18 Aug, 2010
1 commit
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With some hardware combinations, the PCM interrupts are acknowledged
before the period boundary from the emu10k1 chip. The midlevel PCM code
gets confused and the playback stream is interrupted.It seems that the interrupt processing shift by 2 samples is enough
to fix this issue. This default value does not harm other,
non-affected hardware.More information: Kernel bugzilla bug#16300
[A copmile warning fixed by tiwai]
Signed-off-by: Jaroslav Kysela
Cc:
Signed-off-by: Takashi Iwai
15 Aug, 2010
1 commit
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unifdef-y and header-y has same semantic.
So there is no need to have both.Drop the unifdef-y variant and sort all lines again
Signed-off-by: Sam Ravnborg
08 Aug, 2010
1 commit
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* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
05 Aug, 2010
1 commit
02 Aug, 2010
1 commit
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…soc-2.6 into topic/asoc
30 Jul, 2010
1 commit
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Signed-off-by: Kuninori Morimoto
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown
29 Jul, 2010
2 commits
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Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.Signed-off-by: Peter Ujfalusi
Acked-by: Mark Brown
Signed-off-by: Liam Girdwood -
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.Signed-off-by: Peter Ujfalusi
Acked-by: Mark Brown
Signed-off-by: Liam Girdwood
19 Jul, 2010
1 commit
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All current users of pm_qos_add_request() have the ability to supply
the memory required by the pm_qos routines, so make them do this and
eliminate the kmalloc() with pm_qos_add_request(). This has the
double benefit of making the call never fail and allowing it to be
called from atomic context.Signed-off-by: James Bottomley
Signed-off-by: mark gross
Signed-off-by: Rafael J. Wysocki
18 Jul, 2010
1 commit
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Specified ID is necessary, when some codecs are used with FSI.
Signed-off-by: Kuninori Morimoto
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown
13 Jul, 2010
2 commits
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This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.Signed-off-by: Kuninori Morimoto
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown -
There is no necessity that each bit in this area has the meaning.
This patch modify it to sequence numberSigned-off-by: Kuninori Morimoto
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown
05 Jul, 2010
1 commit
28 Jun, 2010
1 commit
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When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.Signed-off-by: Dave Dillow
Signed-off-by: Jaroslav Kysela
25 Jun, 2010
1 commit
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This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.Signed-off-by: Vladimir Zapolskiy
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown
31 May, 2010
3 commits
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Some systems codecs need to configure some registers before and after
powering down some of their part. As a convenience add a macro for that.Signed-off-by: Arnaud Patard
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown -
This defines the 24bps and 40bps (8khz sample rate) G.723 codec
formats. They are going to be used once I submit the driver for
an mpeg4/g723 compression card.I've updated the signed value to -1 as per Takashi's comments
since these are non-linear formats.Signed-off-by: Ben Collins
Signed-off-by: Takashi Iwai
21 May, 2010
1 commit
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* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
ALSA: hda: Storage class should be before const qualifier
ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
ASoC: TWL6040: Enable earphone path in codec
ASoC: SDP4430: Add support for Earphone speaker
ASoC: SDP4430: Add sdp4430 machine driver
ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
ALSA: sound/pci/asihpi: Use kzalloc
ALSA: hdmi - dont fail on extra nodes
ALSA: intelhdmi - add id for the CougarPoint chipset
ALSA: intelhdmi - user friendly codec name
ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
ALSA: asihpi: incorrect range check
ALSA: asihpi: testing the wrong variable
ALSA: es1688: add pedantic range checks
ARM: McBSP: Add support for omap4 in McBSP driver
ARM: McBSP: Fix request for irq in OMAP4
OMAP: McBSP: Add 32-bit mode support
...
20 May, 2010
3 commits
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Conflicts:
sound/soc/codecs/ad1938.c
17 May, 2010
1 commit
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This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0Signed-off-by: Arnaud Patard
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown
12 May, 2010
1 commit
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This fixes some whitespace/indentation flaws I stumbled over.
Signed-off-by: Daniel Mack
Signed-off-by: Takashi Iwai
11 May, 2010
2 commits
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If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.Signed-off-by: Peter Ujfalusi
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown -
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation. I did this because request more
accurately represents what it actually does.Also, I added a string based ABI for users wanting to use a string
interface. So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface. (someone asked me for it and I don't think
it hurts anything.)This patch updates some documentation input I got from Randy.
Signed-off-by: markgross
Signed-off-by: Rafael J. Wysocki
10 May, 2010
5 commits
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As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked. This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link. It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.Tested-by: Peter Ujfalusi
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown -
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.Tested-by: Peter Ujfalusi
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown -
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.Tested-by: Peter Ujfalusi
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown -
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.Also, a new PnP id is added for the card I acquired (the change
was tested on this card).Signed-off-by: Krzysztof Helt
Signed-off-by: Takashi Iwai -
Allocate the snd_es1688 during the snd_card allocation.
This allows to remove the card pointer from the snd_es1688 structure.Signed-off-by: Krzysztof Helt
Signed-off-by: Takashi Iwai
07 May, 2010
2 commits
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This reverts commit 6f3991152f20933b77eff30413e893bf1a15e578.
Since core has now support for limiting the volume on controls this
patch is not needed. Furthermore, this patch actually prevents the core
to set new volume on the TPA.Signed-off-by: Peter Ujfalusi
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown -
Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);
This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.Signed-off-by: Peter Ujfalusi
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown
06 May, 2010
3 commits
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…soc-2.6 into topic/asoc
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Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.Signed-off-by: Peter Ujfalusi
Acked-by: Mark Brown
Signed-off-by: Liam Girdwood -
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.Signed-off-by: Jarkko Nikula
Acked-by: Peter Ujfalusi
Acked-by: Mark Brown
Signed-off-by: Liam Girdwood
30 Apr, 2010
1 commit
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The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.Signed-off-by: Mark Brown
Acked-by: Liam Girdwood
26 Apr, 2010
1 commit
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This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.Signed-off-by: Vladimir Zapolskiy
Acked-by: Liam Girdwood
Signed-off-by: Mark Brown
17 Apr, 2010
1 commit
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One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.Signed-off-by: Mark Brown
Acked-by: Liam Girdwood
16 Apr, 2010
1 commit
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Signed-off-by: Jaroslav Kysela