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Most callers of snd_wss_pcm(), snd_wss_timer() and snd_cs4236_pcm() pass
NULL as the last parameter, some callers pass a pointer but never use it
after the function has been called and only a few callers pass a pointer and
actually use it. The later is only the case for snd_wss_pcm() for
snd_cs4236_pcm() and it is possible to get the same PCM object by accessing
the pcm field of the snd_wss struct that was passed as the first parameter.
This function removes the last parameters from the functions mentioned above
and updates the callers which used it to use chip->pcm instead. This allows
us to slightly simplify the functions since they don't have to check and set
the last parameter anymore which makes the code slightly shorter and
cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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All callers of snd_sb16dsp_pcm() always pass the pcm field of the first
parameter as the last parameter. Simplify the function by moving this inside
the function itself. This makes the code a bit shorter and cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_sb8dsp_pcm() and snd_sb8dsp_midi() take a pointer to a pointer of a
PCM/MIDI where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_msnd_pcm() takes a pointer to a pointer of a PCM where if this parameter
is provided the newly allocated PCM is stored. All callers pass NULL though,
so remove the parameter. This makes the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_gf1_pcm_new() and snd_gf1_rawmidi_new() take a pointer to a pointer of a
PCM/MIDI where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_es18xx_pcm() takes a pointer to a pointer of a PCM where if this
parameter is provided the newly allocated PCM is stored. All callers pass
NULL though, so remove the parameter. This makes the code a bit cleaner and
shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_es1688_pcm() takes a pointer to a pointer of a PCM where if this
parameter is provided the newly allocated PCM is stored. This PCM is also
available from the pcm field of the snd_es1688 struct that got passed to the
same function. This patch updates all callers which passed a pointer to use
that field instead and then removes the parameter from the function. This
makes the code a bit shorter and cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_ad1816a_pcm() and snd_ad1816a_timer() take a pointer to a pointer of a
PCM/timer where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_ml403_ac97cr_pcm() takes a pointer to a pointer of a PCM where if this
parameter is provided the newly allocated PCM is stored. All callers pass
NULL though, so remove the parameter. This makes the code a bit cleaner and
shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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sound/soc/soc-pcm.c: In function ‘soc_pcm_set_msb’:
sound/soc/soc-pcm.c:307:11: warning: unused variable ‘i’ [-Wunused-variable]
Fixes: 0e2a37513a1f ('ASoC: pcm: Use wildcard msbits constraints')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain.
Some audio devices require notification of drain events
in order to properly drain and shutdown an audio stream.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add missing limits to keep copied data within allocated buffer.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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__fls has the same semantics as ld2, so there is no need to re-implement it.
Furthermore a lot of architectures have custom implementations of __fls that
are able to use special hardware instructions to compute the result. This
makes the code slightly shorter and faster.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Instead of opencoding them use the standard roundup_pow_of_two() and
rounddown_pow_of_two() helper functions. This gets rids one of the few users
of the custom ld2() function and also makes it a bit more obvious what the
code does.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The difference between __ffs and ffs is that ffs will return a one based
index whereas __ffs will return a zero based index. Furthermore ffs will
check if the passed value is zero and return zero in that case, whereas
__ffs behavior is undefined if the passed parameter is 0.
Since we already check if the mask is 0 before calling ffs and also subtract
1 from the result __ffs is the better choice.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The hw_params struct has a parameter that contains the period size in bytes.
This can be used instead of deriving the value from other parameters. This
is similar to e.g. params_buffer_bytes()
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add kernel doc for the remaining undocumented params_*() functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use static inline functions instead of macros for the remaining params_*()
helpers that have not been converted yet. This is slightly cleaner and
offers better type safety.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Both SNDRV_PCM_IOCTL1_FALSE and SNDRV_PCM_IOCTL1_TRUE are unused and have in
fact never been used (at least as far as the git history goes).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Those two functions are not used anywhere and also their name is a bit to
generic to be in a global header, so remove them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix a copy and paste error in the kernel doc description for the params_*()
functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new wildcard msbits constraints instead of installing a constraint
for each available sample format width.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently the msbits constraints requires to specify a specific sample
format width for which the constraint should be applied. But often the
number of most significant bits is not sample format specific, but rather a
absolute limit. E.g. the PCM interface might accept 32-bit and 24-bit
samples, but the DAC has a 16-bit resolution and throws away the LSBs. In
this case for both 32-bit and 24-bit format msbits should be set to 16. This
patch extends snd_pcm_hw_constraint_msbits() so that a wildcard constraint
can be setup that is applied for all formats with a sample width larger than
the specified msbits. Choosing the wildcard constraint is done by setting
the sample width parameter of the function to 0.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If the sound card is made up of discrete components, each with their own
driver (e.g. like in the ASoC case), we might end up with multiple msbits
constraint rules installed. Currently this will result in msbits being set
to whatever the last rule set it to.
This patch updates the behavior of the rule to choose the minimum (other
than zero) of all the installed rules.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
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: Updates for v3.20
Nothing too exciting here yet, a small optimization for DAPM from
Lars-Peter and a few small bits and pieces for drivers but nothing
that really stands out.
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.19
A few fixes for v3.19, a few driver specifics and one core fix which
fixes a boot crash on OMAP if deferred probing kicks in due to
attempting to modify static data.
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at91 will no longer export the mach/cpu.h and mach/hardware.h header files
in the future, which would break building the atmel ac97c driver.
Since the cpu_is_* check is only used to find out whether we are running
on avr32 or arm/at91, we can hardcode that check in the ARM case.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: http://www.spinics.net/lists/arm-kernel/msg382068.html
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The instrument layer codes have been dropped years ago, but this file
was left intentionally with a slight hope for the new implementations.
But, looking at the reality, the probability we'll have a new code for
ISA GUS board is very low, and I won't bet it. So, rather clean this
legacy stuff up. Developers can still refer to the old code via git
history.
Reported-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Removes some functions that are not used anywhere:
ad198x_ch_mode_get() ad198x_ch_mode_info() ad198x_ch_mode_info()
This was partially found by using a static code analysis program called cppcheck.
Signed-off-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Removes some functions that are not used anywhere:
snd_wm8776_set_master_mode() snd_wm8776_set_adc_if() snd_wm8776_set_dac_if()
This was partially found by using a static code analysis program called cppcheck.
Signed-off-by: Rickard Strandqvist <rickard_strandqvist@spectrumdigital.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The total stream number of Skylake's input and output stream
exceeds 15, which will cause some streams do not work because
of the overflow on SDxCTL.STRM field if using the legacy
stream tag allocation method.
This patch uses the new stream tag allocation method by add
the flag AZX_DCAPS_SEPARATE_STREAM_TAG for Skylake platform.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Implemented separate stream_tag assignment for input and output streams.
According to hda specification stream tag must be unique throughout the
input streams group, however an output stream might use a stream tag
which is already in use by an input stream. This change is necessary
to support HW which provides a total of more than 15 stream DMA engines
which with legacy implementation causes an overflow on SDxCTL.STRM
field (and the whole SDxCTL register) and as a result usage of
Reserved value 0 in the SDxCTL.STRM field which confuses HDA controller.
Signed-off-by: Rafal Redzimski <rafal.f.redzimski@intel.com>
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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into asoc-next
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'asoc/topic/rt5670' and 'asoc/topic/rt5677' into asoc-next
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'asoc/topic/omap' into asoc-next
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'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus
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Set Transmit Data Level(TDL) and Receive Data Level(RDL) to 16 samples.
Without this setting, the TDL is default to be 0x00 (means 0 sample),
and the RDL is default to be 0x1f (means 32 samples).
Signed-off-by: Jianqun Xu <jay.xu@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Since RK3288 DMAC's burst length only support max to 4, here
set maxburst of playback and capture dma data to 4.
Signed-off-by: Jianqun Xu <jay.xu@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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According to description about "Transmit Data Level",
This bit field controls the level at which a DMA request
is made by the transmit logic.
It is equal to the watermark level.
That is, the dma_tx_req signal is generated when the number
of valid data entries in the TXFIFO
(TXFIFO0 if CSR=00
TXFIFO1 if CSR=01
TXFIFO2 if CSR=10
TXFIFO3 if CSR=11)
is equal to or below this field value.
Different to receive data level, transmit data level does not need
to "-1".
Signed-off-by: Jianqun Xu <jay.xu@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Remove the duplicated PLL1 connections of the adc stereo filters,
and remove the PLL1 connections of the DACs because the PLL1 should be
connected to dac filters.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add the ASRC support
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add the filter powers of the dac mono3 and mono4, and remove the connection
of dac stereo1 filter that connect to DAC1 MIX.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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To enable some modules from other than base FW, according
to FW interface spec, we need pass the correct entry point
param to FW, so here store the entry_point read from FW
file for later usage.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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