To be more readable code, this patch adds
new for_each_dpcm_fe() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_comp_order() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_components() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_rtds() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_links() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_prelinks() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 0b7990e389 ("ASoC: add for_each_rtd_codec_dai() macro")
added for_each_rtd_codec_dai_reverse(). but _rollback() is better
naming than _reverse(). This patch rename it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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: 60767abcea ("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>
Current behaviour of ASoC core w.r.t to component removal is that it
unregisters dependent sound card totally. There is no support to
rebind the card if the component comes back.
Typical use case is DSP restart or kernel modules itself.
With this patch, core now maintains list of cards that are unbind due to
any of its depended components are removed and card not unregistered yet.
This list is cleared when the card is rebind successfully or when the
card is unregistered from machine driver.
This list of unbind cards are tried to bind once again after every new
component is successfully added, giving a fair chance for card bind
to be successful.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC is counting card->dai_link_list user,
but no-one is using it.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Larger CODECs may contain many several hundred widgets and which set of
parameters is selected only needs to be recorded on a per DAI basis. As
such move the selected CODEC to CODEC link params to be stored in the
runtime rather than the DAPM widget, to save some memory.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, snd_soc_dapm_connect_dai_link_widgets connects up the routes
representing normal DAIs, however CODEC to CODEC links are hooked up
through separate infrastructure in soc_link_dai_widgets. Improve the
consistency of the code by using snd_soc_dapm_connect_dai_link for both
types of DAIs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai.
To be more readable code, this patch adds
new for_each_rtd_codec_dai() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC snd_soc_dai_link has snd_soc_dai_link_component array
for codecs.
To be more readable code, this patch adds
new for_each_link_codecs() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-card-util for platform.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current struct snd_soc_dai_link is supporting multicodec,
and it is supporting legacy style of
codec_name
codec_of_node
code_dai_name
This is handled as single entry of multicodec.
We don't have multicpu support yet, but in the future we will.
In such case, we can use snd_soc_dai_link_component for both
cpu/codec. Then the code will be more simple and readble.
As next step, we want to use it for platform, too.
This patch adds snd_soc_dai_link_component style for platform.
We might have multiplatform support in the future, but we
don't know yet. To avoid un-known issue / complex code,
this patch supports just single-platform as 1st step.
If we could use snd_soc_dai_link_component for all CPU/Codec/Platform,
we will switch to new style, and remove legacy code.
This is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple_card_utils for codec.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Expose a table containing machine driver information for HDAudio-based
platforms handled by ASoC on Intel hardware.
We only set constant values that are valid across multiple
platforms. The firmware name used by the DSP will be set dynamically
for each platform.
The table is made of a single entry for now, if we need more
complicated set-up where HDAudio is mixed with ACPI-enumerated devices
(I2C, SoundWire) then we'd expect the differentiation to be handled
through information provided by the BIOS (as done for KBL
Chromebooks).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As suggested by Takashi, move this header file to make it easier
to include from e.g. the Intel Skylake driver in follow-up patches
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a655de808c ("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>
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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Merge tag 'asoc-v4.19' of 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.
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>
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>
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>
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>
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: 4cae99d9b5 ("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>
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>
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>
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>
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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Merge tag 'alsa-add-snd-sgbuf-aligned-pages' of 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.
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>
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: 6854121372 ("ALSA: pcm: Direct in-kernel read/write support")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
