This typo caused headphone pins not to be initialized correctly.
BugLink: https://bugs.launchpad.net/bugs/871582
Reported-by: Effenberg
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The purpose of this patch is to remove a section of "bad" code that
assigns the last DAC to ports E or F in order to support notebooks
with docking in earlier days, around ALSA 1.0.19 - 21. This is not
necessary now and actually breaks some configurations that use these
ports as other devices. This have been tested on several different
configurations to make sure that it is working for different combinations.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit ef18beded8 introduced a
mechanism to assign the previously used slot for the next reopen of a
PCM stream. But the PCM device number isn't always unique (it may
have multiple substreams), and also the code doesn't check the stream
direction, thus both playback and capture streams share the same
device number.
For avoiding this conflict, make a unique key for each substream and
store/check this value at reopening.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the speaker outputs are more than the headphone outputs, it implies
that the system has surround speakers while the headphones are only for
monitoring the front. In such a case, it's better to put speakers as
the primary outputs so that the driver can build up and keep the
surround setup. Otherwise the system will pick up the headphone as
primary, and offers less channels than the speakers do support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since hda_proc.c is now the only user of snd_print_pcm_rates(), better to
put it back locally to hda_proc.c and revert to the old style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SAD sampling rate information reported in
/proc/asound/cardX/eldX is incorrect due to a mismatch
between HDA and HDMI frequencies. Add new routine to provide
relevant values.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar to Line Out, these constants form the base for future
patches enabling input jack reporting for Line in jacks.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we run out of DACs when trying to assign a DAC to a secondary
headphone, prefer the DAC of the first headphone to the primary
(usually line out) DAC.
BugLink: http://bugs.launchpad.net/bugs/845275
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Sigmatel/IDT parser should have the same naming convention
for input jacks as the other codecs have.
BugLink: http://bugs.launchpad.net/bugs/859704
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Applications may want to read ELD information to
understand what codecs are supported on the HDMI
receiver and handle the a-v delay for better lip-sync.
ELD information is exposed in a device-specific
IFACE_PCM kcontrol. Tested both with amixer and
PulseAudio; with a corresponding patch passthrough modes
are enabled automagically.
ELD control size is set to zero in case of errors or
wrong configurations. No notifications are implemented
for now, it is expected that jack detection is used to
reconfigure the audio outputs.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit a810364a04
ALSA: hda - Handle -1 as invalid position, too
caused a regression on some machines that require the position-buffer
instead of LPIB, e.g. resulting in noises with mic recording with
PulseAudio.
This patch fixes the detection by delaying the test at the timing as
same as 3.0, i.e. doing the position check only when requested in
azx_position_ok().
Reported-and-tested-by: Rocko Requin <rockorequin@hotmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the ugly real world, there area really broken devices that don't set
codec SSID correctly. In such a case, the ID can be random, thus the
patching won't work reliably.
For applying the patch forcibly to such a device, the driver will skip
the vendor and/or subsystem ID checks when zero or a negative number is
given in [codec] section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new option "snoop" for the traffic control of the HD-audio
controller chip. When set to 0, the non-snooping mode is used with
the traffic control bit is set in each stream control register.
This may allow better operations in the low power mode, but the actual
implementation is depending pretty much on the chipset.
As already implemented, more or less each chipset has own snoop-control
register bit. Now this setup refers to the snoop option, too.
Also, a new VIA chipset may require the non-snooping mode when set so
in BIOS. In such a case, the option value is overridden.
As default, it's still set to snoop=1 for keeping the same behavior as
before. In near future, it'll be set to 0 as default after checking
it works in every system well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since really_cleanup_stream() is called from both purity_inactive_streams()
and hda_cleanup_all_streams(), the verbs to clear the PCM channel and
format may be called multiple times unnecessarily.
This patch adds checks to skip these unneeded verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The spec->autocfg.line_out_pins[] may contain the same pins as hp_pins[]
depending on the configuration. When they are identical, detecting the
line_jack_present flag screws up the auto-mute because alc_line_automute()
is called unconditionally at initialization while it won't be triggered
by unsol events, thus the old line_jack_present flag is kept for the
whole run.
For fixing this buggy behavior, the driver needs to check whether the
line-outs are really individual, and skip if same as headphone jacks.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone pin is assigned as primary output to line_out_pins[],
the automatic HP-pin assignment by ASSID must be suppressed. Otherwise
a wrong pin might be assigned to the headphone and breaks the auto-mute.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
This patch is necessary to make internal speakers work on this chip.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Tested-by: Alex Wolfson <alex.wolfson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add "AD198x Headphone" playback device for independent headphone playback
while playing 7.1 surround using rear panel audio jacks.
- Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using
DAC0 instead of DAC1 (HDA_FRONT) was already added to all models.
- Add "Independent HP" switch to enable/disable this playback device.
When the switch is OFF, headphone use "copy front" mode to get the front
channel as the green jack.
When the switch is ON, you can play stereo sound through "AD198x Headphone"
device to headphone while playing 7.1 surround sound through "AD198x Analog"
device.
The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog"
is open.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smatch has a new check for Rosenberg type information leaks where
structs are copied to the user with uninitialized stack data in them.
The status struct has a hole in it, and on some paths not all the
members were initialized.
struct hdspm_status {
unsigned char card_type; /* 0 1 */
/* XXX 3 bytes hole, try to pack */
enum hdspm_syncsource autosync_source; /* 4 4 */
long long unsigned int card_clock; /* 8 8 */
The hdspm_version struct had holes in it as well.
struct hdspm_version {
unsigned char card_type; /* 0 1 */
char cardname[20]; /* 1 20 */
/* XXX 3 bytes hole, try to pack */
unsigned int serial; /* 24 4 */
short unsigned int firmware_rev; /* 28 2 */
/* XXX 2 bytes hole, try to pack */
int addons; /* 32 4 */
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") seems to
break systems that were previously working without a tuner.
As a bonus, this should fix init and cleanup for the case where the
tuner is explicitly disabled.
Reported-and-tested-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") added
incorrect error handling.
Once we have successfully called snd_device_new(), the cleanup
function fm801_free() will automatically be called by snd_card_free()
and we must *not* also call fm801_free() directly.
Reported-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled],
We run all interrupt handlers with interrupts disabled
and we even check and yell when an interrupt handler
returns with interrupts enabled (see commit [b738a50a:
genirq: Warn when handler enables interrupts]).
So now this flag is a NOOP and can be removed.
Signed-off-by: Yong Zhang <yong.zhang0@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since modern HDMI cards often have more than one output pin and thus
input device, we need to know which one has actually been plugged in.
This patch adds a name hint that indicates which PCM device is connected
to which pin.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Increase readability and understandability in the automute code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rules to allow disabling the PCM playback and capture SRCs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The installation of the minimum period size constraint in the PCM open
callbacks was not checked for errors. Add this check, and move the call
to the beginning of the function to avoid having to do any cleanups in
the error case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the system has only the headphone and the line-out jacks without
speakers, the current auto-mute code doesn't work. It's because the
spec->automute_lines flag is wrongly referred in update_speakers().
This flag must be meaningless when spec->automute_hp_lo isn't set, thus
they should be always coupled.
The patch fixes the problem and add a comment to indicate the
relationship briefly.
BugLink: http://bugs.launchpad.net/bugs/851697
Reported-by: David Henningsson <david.henningsson@canonical.com>
Tested-By: Jayne Han <jayne.han@canonical.com>
Cc: stable@kernel.org (3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive: To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero. At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller. This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.
With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value,
which forces us to handle the parameters differently.
This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the
device interrupt is handled by the caller, and makes the allocation of
the interrupt to depend only on the irq parameter. As suggested by
Takashi, the irq_flags parameter was dropped because, when used, it had
the constant value IRQF_DISABLED.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recursive search of widget connections in snd_hda_get_conn_index()
must be terminated at the pin and the audio-out widgets. Otherwise
you'll get "too deep connection" warnings unnecessarily.
Reported-by: Francis Moreau <francis.moro@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- use DAC0 instead of DAC1 for Port-A Headphone
- assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp"
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headphones has stopped working for the original reported (a regression
compared to 2.6.38). This is because Speaker and Headphones share the
same DAC, in which case no Headphones volume control was created.
This patch fixes so that both Speaker and Headphones volume
controls are created in such scenario.
BugLink: http://bugs.launchpad.net/bugs/817943
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the multi-io jacks are available, parse them first and assign DACs
before parsing speakers and headphones. This allows a better chance of
surround I/O in some desktops and laptops with limited DACs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 23c09b0090
ALSA: hda - Support multiple speakers by Realtek auto-parser
changes the return value from alc_get_line_out_pfx(), and it breaks
the center/LFE mixer split check. The caller must test with a string
"CLFE" now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the pincfg table to patch_conexant.c for fixing up the extra
pin-configuration for auto-parser. As an example, Lenovo X200 model is
replaced with this new mechanism. (This also fixes the wrong mixer
elements for docking-station I/O in the previous model quirk
automagically.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are references in the code to 256 sources, so I tested it with 256 aplays,
of which the first and last with real data and the rest playing /dev/zero .
Also increase amount of page tables, so the default aplay size works.
Signed-off-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like ALC662 asus-mode* models, rewrite the laptop-amic and
dmic models with the static pin-config tables.
Now we can get rid of all alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-implement the asus-mode[1-8] quirks with the pin-config tables.
They are provided in case where BIOS is broken on the device, so it's
not enabled in PCI SSID lookup table. User needs to specify it via model
option explicitly if the driver doesn't work with the BIOS setup as is.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For supporting both the multiple headphones and the multiple speakers,
add the new field in struct hda_multi_out, and evaluate in the standard
setup functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's remove the rest of ALC861 and ALC861-VD quirks.
If any breakage is found, it can be fixed easily via the pin-config
table update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the cleanup by commit 6727b12669,
the specific setups for dallas and hp models, using VREF50 for mic pins,
were lost. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... and add a new bit-flags argument to specify the behavior of the
function. The older function is kept as is (as a wrapper).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple headphones or speakers are assigned but no individual
DACs are available, the driver should take the first HP/SPK DAC instead
of another primary output. The patch adds a bit-flag to dac field of
struct pin_dac_pair indicating that it's a slave DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal states, jack_present and line_jack_present should be
updated upon unsolicited events even if no automute is set.
Otherwise the wrong state is referred when the automute behavior is
changed by the mixer control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone or speaker output has no own DAC, initialize the path
using the primary DAC. Otherwise the path won't be set properly and
can result in the silence.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_get_conn_index() returns a negative value while the current code
stores it in an unsigned int. It must be stored in a signed integer.
Reported-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently HD-audio driver shows the all error ELD byte as an error
in the kernel message. This is annoying when the video driver doesn't
set the correct ELD from the beginning. e.g. radeon sends a zero-byte
data, but we still check ELD with the fixed 128 byte as a workaround
for some broken devices, it spews 128-times errors.
For avoiding this, the driver aborts reading when the first byte is
invalid. In such a case, the whole data is certainly invalid.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_hda_parse_pin_def_config(), we checked the associated number
of speaker pins and accepts only one number exclusively. But many BIOS
seem to give different assoc number for surround speakers, thus we'd
better to accept all speaker pins no matter which assoc number, and sort
like done for the headphone pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of multiple speakers by Realtek auto-parser.
When all speaker pins have individual DACs, create each speaker volume
control. Otherwise, create a bind-volume control for all speaker outs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new parser may use "PCM" volume, but it was missing the vmaster
slave list, thus "Master" volume didn't control it.
Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement new fixup entries for Quanta FL1 and Fujitsu Lifebook
specific COEF and pin configurations. Removed the model entries
from alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the duplicated creation of capture-mixer elements for some static
ALC268 configurations. The capture mixers must be put to cap_mixer field
instead of mixers array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refactoring the code using snd_pcm_hw_constraint_pow2() helper function.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AES32 supports the non-standard 128kHZ, and this is enabled only when
SNDRV_PCM_RATE_KNOT is set in hw.rates field.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar Essence ST/STX, the connector J14 has been confirmed to be
a digital input, so enable it in the driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC260 HP models work with the BIOS auto-parser. Let's cut them off.
Also move alc260_hp_master_*() to alc262_quirks.c as these are still
referred from there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/826081
The original reporter needs 'Headphone Jack Sense' enabled to have
audible audio, so add his PCI SSID to the whitelist.
Reported-and-tested-by: Muhammad Khurram Khan
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Newer RME cards like RayDAT and AIO support 32 samples per period. This
value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control
register.
Since {1,1,1} is also the representation for 8192 samples/period on
older RME cards, we have to special case 32 samples and 32768 bytes
according to the actual card.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, hdspm_decode_latency is called several times, violating the
DRY principle. Given that we need to distinguish between old and new
cards when decoding the latency bits in the control register, introduce
hdspm_get_latency() to provide the required functionality.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On newer RME cards like RayDAT and AIO, the 8192 samples per period size
are no longer supported. Instead, setting all three bits of
HDSP_LatencyMask to one ({1,1,1}) now corresponds to 32 samples per
period.
To make this more obvious to future developers, let's reorder the array
according to their bit representation, starting at 64 ({0,0,0}) up to
4096 ({1,1,0}) and finally 32 ({1,1,1}).
Note that this patch doesn't change semantics.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On newer RME cards like RayDAT and AIO, the lower bound is 32 samples
per period in contrast to 64 samples as seen on older cards.
We hence lower period_bytes_min to 32 * 4. Four bytes per sample.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kailang Yang