18 Apr, 2019
3 commits
-
As per commit e99e88a9d2b0 ("treewide: setup_timer() -> timer_setup()")
converts setup_timer() API into using timer_setup().Signed-off-by: Vipul Kumar
-
In order to support multi-fifo sdma script, the audio driver need to send
the fifo number to dma driver through dma_slave_config, and the cpu_dai
driver should config fifo_num for the audio platform driver, then platform
driver can config fifo_num to dma.
So add new variable fifo_num for struct snd_dmaengine_dai_dma_data.Signed-off-by: Shengjiu Wang
Reviewed-by: Robin Gong
Signed-off-by: Vipul Kumar
Signed-off-by: Srikanth Krishnakar -
cherry-pick below patch from imx_3.14.y
ENGR00307835-2 ASoC: dmaengine: Add two function for dmaengine_pcmAdd check_xrun and device_reset for dmaengine_pcm
Signed-off-by: Shengjiu Wang
(cherry picked from commit 405588f31e432e5c6634a0817e4d804ead87882b)
Signed-off-by: Vipul Kumar
Signed-off-by: Srikanth Krishnakar
13 Feb, 2019
1 commit
-
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
Cc:
Signed-off-by: Takashi Iwai
Signed-off-by: Greg Kroah-Hartman
13 Dec, 2018
1 commit
-
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:
Signed-off-by: Takashi Iwai
Signed-off-by: Greg Kroah-Hartman
18 Sep, 2018
1 commit
-
…oonie/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.
12 Sep, 2018
1 commit
-
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
Signed-off-by: Yu Zhao
Signed-off-by: Mark Brown
15 Aug, 2018
1 commit
-
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:[] (soc_dai_hw_params) from
[] (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
Signed-off-by: Mark Brown
13 Aug, 2018
1 commit
-
…e/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.
03 Aug, 2018
1 commit
-
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
Reported-by: Stefan Wahren
Signed-off-by: Srinivas Kandagatla
Signed-off-by: Mark Brown
02 Aug, 2018
2 commits
-
Signed-off-by: Kuninori Morimoto
Signed-off-by: Mark Brown -
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
01 Aug, 2018
1 commit
-
…/kernel/git/tiwai/sound into asoc-4.19
30 Jul, 2018
3 commits
-
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
-
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
-
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
Cc: Pierre-Louis Bossart
Signed-off-by: Takashi Iwai
Signed-off-by: Mark Brown
27 Jul, 2018
5 commits
-
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 integerSigned-off-by: Takashi Iwai
-
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
-
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
Signed-off-by: Mark Brown -
…inux/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
Signed-off-by: Pierre-Louis Bossart
Reviewed-by: Takashi Iwai
Signed-off-by: Mark Brown
26 Jul, 2018
1 commit
-
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 expressionFixes: 68541213720d ("ALSA: pcm: Direct in-kernel read/write support")
Signed-off-by: Takashi Iwai
24 Jul, 2018
1 commit
-
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
Signed-off-by: Mark Brown
21 Jul, 2018
1 commit
-
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
Signed-off-by: Mark Brown
20 Jul, 2018
1 commit
-
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
18 Jul, 2018
5 commits
-
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
Reviewed-by: Vinod Koul
Signed-off-by: Mark Brown -
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
Signed-off-by: Takashi Iwai -
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
-
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
-
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.hReviewed-by: Rodrigo Vivi
Signed-off-by: Takashi Iwai
11 Jul, 2018
1 commit
-
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
Signed-off-by: Mark Brown
06 Jul, 2018
1 commit
-
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
Signed-off-by: Takashi Iwai
03 Jul, 2018
2 commits
-
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
Signed-off-by: Mark Brown -
Merge the development branch for HD-audio ext bus refactoring.
Signed-off-by: Takashi Iwai
02 Jul, 2018
6 commits
-
Signed-off-by: Kuninori Morimoto
Signed-off-by: Mark Brown -
Signed-off-by: Kuninori Morimoto
Signed-off-by: Mark Brown -
Signed-off-by: Kuninori Morimoto
Signed-off-by: Mark Brown -
Signed-off-by: Kuninori Morimoto
Signed-off-by: Mark Brown -
Signed-off-by: Kuninori Morimoto
Signed-off-by: Mark Brown -
Signed-off-by: Kuninori Morimoto
Signed-off-by: Mark Brown