesn't make any practical difference given that _SUSPEND and _OFF are
equivalent for the driver but it's what we're really doing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make it easier to integrate the management of the clock supplying the
WM0010 with DAPM by providing a dummy supply widget which supplies the
interface widgets, this can be connected to clock outputs by the machines.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With appropriate clocking configuration the WM0010 driver supports 44.1kHz
audio; enable that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Speyside platform by default has a WM0010 fitted. Now that we have
a public driver hook it up in the machine integration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means
that any DAPM context being updated will have the bias level automatically
set, including the card. We can't safely do this as the card callbacks are
called for each device context and so the management of the card bias is
more complex. Several multi-component cards rely on this behaviour.
Skip updates during the asynchronous run entirely. We should really do them
in the synchronous section but it's not 100% clear which values to pick as
the different DAPM contexts may have different bias levels.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated)
ensures that we update non-CODEC DAPM contexts but means that if a
CODEC has no set_bias_level() operation it'll not be updated. Fix
that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/isa/cmi8328.c: In function 'snd_cmi8328_remove':
sound/isa/cmi8328.c:416:24: error: 'cmi' undeclared (first use in this function)
sound/isa/cmi8328.c:416:24: note: each undeclared identifier is reported only once for each function it appears in
make[3]: *** [sound/isa/cmi8328.o] Error 1
Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch to support runtime PM introduced a bug:
Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk'
'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But
there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE.
This patch is to fix the bug.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM0010 is a compact digital signal processor that has been
highly optimised for low-power audio applications. Extensive memory
resources and core optimisation allow the device to manage all audio
processing algorithms efficiently and autonomously, while the host
processor sleeps or performs other tasks.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are some new WM1811 variants distinguished by both revision and
cust_id which need slightly different handling.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Runtime PM can bring more power saving:
- When the controller is suspended, its parent device will also have a chance
to suspend.
- PCI subsystem can choose the lowest power state the controller can signal
wake up from. This state can be D3cold on platforms with ACPI PM support.
And runtime PM can provide a gerneral sysfs interface for a system policy
manager.
Runtime PM support is based on current HDA power saving implementation. The user
can enable runtime PM on platfroms that provide acceptable latency on transition
from D3 to D0.
Details:
- When both power saving and runtime PM are enabled:
-- If a codec supports 'stop-clock' in D3, it will request suspending the
controller after it enters D3 and request resuming the controller before
back to D0. Thus the controller will be suspended only when all codecs are
suspended and support stop-clock in D3.
-- User IO operations and HW wakeup signal can resume the controller back to
D0.
- If runtime PM is disabled, power saving just works as before.
- If power saving is disabled, the controller won't be suspended because the
power usage counter can never be 0.
More about 'stop-clock' feature:
If a codec can support targeted pass-through operations in D3 state when there
is no BCLK present on the link, it will set CLKSTOP flag in the supported power
states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec
section 7.3.3.10 Power state and 7.3.4.12 Supported Power State.
[Fixed CONFIG_PM_RUNTIME dependency in hda_intel.c by tiwai]
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Device tree support for McBSP modules on OMAP2+ SoC.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can use the has_ccr flag to replace the cpu_is_omap* checks.
This provides future proof implementation and we do not need to update the
code if new OMAP revision starts to use the McBSP driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
NUM_LINKS is no longer in use by the code.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the feature to configure the CLKR/FSR mux on McBSP port with 6pin
configuration.
When moving to devicetree these callback can no longer be used in a clean
way anymore.
If a board require to change the 6pin port to work in 4pin setup it needs
to set up the mux in the board file.
For OMAP2/3:
u32 devconf0;
/* McBSP1 CLKR/FSR signal to be connected to CLKX/FSX pin */
devconf0 = omap_ctrl_readl(OMAP2_CONTROL_DEVCONF0);
devconf0 |= OMAP2_MCBSP1_CLKR_MASK | OMAP2_MCBSP1_FSR_MASK;
omap_ctrl_writel(devconf0, OMAP2_CONTROL_DEVCONF0);
For OMAP4:
u32 mcbsp_pad;
/* McBSP4 CLKR/FSR signal to be connected to CLKX/FSX pin */
mcbsp_pad = omap4_ctrl_pad_readl(OMAP2_CONTROL_DEVCONF0);
mcbsp_pad |= ((1 << 31) | (1 << 30));
omap4_ctrl_pad_writel(mcbsp_pad, OMAP2_CONTROL_DEVCONF0);
In case when the kernel is booted with DT blob the pinctrl-single will be
provided as soon as it is enabled on the platform.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The muxing is done at board level, no need to do it in the ASoC machine
driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the McBSP CLKS re-parenting code to ASoC driver from
arch/arm/mach-omap2.
The call fort the re-parenting has been already limited to OMAP2+ SoC in
the ASoC driver. There is no longer need to have callback function for it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sometimes the analogue circuitry connected to the microphone needs some
time to settle after power up. Allow systems to configure this delay in
the platform data, the driver will then insert the required delay during
power up of paths that involve the microphone.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of calling the jack sync in the init callback of each codec,
call it generically at initialization and resume. By calling it at
the last of resume sequence, a possible race between the jack sync and
the unsol event enablement in the current code will be closed, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes an issue with a machine where there were no speakers,
but GPIO0 had to be data=1 for the headphone to be functioning.
I'm not sure if we need a more advanced patch to solve all possible cases,
but if so, this patch would still provide a minor optimisation.
BugLink: https://bugs.launchpad.net/bugs/1040077
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
codec_dai is not used in the function.
sound/soc/soc-compress.c: In function ‘soc_compr_set_params’:
sound/soc/soc-compress.c:156:22: warning:
unused variable ‘codec_dai’ [-Wunused-variable]
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as
AudioExcel AV500.
It supports PCM playback and capture (full-duplex) through wss_lib, gameport,
OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected
to the MPU401 port and Aux 1 input internally which works too.
The CDROM interface is not supported (as the drivers for these CDROMs were
removed from the kernel some time ago).
A separate driver is needed because CMI8328 is completely different chip to
CMI8329/CMI8330. It's configured by magic registers (there's no PnP). Sound is
provided by a real WSS codec (CS4231A) and the SB part is just a SB Pro
emulation (for DOS games, useless for Linux).
When SB is enabled, the CMI8328 chip disables access to the WSS codec,
emulates SoundBlaster on one side and outputs sound data to the codec - so SB
and WSS can't work together with this card. The WSS codec can do full duplex
by itself so there's no need for crazy things like snd-cmi8330 does
(combining SB and WSS parts into one driver).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_card_als100_probe() does not set pcm field in struct snd_sb.
As a result, PCM is not suspended and applications don't know that they need
to resume the playback.
Tested with Labway A381-F20 card (ALS120).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
flush[_delayed]_work_sync() are now spurious. Mark them deprecated
and convert all users to flush[_delayed]_work().
If you're cc'd and wondering what's going on: Now all workqueues are
non-reentrant and the regular flushes guarantee that the work item is
not pending or running on any CPU on return, so there's no reason to
use the sync flushes at all and they're going away.
This patch doesn't make any functional difference.
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Russell King <linux@arm.linux.org.uk>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Ian Campbell <ian.campbell@citrix.com>
Cc: Jens Axboe <axboe@kernel.dk>
Cc: Mattia Dongili <malattia@linux.it>
Cc: Kent Yoder <key@linux.vnet.ibm.com>
Cc: David Airlie <airlied@linux.ie>
Cc: Jiri Kosina <jkosina@suse.cz>
Cc: Karsten Keil <isdn@linux-pingi.de>
Cc: Bryan Wu <bryan.wu@canonical.com>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Cc: Alasdair Kergon <agk@redhat.com>
Cc: Mauro Carvalho Chehab <mchehab@infradead.org>
Cc: Florian Tobias Schandinat <FlorianSchandinat@gmx.de>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: linux-wireless@vger.kernel.org
Cc: Anton Vorontsov <cbou@mail.ru>
Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: "James E.J. Bottomley" <James.Bottomley@HansenPartnership.com>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Eric Van Hensbergen <ericvh@gmail.com>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Steven Whitehouse <swhiteho@redhat.com>
Cc: Petr Vandrovec <petr@vandrovec.name>
Cc: Mark Fasheh <mfasheh@suse.com>
Cc: Christoph Hellwig <hch@infradead.org>
Cc: Avi Kivity <avi@redhat.com>
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the support to parse the compress dai's and then also adds the
soc-compress.c file while handles the compress stream operations, mostly analogus
to what is done in the soc-pcm.c and aditional handling of the compress
opertaions
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It was forgotten to initialize ret to the result of calling
snd_soc_dai_set_sysclk, unlike at the other calls in the same function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove unnecessary calls to devm_kfree and replace iounmap by devm_iounmap
(and use resource_size for the third argument). These changes make it
possible to remove the error-handling code at the end of
ux500_msp_i2s_init_msp, and all of the gotos become direct returns.
In the case of the second call to devm_kzalloc, the return variable ret was
not initialized. Here it is changed to a direct return of -ENOMEM.
A simplified version of the semantic match that finds the second problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Initialize ret on the second call to imx_audmux_v2_configure_port so that
the subsequent test checks that result and not the previous one.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the codec turn-on operation is canceled by the immediate
power-on, the driver left the power_transition flag as is.
This caused the persistent avoidance of power-save behavior.
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional updates for 3.6
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
The new LTO EXPORT_SYMBOL references symbols even without CONFIG_MODULES.
Since these functions are macros in this case this doesn't work.
Add a ifdef to fix the build.
Signed-off-by: Andi Kleen <ak@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... by calling the newly introduced snd_hda_power_sync().
I had to reimplement a wheel for adding the trigger at changing the
parameter -- the parameter set ops is overwritten to pass the integer
parameter, then trigger the power-state sync.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new helper function snd_hda_power_sync() to trigger the
power-saving manually. It's an inline function call to
snd_hda_power_save() helper function.
Together with this addition, snd_hda_power_up*() and
snd_hda_power_down() functions are inlined to a call of the same
snd_hda_power_save() helper function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's possible that these amps are settable somehow, e g through
secret codec verbs, but for now, don't create the controls (as
they won't be working anyway, and cause errors in amixer).
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1038651
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement suspend/resume support for AD1816 chips.
Tested with Terratec SoundSystem Base-1.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
struct snd_card_ad1816a is only set but the values are never used then.
Removing it allows struct snd_card's private_data to be used for
struct snd_ad1816a, simplifying the code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize ret before returning on failure, as done elsewhere in the
function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert a nonnegative error return code to a negative one, as returned
elsewhere in the function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize rc before returning on failure, as done elsewhere in the
function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize err before returning on failure, as done elsewhere in the
function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the first case, the second test of whether retval is negative is
redundant. It is dropped and the previous and subsequent tests are
combined.
In the second case, add an initialization of retval on failure of ioremap.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize retval before returning from a failed call to ioremap.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Speed comes from get_user() in audio_ioctl(). We use it to set the "s"
variable before clamping it to valid values so it could lead to a divide
by zero bug.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cpu dai and codec name are passed in through platform data.
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The capture volume increases with the register value so it shouldn't be
flagged as inverted.
Reported-by: Christoph Fritz <chf.fritz@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the microphone input source is not selectable as while there is
a DAPM widget it's not connected to anything so it won't be properly
instantiated. Add something more correct for the input structure to get
things going, even though it's not hooked into the rest of the routing
map and so won't actually achieve anything except allowing the relevant
register bits to be written.
Reported-by: Christop Fritz <chf.fritz@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
If DRC coefficients are not configured via platform data then add bytes
controls for them instead so they can be configured by applications. This
is the normal means of controlling things like this for newer systems, we
maintain compatibility with platform data to avoid disruption to existing
systems.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of blindly initializing a volume knob widget, first check
that there actually is a volume knob widget.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A PCM capture stream on usb-audio causes a scheduling-while-atomic
BUG, as reported in the bugzilla entry below. It's because
snd_usb_endpoint_start() is called at first at trigger START for a
capture stream, and this function contains the left-over EP
deactivation codes. The problem doesn't happen for a playback stream
because the function is called at PCM prepare time, which can sleep.
This patch fixes the BUG by moving the EP deactivation code into the
PCM prepare callback.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new omap-twl4030 handles the boards used the following drivers:
igep0020, omap3beagle, omap3evm and overo.
Remove these drivers since they are mostly identical and we already have
drop in replacement for all of them.
Note: Earlier patch added the needed code for the board files to retain the
audio support for boards I can identify that used one of the removed
drivers.
If I missed something please take a look at for example:
arch/arm/mach-omap2/board-omap3beagle.c on how add support for omap-twl4030
audio.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Machine driver to handle simple devices using twl4030 as audio codec.
The driver supports the following boards:
- Beagleboard or Devkit8000
- Gumstix Overo or CompuLab CM-T35/CM-T3730
- IGEP v2
- OMAP3EVM
All of these boards can be switched to use this driver since their setup is
identical.
Devicetree support for the omap-twl4030 machine driver also implemented.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added a simple support of automute for the front HP jack to AD1882
stack model. Such an addition is basically an exception -- we really
want to avoid the static quirk codes, but AD1882 parser isn't still
ready for moving to the BIOS auto-parser yet. So, as a quick fix, I
merged it for now.
In near future, we really need the big clean up of patch_analog.c to
move on to the auto-parser...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with a proper ifdef check of CONFIG_PM_SLEEP in hda_intel.c.
But other places in HD-audio driver are still marked with CONFIG_PM,
since these can be called for power-saving even without
CONFIG_PM_SLEEP.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Conexant devices (e g CX20590) have no mute capability on
their Beep widgets.
This patch makes sure we don't try setting mutes on those widgets.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't just notify for the bits we've updated, notify the full state of the
jack otherwise users might get confused by misleading reports.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As spec said, 1 indicates no copyright is asserted.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes the following build error:
In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0,
from arch/arm/plat-samsung/include/plat/dma-ops.h:17,
from arch/arm/plat-samsung/include/plat/dma.h:128,
from sound/soc/samsung/pcm.c:23:
arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8:
error: redefinition of ‘struct s3c2410_dma_client’
arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here
make[3]: *** [sound/soc/samsung/pcm.o] Error 1
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With commit 28d528c8 "ASoC: core: Remove pointless error on card
registration failure", the variable ret is no longer used in
soc_probe() and generates an unused variable warning during a build.
Signed-off-by: Jerry Snitselaar <dev@snitselaar.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The return value of snd_hda_param_read() is -1 for an error, otherwise
it's the supported power states of a codec.
The supported power states is a 32-bit value. Bit 31 will be set to 1
if the codec supports EPSS, thus making "sup" negative. And the bit
28:5 is reserved as "0".
So a negative value other than -1 shall be further checked.
Please refer to High-Definition spec 7.3.4.12 "Supported Power
States", thanks!
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When "Beep Playback Switch" had a different value on left and right
channels (such as muting left but not right, or vice versa), this
could result in the right channel being ignored.
This patch enables beep to be sounding from right channel only, and
also give correct result back to userspace (e g amixer).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function returns its own error codes instead of normal negative
error codes.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Wolfson Bells board takes submodules for various audio functions but
since the system integrations are virtually identical for most of them we
can support the overwhemling majority from the same machine driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Arizona devices support two output system clocks. Provide support for
configuring these via set_sysclk(). Once the clock API is more useful
we should migrate over to that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we fail to register the card we should say why somewhere else so there's
no point in repeating the same thing with less information.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the past when ASoC had a custom probe deferral mechanism people
complained about the logspam it generated and didn't want to know about
the fact that we were doing probe deferral so all the error messages for
it were at dev_dbg(), making diagnostics hard. Now that we have probe
deferral as an accepted thing and it's generating log messages anyway
there's no need to worry about this so upgrade the severity of all the
probe deferral sources to dev_err() so that they are displayed by default.
Also add one for missing aux_devs since there wasn't one.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH
Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This branch fixes the stall during probing the HD-audio driver when
the specified "patch" firmware doesn't exist. It's basically a long-
standing issue, but mostly harmless until the recent rework of
firmware loader base code.
For processing the firmware handling properly for built-in kernels,
implement an asynchronous firmware loading with
request_firmware_nowait(). This means that the codec probing is
deferred when the patch option is specified.
Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary work for the deferred probing for
request_firmware() errors at init.
This patch moves the call of request_firmware() to hda_intel.c, and
call it in the earlier stage of probing rather than
azx_probe_continue().
Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When CONFIG_PM is set but CONFIG_PM_SLEEP is unset,
SIMPLE_DEV_PM_OPS() ignores the given functions, and this leads to
compile warnings.
For avoiding this, simply check CONFIG_PM_SLEEP instead of CONFIG_PM.
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The core will bring the bias level up for us since we use idle_bias_off,
duplicating this may be harmful.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Add Runtime PM support to McASP host controller.
* Use Runtime PM API to enable/disable McASP clock.
This was tested on AM18x Board using suspend/resume
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Defines or parameters from <mach/mux.h> isn't used anywhere. Hence
remove the header include.
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FIFO should be flushed before it is enabled for the first time.
This fixes the I/O errors reported by the ASoC core on a fresh boot
Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To turn off pin control for the pin was tested, and helped against
this issue.
BugLink: https://bugs.launchpad.net/bugs/1034779
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Containing only a few really small/trivial fixes.
The only urgent fix is a regression fix of HDMI codec probing,
introduced in 3.6-rc1. The rest are HD-audio specific fixes and
a copule of minor bug fixes in PCM core and the old emu10k1.
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Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Containing only a few really small/trivial fixes. The only urgent fix
is a regression fix of HDMI codec probing, introduced in 3.6-rc1. The
rest are HD-audio specific fixes and a copule of minor bug fixes in
PCM core and the old emu10k1."
* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad
ALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y
ALSA: hda - remove redundant auto quirks for conexant 506x
ALSA: hda - remove quirk for Dell Vostro 1015
ALSA: hda - add dock support for Thinkpad X230
ALSA: hda - Fix regression of HDMI codec probing
ALSA: hda - add dock support for Thinkpad T430s
ALSA: emu10k1: Avoid access to invalid pages when period=1
ALSA: PCM: Fix possible memory leaks in the error path
CA0132 driver had some codes to handle the S/PDIF I/O, but the actual
setups of pins and converters were missing. Now the pins are added.
Also, fixed a few points triggering invalid codec verbs and mixer
elements since the digital I/O audio widgets on CA0132 have no amp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now with the workaround using codec->pcm_format_first flag, we can
clean up the home-baked codes in patch_ca0132.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduced a new flag to set up the PCM stream format at first before
the stream_id and channel tag. Some codecs (e.g. CA0132) seem
preferring this over stream_id -> format order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Playing a mono track results in incorrect playback rate, ie, the audio
is played at a faster rate.
Remove mono support in the driver by setting 'channes_min' to dual-channel
and this allows mono tracks to be played correctly.
Reported-by: Gaëtan Carlier <gcembed@gmail.com>
Tested-by: Gaëtan Carlier <gcembed@gmail.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The same ID is twice in the quirk table, so the second one is not used.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When CONFIG_SND_VERBOSE_PRINTK=y is set, the debug print in
hda_auto_parser.c looks really ugly like:
ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0
ALSA sound/pci/hda/hda_auto_parser.c:334 dig-out=0x12/0x0
ALSA sound/pci/hda/hda_auto_parser.c:335 inputs:
ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x11ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0x10
ALSA sound/pci/hda/hda_auto_parser.c:341
ALSA sound/pci/hda/hda_auto_parser.c:343 dig-in=0x13
Better to put one item at each line.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using devm_clk_get can make the code simpler and smaller.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The idle_bias_off flag is not configured for DAIs not mapped with a codec.
This causes the pm counter to be increased at probe time for the CPU dai
which unbalances the pm counter handling.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
slab.h header file was included twice.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai