The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for ASoC DSP support.
Add a DAPM API call to determine whether a DAPM audio path is valid between
source and sink widgets. This also takes into account all kcontrol mux and mixer
settings in between the source and sink widgets to validate the audio path.
This will be used by the DSP core to determine the runtime DAI mappings
between FE and BE DAIs in order to run PCM operations.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Connect the WM1250-EV1 baseband simulator on Littlemill systems up to
the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider
range of use cases to be represented.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch converts multiple if conditions in to single if with "&&"s.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.
This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,
(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled
This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.
Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ASoC: Merge tag 'v3.4-rc3' into for-3.5
Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.
Conflicts:
sound/soc/soc-core.c
sound/soc/tegra/tegra_i2s.c
sound/soc/tegra/tegra_spdif.c
Fix the following build warning:
sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event':
sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64'
'%llx' should be used with 'u64' type.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct. If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.
This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.
This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there. It is expected that the bias
level callbacks will be used for clock configuration.
Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute(). This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here. We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.
At present we are also restricted to a single DAPM link for the entire
DAI. Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We should always have a CODEC already there when registering a CODEC DAI
and for CODEC<->CODEC links a dai_link will have two CODECs so it's much
simpler to do things at registration time.
This results in a slight change in the error handling for failed CODEC
DAI registrations but practically speaking these are never supposed to
fail so there shouldn't be much issue. The change is that we don't fail
the overall CODEC registration if the DAI registration fails; this seems
more robust anyway as we may not need to use a given DAI in a particular
system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When two CODEC DAIs are linked directly to each other then if we give the
same master mode settings to both devices things won't work as either
neither will drive or they'll drive against each other. Flip the settings
for the DAI in the CPU slot of the DAI link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to allow CODEC<->CODEC links to function we will need to allow
DAPM paths to be created that pass through DAIs rather than only ones
that are source or sunk at the DAI.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This helps us ignore errors in callers if the operation failed due to not
being available as opposed to an error.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
A few regression fixes for Realtek HD-audio codecs, mainly specific to
some laptop models.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull another round of sound fixes from Takashi Iwai:
"A few regression fixes for Realtek HD-audio codecs, mainly specific to
some laptop models."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).
ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines
ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G
ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co
ALSA: hda/realtek - Add a few ALC882 model strings back
We added locking here but there were a couple error paths where we
forgot to drop the lock before returning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All Tegra ASoC drivers will be reworked to use MMIO regmaps. Select
this in Kconfig.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current SuperH FSI require simple-card driver as sound card.
This patch select it on Kconfig when FSI was selected.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch uses simple-card driver instead of fsi-da7210 on each board.
To select DA7210 driver, each boards select it on Kconfig.
This patch removes fsi-da7210 driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch uses simple-card driver instead of fsi-hdmi on each board.
This patch removes fsi-hdmi driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch uses simple-card driver instead of fsi-ak4642 on each board.
To select AK4642 driver, each boards select it on Kconfig.
This patch removes fsi-ak4642 driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC requires card.c file to each platforms in order to
specifies its CPU and Codecs pair.
But the differences between these were only value/strings of setting.
In order to reduce duplicate driver, this patch adds generic/simple-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds Kconfig options for the Tegra30 AHUB and I2S controller, and
updates the Tegra+WM8903 machine driver Kconfig to select those.
Includes a squashed bugfix from Sumit Bhattacharya <sumitb@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This provides an ASoC DAI interface for Tegra 30's I2S controller.
Includes a squashed bugfix from Sumit Bhattacharya <sumitb@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The AHUB (Audio Hub) is a mux/crossbar which links all audio-related
devices except the HDA controller on Tegra30. The devices include the
DMA FIFOs, DAM (Digital Audio Mixers), I2S controllers, and SPDIF
controller. Audio data may be routed between these devices in various
combinations as required by board design/application.
Includes a squashed bugfix from Nikesh Oswal <noswal@nvidia.com>
Includes squashed bugfixes from Sumit Bhattacharya <sumitb@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the
'for (;;)' loop, if the 'badness' value returned from
fill_and_eval_dacs() is negative, then we'll return from the function
without freeing the memory we allocated for 'best_cfg', thus leaking.
Fix the leak by kfree()'ing the memory when badness is negative.
While I was there I also noticed some trailing whitespace in the
function that I removed (along with all other trailing whitespace in
the file) - it didn't seem worth-while to do that as two patches, so I
hope it's OK that I just did it all as one patch.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A user reported that setting model=imac24 used to allow sound to work on their
Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models
to auto-parser" removed this model option. All Mac machines are now explicitly
handled with a quirk and the auto-parser. This adds a quirk for the device
found on the Mac Pro 5,1 machines.
This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559
[sorted the new entry in the ID number order by tiwai]
Reported-by: Gabriel Somlo <somlo@cmu.edu>
Signed-off-by: Josh Boyer <jwboyer@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's compatible with 8930G.
Using the same fixup gives the proper 5.1 sound back.
Reported-and-tested-by: Dany Martineau <dany.luc.martineau@gmail.com>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add GPIO1 setup explicitly for Acer Aspire 493x & co.
This could be set by alc_auto_init_amp(), but it's safer to set it
more explicitly in the fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- A series of fixes for Conexant 20549 HD-audio codec chip
- A workaround for HDMI hotplug debug prints that annoyed people
- A fix for the new support of platform DAPM contexts
- Many driver-specific minor fixes
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
- A series of fixes for Conexant 20549 HD-audio codec chip
- A workaround for HDMI hotplug debug prints that annoyed people
- A fix for the new support of platform DAPM contexts
- Many driver-specific minor fixes
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - hide HDMI/ELD printks unless snd.debug=2
ALSA: sound/isa/sscape.c: add missing resource-release code
sound: sound/oss/msnd_pinnacle.c: add vfrees
ALSA: hda - clean up CX20549 test mixer setup
ALSA: hda - CX20549 doesn't need pin_amp_workaround.
ALSA: hda - Remove CD control from model=benq for CX20549
ALSA: hda - fix record volume controls of CX20459 ("Venice")
ALSA: hda - Rename capture sources of CX20549 to match common conventions
ALSA: hda - Fix proc output for ADC amp values of CX20549
ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
ASoC: set idle_bias_off=1 for all platform DAPM contexts
ASoC: imx-audmux: Check for NULL pointer
ASoC: imx-audmux: Fix ssi port numbers in sysfs
ASoC: ak4642: fixup: mute needs +1 step
MAINTAINERS: Don't list everyone working on Wolfson drivers
MAINTAINERS: Add missing ASoC OMAP co-maintainer
ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro
ASoC: tegra: ensure clocks are enabled when touching registers
ASoC: sgtl5000: Enable VAG when DAC/ADC up
ALSA: asihpi - fix return value of hpios_locked_mem_alloc()
Pull media fixes from Mauro Carvalho Chehab:
- dvb core: there is a regression found when used with xine. For
whatever unknown reason, xine (and xine-lib clients) wants that the
frontend to tell what frequency he is using even before the PLL lock
(or at least, it expects a non-zero frequency).
On DVB, the frequency is only actually known after a frequency
zig-zag seek, done by the DVB core. Anyway, the fix was trivial.
That solves Fedora BZ#808871.
- ivtv: fix a regression when selecting the language channel
- uvc: fix a race-related crash
- it913x: fixes firmware loading
- two trivial patches (a dependency issue at a radio driver at sound
Kconfig, and a warning fix on dvb).
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
[media] uvcvideo: Fix race-related crash in uvc_video_clock_update()
[media] Drivers/media/radio: Fix build error
[media] dvb_frontend: fix compiler warning
[media] it913x: fix firmware loading errors
[media] ivtv: Fix AUDIO_(BILINGUAL_)CHANNEL_SELECT regression
[media] dvb_frontend: regression fix: userspace ABI broken for xine
Since there are still many Acer models that might not be covered by
the current fixup table, let's add back a few typical model names so
that user can test the fixup without recompiling.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/soc/codecs/wm9705.c: In function 'ac97_prepare':
sound/soc/codecs/wm9705.c:251: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/ac97.c: In function 'ac97_prepare':
sound/soc/codecs/ac97.c:33: error: 'runtime' undeclared (first use in this function)
This was caused by commit e6968a (ASoC: codecs: Remove rtd->codec usage from CODEC drivers),
which removed the 'struct snd_pcm_runtime *runtime = substream->runtime' definition.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following build error:
sound/soc/codecs/wm9712.c:482:32: error: 'runtime' undeclared (first use in this function)
sound/soc/codecs/wm9712.c:499:33: error: 'runtime' undeclared (first use in this function)
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Recent list discussions concluded that drivers should not be calling
of_have_populated_dt(), and hence of_have_populated_dt() should not be
exported. Use a different mechanism to detect DT vs. non-DT boot.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also remove two warnings when CONFIG_SND_DEBUG is not set:
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Later WM8994 class devices can bypass the FLL from BCLK. Do this
automatically when the FLL input and output frequencies match up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
release_firmware() does its own NULL ptr testing, it's redundant to
also test before calling it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
At the point of this error-handling code, both regions and the dma have
been allocated, so free it as done in previous and subsequent
error-handling code.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the point of this error-handling code, HAVE_DSPCODEH may be undefined,
so free INITCODE and PERMCODE as done elsewhere. A jump and label are
introduced to avoid code duplication.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Sunday, April 01, 2012 21:09:34 Tracey Dent wrote:
> radio-maxiradio depends on SND_FM801_TEA575X_BOOL to build or will
> result in an build error such as:
>
> Kernel: arch/x86/boot/bzImage is ready (#1)
> ERROR: "snd_tea575x_init" [drivers/media/radio/radio-maxiradio.ko] undefined!
> ERROR: "snd_tea575x_exit" [drivers/media/radio/radio-maxiradio.ko] undefined!
> WARNING: modpost: Found 6 section mismatch(es).
> To see full details build your kernel with:
> 'make CONFIG_DEBUG_SECTION_MISMATCH=y'
> make[1]: *** [__modpost] Error 1
> make: *** [modules] Error 2
>
> Select CONFIG_SND_TEA575X to fixes problem and enable
> the driver to be built as desired.
>
> v2:
> instead of selecting CONFIG_SND_FM801_TEA575X_BOOL, select
> CONFIG_SND_TEA575X, which in turns selects CONFIG_SND_FM801_TEA575X_BOOL
> and any other dependencies for it to build.
No, this is the correct patch:
RADIO_MAXIRADIO should be treated just like RADIO_SF16FMR2, I just didn't
realize at the time that it had to be added as a SND_TEA575X dependency.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Tested-by: Shea Levy <shea@shealevy.com>
Acked-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
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Merge tag 'v3.4-rc2' into for-3.5
Linux 3.4-rc2 contains some bug fixes we need, including the addition of
an export for regcache_sync_region().
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: fixes for 3.4
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
