ALSA: asihpi - Don't leak firmware if mem alloc fails
We leak the memory allocated to 'firmware' when we fail to
release_firmware() after a kmalloc() failure in hpi_dsp_code_open().
This patch should take care of the leak.
CC [M] sound/soc/txx9/txx9aclc.o
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_new':
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: error: 'card' undeclared (first use in this function)
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: note: each undeclared identifier is reported only once for each function it appears in
make[5]: *** [sound/soc/txx9/txx9aclc.o] Error 1
Adrian Knoth [Fri, 29 Jul 2011 01:11:04 +0000 (03:11 +0200)]
ALSA: hdspm - Add firmware revision 0xcc for RME MADI
Apparently, there are multiple old firmware revisions in the wild for
the PCI RME MADI cards. Just add them to the list of supported devices
and treat them like their modern counterparts.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adrian Knoth [Fri, 29 Jul 2011 01:11:03 +0000 (03:11 +0200)]
ALSA: hdspm - Fix reported external sample rate on RME MADI and MADIface
In slave mode, the card can only detect the base frequency (32..48kHz)
on the MADI link (exception: 96k frames), so the real external sample
rate is this base frequency multiplied by 1, 2 or 4 depending on the
speed mode.
This patch enables 64..192kHz sample rates in clock slave mode, which
failed before due to an alleged sample rate mismatch between the MADI
link (e.g., 48kHz) and the application in DS/QS mode (e.g., 96kHz,
192kHz).
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adrian Knoth [Fri, 29 Jul 2011 01:11:02 +0000 (03:11 +0200)]
ALSA: hdspm - Provide MADI speed mode selector on RME MADI and MADIface
When running in slave mode (no clock master), there is no way to
determine the real wirespeed on the MADI link (single/double/quad
speed). Like physical gear, simply provide the user with a tristate
switch to select the appropriate format.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda - Fix duplicated DAC assignments for Realtek
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment. The problem appears in 3.0 kernel code.
ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.
Reported-and-tested-by: Oliver Neukum <oneukum@suse.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda - Add support for vref-out based mute LED control on IDT codecs
This patch also registers all necessary callbacks to support mute LED
only when such control is enabled. And it keeps codec AFG in D0 or D1
state all the time when aggressive power managemnt is enabled for vref-out
control (and mute LED) work correctly.
Tim Howe [Fri, 22 Jul 2011 21:41:00 +0000 (16:41 -0500)]
ALSA: hda - Cirrus Logic CS421x support
This update includes the changes necessary for supporting the
CS421x family of codecs. Previously this file only supported
the CS420x family of codecs.
This file also contains init verbs to correct several issues in
the CS421x hardware.
Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.
Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.
[Fix const usages and adaption for new APIs by tiwai]
Signed-off-by: Tim Howe <tim.howe@cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the macros depending on snd_mask_min() and co out of pcm.h into
pcm_params.h. Otherwise using some params_*() macros will give comiple
errors without inclusion of pcm_params.h.
Also use hw_param_interval_c() and hw_param_mask_c() for const pointer.
Reported-by: Tim Blechmann <tim@klingt.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.
This patch adds a new codec ops, set_power_state() to override the system
default function. For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.
As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.
Add a new ops, post_suspend(), which is called after suspend() ops is
performed. This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.
ALSA: hda - Make sure mute led reflects master mute state
This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.
ALSA: hda - Fix invalid mute led state on resume of IDT codecs
Codec state is not restored immediately on resume but on the first
access when power-save is enabled. That leads to an invalid mute led
state after resume until either sound is played or some control is
changed. This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required. And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.
ALSA: hda - Fix DAC filling for multi-connection pins in Realtek parser
Fix a regression in the DAC filling code in patch_realtek.c. The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.
HPI Version is used to check for firmware compatibility.
This version will accept 4.08.xx released firmware,
and will also accept 4.09.xx beta firmware
ALSA: asihpi - Remove spurious adapter index check
Subsystem requests don't have or need a valid adapter index.
The adapter index is already checked further on, before it is used to index
the adapters array. (Reverts 4a122c10f)
ALSA: asihpi - Remove controlex structs and associated special data transfer code
Some cobranet control data would not fit in an original HPI message.
Now that HPI is able to transfer larger messages, this special handling
is no longer required.
David G Turner [Thu, 21 Jul 2011 17:00:57 +0000 (19:00 +0200)]
ALSA: usb-audio - Add quirk for Roland / BOSS BR-800
Add support for Roland/BOSS BR-800 (0582:011e) to snd-usb-audio driver.
This allows playback and recording, which has been tested and found to
work. The third interface should be MIDI (MTC/SMPTE?) for DAW interface
and is set as per ME-25, but this has not been tested. SDHC card access
is already supported by usb-storage for Backup/Rhythm Editor/Wave
Convertor mode which should not conflict with this.
Signed-off-by: David G Turner <dgturner@iee.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
This patch fixes non-working indep-HP control on VT1708* codecs.
The problems are that via_independent_hp_put() wasn't fixed to follow
the recent change of three HP paths, and hp_indep_path didn't contain
the amp nids of mixer elements.
Together with the fixes, a few code clean-ups are done.
Sangbeom Kim [Wed, 20 Jul 2011 08:07:12 +0000 (17:07 +0900)]
ASoC: SAMSUNG: Modify I2S driver to support idma
Previously, I2S driver only can support system dma.
In this patch, i2s driver can support internal dma too.
IDMA h/w configuration is initialized on idma.c
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com> Acked-by: Liam Girdwood <lrg@ti.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.
Mark Brown [Wed, 20 Jul 2011 12:50:10 +0000 (13:50 +0100)]
ASoC: Acknowledge WM8962 interrupts before acting on them
This closes the small race between a status being read in response to an
interrupt and clearing the interrupt, meaning that if the status changes
between those periods we might not get a reassertion of the interrupt.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Wolfram Sang [Sun, 17 Jul 2011 16:00:26 +0000 (18:00 +0200)]
ASoC: core: make comments fit the code
In one comment, cpu_dai was mentioned although codec_dai was used in the
code. Also, fix the name for the card dai list which has no seperation
into card_dai and codec_dai.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Mon, 18 Jul 2011 04:17:13 +0000 (13:17 +0900)]
ASoC: Mark cache as dirty when suspending
Since quite a few drivers are not managing to flag the cache as needing
to be resynced after suspend and it's a reasonable thing to do flag the
cache as needing sync automatically when suspending.
The expectation is that systems will mainly only keep the CODEC powered
when doing audio through the CODEC so we won't actually suspend the
device anyway; drivers which want to can override this behaviour when
they resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Cc: stable@kernel.org
ALSA: hda - Switch HP DAC dynamically with indep-HP switch for VIA
This patch changes the behavior of independent-HP enum switch. Now
instead of returning a busy error, the driver switches dynamically the
stream of the HP (and shared) DACs according to the current mode.
The logic is similar like the dual-mic ADC switch, but a bit more
complicated because of the presence of shared DAC.
Together with the change, a mutex is introduced to protect against the
possible races for the indep-HP mode setting.
ALSA: hda - Implement dynamic loopback control for VIA codecs
This patch adds the dynamic control of analog-loopback for VIA codecs.
When the loopback is enabled, the inputs from line-ins and mics are
mixed with the front DAC, and sent to the front outputs. The very same
input is routed to the headhpones and speakers in loopback mode.
However, since the loopback mix can't take other than the front DAC,
there is no longer individual volume controls for headphones and
speakers. Once when the loopback control is off, these volumes take
effect.
Since the individual volumes are more desired in general use caess, the
loopback mode is set to off as default for now.
Clemens Ladisch [Sun, 17 Jul 2011 20:18:05 +0000 (22:18 +0200)]
ALSA: virtuoso: fix silent analog output on Xonar Essence ST Deluxe
Commit dd203fa97bd5 (ALSA: virtuoso: remove non-working controls on
Essence ST Deluxe) made it impossible to adjust the volume after the
driver initialized it to muted.
Ensure that those DACs that can be accessed with I2C are initialized
to the same volume that is the reset default of the DAC without I2C.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: 2.6.38+ <stable@kernel.org>
Mark Brown [Thu, 14 Jul 2011 09:21:37 +0000 (18:21 +0900)]
ASoC: Correct WM8994 MICBIAS supply widget hookup
The WM8994 and WM8958 series of devices have two MICBIAS supplies rather
than one, the current widget actually manages the microphone detection
control register bit (which is managed separately by the relevant API).
Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2
widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Mark Brown [Thu, 14 Jul 2011 08:11:38 +0000 (17:11 +0900)]
ASoC: Reduce power consumption for idle DAIs in WM8994
If DAIs are idle but their clocks are in use for some reason (eg, as
SYSCLK or for accessory detect) then set the clock dividers to the maximum
to reduce slightly the power consumption of the unclocked circuits.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 15 Jul 2011 18:12:18 +0000 (03:12 +0900)]
ASoC: Handle failed WM8994 FLL lock waits
Try the completion before we start the FLL so that if an interrupt was
delayed long enough for us to miss it we don't wait for the completion
it signalled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8983 is a low power, high quality stereo CODEC
designed for portable multimedia applications. Highly flexible
analogue mixing functions enable new application features,
combining hi-fi quality audio with voice communication.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel T Chen [Fri, 15 Jul 2011 02:06:06 +0000 (22:06 -0400)]
ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
BugLink: https://bugs.launchpad.net/bugs/774895
The original reporter states that his volume keys do not change the
desired Master and PCM mixer elements together, so apply the hp+mute led
quirk for his PCI SSID.
Reported-by: Jeffrey Finkelstein Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: SAMSUNG: 24-bit audio playback on Exynos4210
Using 256fs or 512fs will result in distortion of 24-bit
audio samples. This is because the lrclk generated is not
proper. Using 384 fs generates proper output.
Signed-off-by: Giridhar Maruthy <giridhar.maruthy@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 12 Jul 2011 06:25:03 +0000 (15:25 +0900)]
ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
The individual devices should set the flag dcs_done_irq in the hubs
shared data structure to indicate that they will flag the interrupt
by calling wm_hubs_dcs_done().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched. It is possible
there are differences between STA326 and STA328 or future
chip revisions in these bits, and clobbering them might
cause malfunction.
Signed-off-by: Johannes Stezenbach <js@sig21.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: STA32x: Add mixer controls for biquad coefficients
The STA32x has a number of preset EQ settings, but also
allows full user control of the biquad filter coeffcients
(when "Automode EQ" is set to "User").
Each biquad has five signed, 24bit, fixed-point coefficients
representing the range -1...1. The five biquad coefficients
can be uploaded in one atomic operation into on-chip
coefficient RAM.
There are also a few prescale, postscale and mixing
coefficients, in the same numeric format and range
(a negative coefficient inverts phase).
These coefficients are made available as SNDRV_CTL_ELEM_TYPE_BYTES
mixer controls.
Signed-off-by: Johannes Stezenbach <js@sig21.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: usb-audio - Add quirks for M-Audio Fast Track Pro and Quattro
This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and
endpoints to boot and setup those devices with special options (digital
inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are
just adapted to match the new global M-Audio parameters.
Special configurations can be then loaded through a modprobe conf file.
For example, to set the 24 bits mode on the Fast Track Pro add
/etc/modprobe.d/fast_track_pro.conf :
ALSA: hda - Expose secret DAC-AA connection of some VIA codecs
VT1718S and co have a secret connection from DAC to AA-mix, which
doesn't appear in the connection list obtained from the h/w.
Currently the driver fixes the connection index locally at init, but
now we can expose it statically via snd_hda_override_connections()
so that this conection can be checked better by the parser in future.
ALSA: hda - Turn on extra EAPDs on Conexant codecs
Some machines seem to use EAPD control of the unused pin for controlling
the overall EAPD. Since the driver currently doesn't check the EAPD of
unused pins, the EAPD isn't enabled. For avoiding such a problem, turn
all extra EAPDs on as default.
ALSA: hda - Preserve input pin-ctl bits in HP-automute for VIA codec
For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.
ALSA: hda - Set line-out pin-ctls properly when indep-HP mode changes
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.
ALSA: hda - Via Fix speaker-mute checks in VIA driver
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack. Currently
it checks only the line-out state and ignores the headphone.
projects / mirror_ubuntu-zesty-kernel.git / log