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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Most of changes are small and easy cleanup or fixes:
- a few HD-audio Realtek codec fixes and quirks
- Intel HDMI audio fixes for Broadwell and Haswell / ValleyView
- FireWire sound stack cleanups
- a couple of sequencer core fixes
- compress ABI fix for 64bit
- conversion to modern ktime*() API"
* tag 'sound-fix-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (23 commits)
ALSA: hda/realtek - Add more entry for enable HP mute led
ALSA: hda - Add quirk for external mic on Lifebook U904
ALSA: hda - fix a fixup value for codec alc293 in the pin_quirk table
ALSA: intel8x0: Use ktime and ktime_get()
ALSA: core: Use ktime_get_ts()
ALSA: hda - verify pin:converter connection on unsol event for HSW and VLV
ALSA: compress: Cancel the optimization of compiler and fix the size of struct for all platform.
ALSA: hda - Add quirk for ABit AA8XE
Revert "ALSA: hda - mask buggy stream DMA0 for Broadwell display controller"
ALSA: hda - using POS_FIX_LPIB on Broadwell HDMI Audio
ALSA: hda/realtek - Add support of ALC667 codec
ALSA: hda/realtek - Add more codec rename
ALSA: hda/realtek - New vendor ID for ALC233
ALSA: hda - add two new pin tables
ALSA: hda/realtek - Add support of ALC891 codec
ALSA: seq: Continue broadcasting events to ports if one of them fails
ALSA: bebob: Remove unused function prototype
ALSA: fireworks: Remove meaningless mutex_destroy()
ALSA: fireworks: Remove a constant over width to which it's applied
ALSA: fireworks: Improve comments about Fireworks transaction
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More HP machine need mute led support.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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According to the bug reporter (Данило Шеган), the external mic
starts to work and has proper jack detection if only pin 0x19
is marked properly as an external headset mic.
AlsaInfo at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1328587/+attachment/4128991/+files/AlsaInfo.txt
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1328587
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The fixup value for codec alc293 was set to
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE by a mistake, if we don't fix it,
the Dock mic will be overwriten by the headset mic, this will make
the Dock mic can't work.
Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts() and returns
the monotonic time in a timespec.
Use ktime based ktime_get() and use the ktime_delta_us() function to
calculate the delta instead of open coding the timespec math.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts().
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch will verify the pin's coverter selection for an active stream
when an unsol event reports this pin becomes available again after a display
mode change or hot-plug event.
For Haswell+ and Valleyview: display mode change or hot-plug can change the
transcoder:port connection and make all the involved audio pins share the 1st
converter. So the stream using 1st convertor will flow to multiple pins
but active streams using other converters will fail. This workaround
is to assure the pin selects the right conveter and an assigned converter is
not shared by other unused pins.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit 432481220 (ASoC: fsl-ssi: Use regmap) removed struct ccsr_ssi.
Unfortunately, the structure is still used. This causes
mpc85xx_smp_defconfig and mpc85xx_defconfig builds to fail with
sound/soc/fsl/fsl_dma.c:926:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
ound/soc/fsl/fsl_dma.c:927:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
Fix by using constants, similar to original commit.
Cc: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Guenter Roeck <linux@roeck-us.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Index of dma name should use -1, not +1 when capture case.
Thank you Dan.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Bios does not set up the pin config default correctly (everything
is set to zero). Reporter claims that 6stack-dig and 6stack-automute
solve the problem.
Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05
BugLink: https://bugs.launchpad.net/bugs/1319291
Reported-by: Stefano Statuti <stefano.statuti@hotmail.it>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The calculation code does
u64 = (u32 - u32) * 100000;
The 64 bits are of no help here as the type is casted only after the
multiplication, and therefore the result may overflow, possibly causing
inoptimal or wrong clock setup in an unfortunate case (the maximum
result value of the first substraction is currently 47999).
Fix the code to cast before multiplication.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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We should not copy the return value into this val since it's supposed to
get the value of the register not the success result of regmap_read().
Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Commit c9e065c27fe9 ("ASoC: dapm: Make sure to always update the DAPM graph
in _put_volsw()") stopped updating register values in those cases where
initial after boot state of kcontrol appears to not change but where
register value still needs update because it is not in sync with the
kcontrol state.
Fix this by doing snd_soc_test_bits() unconditionally as it was before but
by using separate flags for kcontrol and register state changes. This allow
both DAPM graph to be updated when disabling auto-muted control and update
register if it is out-of-sync in respect of kcontrol state.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This reverts commit 7189eb9b8f7962474956196c301676470542f253.
It will use LPIB to get the DMA position on Broadwell HDMI Audio.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Broadwell HDMI can't use position buffer reliably, force to use LPIB
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the SigmaDSP module is built-in, but the I2C core is build as a module
we'll get a undefined reference:
sound/built-in.o: In function `sigma_action_write_i2c':
:(.text+0x5d8d4): undefined reference to `i2c_master_send'
This can happen if a audio driver that is using the regmap SigmaDSP interface is
built into the kernel, but core I2C support is build as a module. To fix this
split the SigmaDSP module into three modules, one module providing the core
infrastructure and two small modules implementing the regmap and I2C interfaces.
This allows e.g. the core infrastructure and regmap support to be built into the
kernel while I2C support can still be build as a module.
Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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New codec suooprt of ALC667.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some vendor has special bonding options.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is compatible with ALC255.
It is use for Lenovo.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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These two new pin tables can fix headset mic problems for several
new Dell machines.
And also delete some machines from old quirk table since the existing
pin talbes already cover them.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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From e7a94bb7fb871c73cc85712d89c1f48d0271c1be Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 5 Jun 2014 12:31:28 +0200
Subject: [PATCH] ASoC: MMP audio needs sram support
Building the pxa/mmp audio driver without support for the mmp
sram driver enabled results in this link error:
sound/built-in.o: In function `mmp_pcm_free_dma_buffers':
:(.text+0x3e734): undefined reference to `sram_get_gpool'
sound/built-in.o: In function `mmp_pcm_new':
:(.text+0x3e7c0): undefined reference to `sram_get_gpool'
The sram driver is cannot be manually enabled and needs to
be turned on by selecting MMP_SRAM from each module that
needs it, which is what this patch does.
Ideally, MMP should move over to the generic SRAM support, but
for the moment, we can avoid the build error.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Haojian Zhuang <haojian.zhuang@gmail.com>
Cc: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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New codec support for ALC891.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next
Pull sound updates from Takashi Iwai:
"At this time, majority of changes come from ASoC world while we got a
few new drivers in other places for FireWire and USB. There have been
lots of ASoC core cleanups / refactoring, but very little visible to
external users.
ASoC:
- Support for specifying aux CODECs in DT
- Removal of the deprecated mux and enum macros
- More moves towards full componentisation
- Removal of some unused I/O code
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers
- Several drivers exposed directly in Kconfig for use with
simple-card
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651
and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677
HD-audio:
- Clean up Dell headset quirks
- Noise fixes for Dell and Sony laptops
- Thinkpad T440 dock fix
- Realtek codec updates (ALC293,ALC233,ALC3235)
- Tegra HD-audio HDMI support
FireWire-audio:
- FireWire audio stack enhancement (AMDTP, MIDI), support for
incoming isochronous stream and duplex streams with timestamp
synchronization
- BeBoB-based devices support
- Fireworks-based device support
USB-audio:
- Behringer BCD2000 USB device support
Misc:
- Clean up of a few old drivers, atmel, fm801, etc"
* tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits)
ASoC: Fix wrong argument for card remove callbacks
ASoC: free jack GPIOs before the sound card is freed
ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation
ASoC: cache: Fix error code when not using ASoC level cache
ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup
ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
ASoC: add RT5677 CODEC driver
ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support
ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support
ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
ASoC: Add helper functions to cast from DAPM context to CODEC/platform
ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
ASoC: wm9713: correct mono out PGA sources
ALSA: synth: emux: soundfont.c: Cleaning up memory leak
ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
ASoC: fsl-ssi: Use regmap
ASoC: fsl-ssi: reorder and document fsl_ssi_private
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Sometimes PORT_EXIT messages are lost when a process is exiting.
This happens if you subscribe to the announce port with client A,
then subscribe to the announce port with client B, then kill client A.
Client B will not see the PORT_EXIT message because client A's port is
closing and is earlier in the announce port subscription list. The
for each loop will try to send the announcement to client A and fail,
then will stop trying to broadcast to other ports. Killing B works fine
since the announcement will already have gone to A. The CLIENT_EXIT
message does not get lost.
How to reproduce problem:
*** termA
$ aseqdump -p 0:1
0:1 Port subscribed 0:1 -> 128:0
*** termB
$ aseqdump -p 0:1
*** termA
0:1 Client start client 129
0:1 Port start 129:0
0:1 Port subscribed 0:1 -> 129:0
*** termB
0:1 Port subscribed 0:1 -> 129:0
*** termA
^C
*** termB
0:1 Client exit client 128
<--- expected Port exit as well (before client exit)
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_bebob_stream_map() is not defined.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently mutex_destroy() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.
[fixed a typo in changelog by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The constants of enum snd_efw_grp_type is for struct snd_efw_phys_grp.type.
But this member is 1 byte. Although the value is between 0x00-0xff, a constant
has 0x10000. This constant is meaningless.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It includes descriptions to cause misreading.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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To reverse a pointer for the ring buffer, subtraction by buffer
size is better than assignment to the beginning of the buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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All assignment for local variables in these functions are not related to
critical section.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_seq_event_dup returns -ENOMEM in some buffer-full conditions,
but usually returns -EAGAIN. Make -EAGAIN trigger the overflow
condition in snd_seq_fifo_event_in so that the fifo is cleared
and -ENOSPC is returned to userspace as stated in the alsa-lib docs.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We have in the past added 'depends on I2C' for some of the PXA boards
after hitting randconfig build bugs. I have seens a couple of new
bugs in this area during the linux-next cycle for 3.16, after it
became possible to build some more PXA machines with I2C disabled.
To shut this up for good, this adds the dependency to every board
that uses I2C as the interface to the codec. I have gone through
all board files and verified that they all either use AC97 or
I2C, and this annotates the latter. Some of these already enable
I2C from mach-pxa/Kconfig, but since that can change it's better
to be explicit here.
The link error that can result otherwise happens when CONFIG_I2C
is set to 'm' and the codec driver is built-in as a result of being
selected by the platform specific glue.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip into next
Pull core locking updates from Ingo Molnar:
"The main changes in this cycle were:
- reduced/streamlined smp_mb__*() interface that allows more usecases
and makes the existing ones less buggy, especially in rarer
architectures
- add rwsem implementation comments
- bump up lockdep limits"
* 'locking-core-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (33 commits)
rwsem: Add comments to explain the meaning of the rwsem's count field
lockdep: Increase static allocations
arch: Mass conversion of smp_mb__*()
arch,doc: Convert smp_mb__*()
arch,xtensa: Convert smp_mb__*()
arch,x86: Convert smp_mb__*()
arch,tile: Convert smp_mb__*()
arch,sparc: Convert smp_mb__*()
arch,sh: Convert smp_mb__*()
arch,score: Convert smp_mb__*()
arch,s390: Convert smp_mb__*()
arch,powerpc: Convert smp_mb__*()
arch,parisc: Convert smp_mb__*()
arch,openrisc: Convert smp_mb__*()
arch,mn10300: Convert smp_mb__*()
arch,mips: Convert smp_mb__*()
arch,metag: Convert smp_mb__*()
arch,m68k: Convert smp_mb__*()
arch,m32r: Convert smp_mb__*()
arch,ia64: Convert smp_mb__*()
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The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is
freed] introduced snd_soc_card remove callbacks to a few drivers, but
they are implemented with a wrong argument type. The callback should
receive snd_soc_card pointer instead of snd_soc_pcm_runtime.
Fixes: e1d4d3c854f2 ('ASoC: free jack GPIOs before the sound card is freed')
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.16
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
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This is the same change as commit fb6b8e71448a "ASoC: tegra: free jack
GPIOs before the sound card is freed", but applied to all other ASoC
machine drivers where code inspection indicates the same problem exists.
That commit's description is:
==========
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, guard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
==========
Note that I have not even compile-tested this in most cases, since most
of the drivers rely on specific mach-* support I don't have enabled, and
don't support COMPILE_TEST. Testing by the relevant board maintainers
would be useful.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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into asoc-next
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'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next
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into asoc-next
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asoc-next
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'asoc/topic/max98095' into asoc-next
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asoc-next
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asoc-next
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into asoc-linus
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