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[ Upstream commit 576ce4075bfa0f03e0e91a89eecc539b3b828b08 ]
gcc notices that without either the ac97 bus or the pdata, we never
initialize the regmap pointer, which leads to an uninitialized variable
access:
sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe':
sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized]
Since that configuration is invalid, it's better to return an error
here. I tried to avoid adding complexity to the conditions, and turned
the #ifdef into a regular if(IS_ENABLED()) check for readability.
This in turn requires moving some header file declarations out of
an #ifdef.
The same code is used in three drivers, all of which I'm changing
the same way.
Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit cfc8f568aada98f9608a0a62511ca18d647613e2 upstream.
Prepare to use SND_SOC_DAPM_PRE_POST_PMU definition to
reduce coming code size and make it more readable.
Cc: stable@vger.kernel.org
Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Reviewed-by: Igor Opaniuk <igor.opaniuk@toradex.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20190719100524.23300-2-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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[ Upstream commit 4475f8c4ab7b248991a60d9c02808dbb813d6be8 ]
A previous fix to the stop handling on compressed capture streams causes
some knock on issues. The previous fix updated snd_compr_drain_notify to
set the state back to PREPARED for capture streams. This causes some
issues however as the handling for snd_compr_poll differs between the
two states and some user-space applications were relying on the poll
failing after the stream had been stopped.
To correct this regression whilst still fixing the original problem the
patch was addressing, update the capture handling to skip the PREPARED
state rather than skipping the SETUP state as it has done until now.
Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit 4f2ab5e1d13d6aa77c55f4914659784efd776eb4 upstream.
It is normal user behaviour to start, stop, then start a stream
again without closing it. Currently this works for compressed
playback streams but not capture ones.
The states on a compressed capture stream go directly from OPEN to
PREPARED, unlike a playback stream which moves to SETUP and waits
for a write of data before moving to PREPARED. Currently however,
when a stop is sent the state is set to SETUP for both types of
streams. This leaves a capture stream in the situation where a new
start can't be sent as that requires the state to be PREPARED and
a new set_params can't be sent as that requires the state to be
OPEN. The only option being to close the stream, and then reopen.
Correct this issues by allowing snd_compr_drain_notify to set the
state depending on the stream direction, as we already do in
set_params.
Fixes: 49bb6402f1aa ("ALSA: compress_core: Add support for capture streams")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 5363857b916c1f48027e9b96ee8be8376bf20811 upstream.
As addressed in alsa-lib (commit b420056604f0), we need to fix the
case where the evaluation of PCM interval "(x x+1]" leading to
-EINVAL. After applying rules, such an interval may be translated as
"(x x+1)".
Fixes: ff2d6acdf6f1 ("ALSA: pcm: Fix snd_interval_refine first/last with open min/max")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.19
This is the usual set of small fixes scatterd around various drivers,
plus one fix for DAPM and a UAPI build fix. There's not a huge amount
that stands out here relative to anything else.
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Internally, skl_init_chip() calls snd_hdac_bus_init_chip() which
1) sets bus->chip_init to prevent multiple entrances before device
is stopped; 2) enables interrupt.
We shouldn't use it for the purpose of resetting device only because
1) when we really want to initialize device, we won't be able to do
so; 2) we are ready to handle interrupt yet, and kernel crashes when
interrupt comes in.
Rename azx_reset() to snd_hdac_bus_reset_link(), and use it to reset
device properly.
Fixes: 60767abcea3d ("ASoC: Intel: Skylake: Reset the controller in probe")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit a655de808cbde ("ASoC: core: Allow topology to override
machine driver FE DAI link config.") caused soc_dai_hw_params to
be come dependent on the substream private_data being set with
a pointer to the snd_soc_pcm_runtime. Currently, CODEC to CODEC
links don't set this, which causes a NULL pointer dereference:
[<4069de54>] (soc_dai_hw_params) from
[<40694b68>] (snd_soc_dai_link_event+0x1a0/0x380)
Since the ASoC core in general assumes that the substream
private_data will be set to a pointer to the snd_soc_pcm_runtime,
update the CODEC to CODEC links to respect this.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.19
A fairly big update, including quite a bit of core activity this time
around (which is good to see) along with a fairly large set of new
drivers.
- A new snd_pcm_stop_xrun() helper which is now used in several
drivers.
- Support for providing name prefixes to generic component nodes.
- Quite a few fixes for DPCM as it gains a bit wider use and more
robust testing.
- Generalization of the DIO2125 support to a simple amplifier driver.
- Accessory detection support for the audio graph card.
- DT support for PXA AC'97 devices.
- Quirks for a number of new x86 systems.
- Support for AM Logic Meson, Everest ES7154, Intel systems with
RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707.
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DRM based audio components get registered inside the component framework
bind callback. However component framework has a big mutex lock taken for
every call to component_add, component_del and bind, unbind callbacks.
This can lead to deadlock situation if we are trying to add new/remove
component within a bind/unbind callbacks. Which is what was happening
with bcm2837 rpi 3.
Revert this change till we sort out the mutex issue.
Reported-by: Guillaume Tucker <guillaume.tucker@collabora.com>
Reported-by: Stefan Wahren <stefan.wahren@i2se.com>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_midi_event_encode_byte() can never fail, and it can return rather
true/false. Change the return type to bool, adjust the argument to
receive a MIDI byte as unsigned char, and adjust the comment
accordingly. This allows callers to drop error checks, which
simplifies the code.
Meanwhile, snd_midi_event_encode() helper is used only in seq_midi.c,
and it can be better folded into it. This will reduce the total
amount of lines in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-4.19
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The trigger flag in vmidi object can be referred in different contexts
concurrently, hence it's better to be put with READ_ONCE() and
WRITE_ONCE() macros to assure the accesses.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The virmidi sequencer stuff tries to translate the rawmidi bytes to
sequencer events and deliver the packets at trigger callback. The
amount of the whole process of these translations and deliveries
depends on the incoming rawmidi bytes, and we have no limit for that;
this was the cause of a CPU soft lockup that had been reported and
fixed recently.
Although we've fixed the soft lockup by putting the temporary unlock
and cond_resched(), it's rather a quick band aid. In this patch,
meanwhile, the event parsing and delivery process is offloaded to a
dedicated work, and the trigger callback just kicks it off. It has
three merits, at least:
- The processing is always done in a sleepable context, which can
assure the event delivery with non-atomic flag without hackish
is_atomic() usage.
- Other relevant codes can be simplified, reducing the lines
- It makes me happier
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent fix moved the inline snd_sgbuf_aligned_pages() outside the
ifdef, and this triggered a build error on some architectures due to
the undefined PAGE_SIZE, as spotted by 0day bot.
Fix it by adding the missing header inclusion.
Fixes: 4cae99d9b530 ("ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally")
Reported-by: kbuild test robot <lkp@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The PCM format type is with __bitwise, and it can't be converted from
integer implicitly. Instead of an ugly cast, declare the function
argument of snd_sb_csp_autoload() with the proper snd_pcm_format_t
type.
This fixes the sparse warnings like:
sound/isa/sb/sb16_csp.c:743:22: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The PCM format type is defined with __bitwise, hence it can't be
passed as integer but needs an explicit cast. In this patch, instead
of the messy cast flood, define the format argument of
snd_hdac_calc_stream_format() to be the proper snd_pcm_format_t type.
This fixes sparse warnings like:
sound/hda/hdac_device.c:760:38: warning: incorrect type in argument 1 (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Many drivers calling snd_mask_set() need to do ugly cast with __force
for shutting up the sparse warnings. Actually almost all of them are
about setting the format, so it's far better to provide a common
helper snd_mask_set_format() to pass SNDRV_PCM_FORMAT_* directly
without the cast.
There are a few other calls of snd_mask_set(), but they are in the PCM
core code, so we leave them for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-4.19
ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally
Make this helper inline function available for all platforms. This
helps solve 0-day compilation issues when CONFIG_SND_DMA_SGBUF is not
defined.
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Make this helper inline function available for all platforms. This
helps solve 0-day compilation issues when CONFIG_SND_DMA_SGBUF is not
defined.
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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The snd_pcm_lib_read() and snd_pcm_lib_write() inline functions have
the explicit cast from a user pointer to a kernel pointer, but they
lacks of __force prefix.
This fixes sparse warnings like:
./include/sound/pcm.h:1093:47: warning: cast removes address space of expression
Fixes: 68541213720d ("ALSA: pcm: Direct in-kernel read/write support")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On Qualcomm platforms, specifically with SLIMbus interfaced codecs,
the codec slim channel numbers are passed to DSP while configuring
the slim audio path. Having get_channel_map() would allow dais to
share such information across multiple dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot()
directly because it needs to handle 4 mask for each direction.
Yet the parsing of each mask is the same, so export
snd_soc_of_get_slot_mask() to reuse the the existing code.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Pull the generic drm_audio_component support, which will be used later
for AMD/ATI and other HD-audio HDMI codec drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch aims at achieving dynamic behaviour of audio card when
the dependent components disappear and reappear.
With this patch the card is removed if any of the dependent component
is removed and card is added back if the dependent component comes back.
All this is done using component framework and matching based on
component name.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with
somewhat special for vmalloc handling, but in the end, this turned to
just the default handler, i.e. NULL. As the situation has never
changed over decades, let's rip it off.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is the final step for more generic support of DRM audio
component. The generic audio component code is now moved to its own
file, and the symbols are renamed from snd_hac_i915_* to
snd_hdac_acomp_*, respectively. The generic code is enabled via the
new kconfig, CONFIG_SND_HDA_COMPONENT, while CONFIG_SND_HDA_I915 is
kept as the super-class.
Along with the split, three new callbacks are added to audio_ops:
pin2port is for providing the conversion between the pin number and
the widget id, and master_bind/master_unbin are called at binding /
unbinding the master component, respectively. All these are optional,
but used in i915 implementation and also other later implementations.
A note about the new snd_hdac_acomp_init() function: there is a slight
difference between this and the old snd_hdac_i915_init(). The latter
(still) synchronizes with the master component binding, i.e. it
assures that the relevant DRM component gets bound when it returns, or
gives a negative error. Meanwhile the new function doesn't
synchronize but just leaves as is. It's the responsibility by the
caller's side to synchronize, or the caller may accept the
asynchronous binding on the fly.
v1->v2: Fix missing NULL check in master_bind/unbind
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HD-audio i915 binding code contains a single pointer, hdac_acomp,
for allowing the access to audio component from the master bind/unbind
callbacks. This was needed because the callbacks pass only the device
pointer and we can't guarantee the object type assigned to the drvdata
(which is free for each controller driver implementation).
And this implementation will be a problem if we support multiple
components for different DRM drivers, not only i915.
As a solution, allocate the audio component object via devres and
associate it with the given device, so that the component callbacks
can refer to it via devres_find().
The removal of the object is still done half-manually via
devres_destroy() to make the code consistent (although it may work
without the explicit call).
Also, the snd_hda_i915_register_notifier() had the reference to
hdac_acomp as well. In this patch, the corresponding code is removed
by passing hdac_bus object to the function, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For allowing other drivers to use the DRM audio component, rename the
i915_audio_component_* with drm_audio_component_*, and split the
generic part into drm_audio_component.h. The i915 specific stuff
remains in struct i915_audio_component, which contains
drm_audio_component as the base.
The license of drm_audio_component.h is kept to MIT as same as the the
original i915_component.h.
This is a preliminary change for further development, and no
functional changes by this patch itself, merely code-split and
renames.
v1->v2: Use SPDX for drm_audio_component.h, fix remaining i915
argument in drm_audio_component.h
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As done for format and channels, add the possibility to merge
the backend rates on the frontend rates.
This useful if the backend does not support all rates supported by the
frontend, or if several backends (cpu and codecs) with different
capabilities are connected to the same frontend.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO.
This needs to be configurable for modern hardware like DSPs where no
pointer update in milliseconds can indicate terminal DSP errors.
Add a substream variable to set the wait time in ms. This allows userspace
and drivers to recover more quickly from terminal DSP errors.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Machine drivers statically define a number of DAI links that currently
cannot be changed or removed by topology. This means PCMs and platform
components cannot be changed by topology at runtime AND machine drivers
are tightly coupled to topology.
This patch allows topology to override the machine driver DAI link config
in order to reuse machine drivers with different topologies and platform
components. The patch supports :-
1) create new FE PCMs with a topology defined PCM ID.
2) destroy existing static FE PCMs
3) change the platform component driver.
4) assign any new HW params fixups.
5) assign a new card name prefix to differentiate this topology to userspace.
The patch requires no changes to the machine drivers, but does add some
platform component flags that the platform component driver can assign
before loading topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge the development branch for HD-audio ext bus refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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To get rid of some intermediate platform layers, move pxa2xx_soc_pcm_new()
and pxa2xx_pcm_ops in pxa2xx-lib.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Clean up the namespace a bit and drop the __ prefix of all functions
exported by pxa2xx-lib. This improves the readability of the code.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add extended ops in the hdac_bus to allow calling the ASoC HDAC library
ops to reuse the legacy HDA codec drivers with ASoC framework.
Extended ops are used by the legacy codec drivers to call into
hdac_hda library, in the subsequent patches..
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Remove memory allocation within snd_hdac_ext_bus_device_init, to make
its behaviour identical to snd_hdac_bus_device_init. So that caller
can allocate the parent data structure containing hdac_device.
This API change helps in reusing the legacy HDA codec drivers with
ASoC platform drivers.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch removes the hdac_ext_driver structure. The legacy and
enhanced HDaudio capabilities can be handled in a backward-compatible
way without separate definitions.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Arnd Bergmann
Oleksandr Suvorov
Charles Keepax
Takashi Iwai
Yu Zhao
Srinivas Kandagatla
Kuninori Morimoto
Mark Brown
Pierre-Louis Bossart
Jerome Brunet
Liam Girdwood
Daniel Mack
Rakesh Ughreja