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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
