Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Otherwise capture activity on a compressed DAI would mute any playback
on the same DAI.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This patch completes the replacement of the existing max98090 driver,
by installing a more complete driver.
Signed-off-by: Jerry Wong <jerry.wong@maximintegrated.com>
Tested-by: Matthew Mowdy <matthew.mowdy@maximintegrated.com>
Reviewed-by: Ralph Birt <ralph.birt@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes the existing max98090 driver prior to installing a more
complete one.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC compressed API did not implement the copy callback in its
compressed ops which is required for DSPs that are not memory mapped.
This patch creates a local copy of the compress ops for each runtime and
modifies them with a copy callback as appropriate.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the regmap API to use asynchronous I/O where supported to minimise
the delay between transfers, reducing firmware download times.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
disable clock and correct sequence when unload
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
correct the sequence when unload this module
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ssc private data has been set in ssc driver, this cause the error
private data set to ssc, remove it
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When build as module, it reports following error, using this patch
fix it
sound/soc/atmel/atmel-pcm-pdc.c:387: error: redefinition of 'atmel_pcm_pdc_platform_register'
sound/soc/atmel/atmel-pcm.h:95: note: previous definition of 'atmel_pcm_pdc_platform_register' was here
sound/soc/atmel/atmel-pcm-pdc.c:393: error: redefinition of 'atmel_pcm_pdc_platform_unregister'
sound/soc/atmel/atmel-pcm.h:99: note: previous definition of 'atmel_pcm_pdc_platform_unregister' was here
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Previously arizona_calc_fll() was checking if the target frequency is
exactly divisible by reference frequency, but should have been product
of the ratio and the reference frequency.
Also scale down the Lamba and Theta coefficients be under 16-bits in
order to match the registers.
Signed-off-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Some compilers complain when building sound for MOP500:
sound/soc/ux500/mop500.c:27:27:
fatal error: mop500_ab8500.h: No such file or directory
That file is actually in the local directory rather than include.
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit a08485d8 ("ASoC: Samsung: Do not register samsung audio dma
device as pdev") introduced compilation error.
Combination of CONFIG_CPU_S3C2440 and CONFIG_SND_S3C24XX_I2S
caused undefined symbols 's3c_i2sv2_register_dai' and 's3c2412_i2s_dai'
in sound/soc/samsung/s3c24xx-i2s.c.
This patch fixes the problem and makes S3C2440 I2S usable again. It does
not affect S3C2412 (aka I2S-v2).
Signed-off-by: Alexey Galakhov <agalakhov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Playing a mono stream on the TLV320AIC3x results in too fast playback rate.
Remove mono support so that mono streams can be played correctly on this codec.
Tested with imx-ssi (i.MX25) and TLV320AIC3104.
Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For non-snoop mode, we fiddle with the page attributes of CORB/RIRB
and the position buffer, but also the ring buffers. The problem is
that the current code blindly assumes that the buffer is contiguous.
However, the ring buffers may be SG-buffers, thus a wrong vmapped
address is passed there, leading to Oops.
This patch fixes the handling for SG-buffers.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=800701
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we use LPIB forcibly for both playback and capture for
Poulsbo and Oaktrail devices, and this seems rather problematic.
The recent fix for LPIB delay count seems working well with these
devices, so let's enable it instead.
Reported-by: Martin Weishart <martin.weishart@telosalliance.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Maximise robustness for the widest range of firmwares by ensuring the DSP
is in a consistent state when halted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When building modules with CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig,
we will see the following link error.
LD [M] sound/soc/fsl/snd-soc-fsl-ssi.o
LD [M] sound/soc/fsl/snd-soc-fsl-utils.o
LD [M] sound/soc/fsl/snd-soc-imx-ssi.o
LD [M] sound/soc/fsl/snd-soc-imx-audmux.o
LD [M] sound/soc/fsl/snd-soc-imx-pcm.o
sound/soc/fsl/imx-pcm-dma.o: In function `init_module':
imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here
sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module':
imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here
make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1
The module snd-soc-imx-pcm is designed to link imx-pcm.o with
imx-pcm-dma.o or imx-pcm-fiq.o depending on if option SND_SOC_IMX_PCM_DMA
or SND_SOC_IMX_PCM_FIQ is enabled. Both imx-pcm-dma and imx-pcm-fiq
register their own module_platform_driver. However, these two options
are not mutually exclusive and can be enabled together. And that's
why we see above multiple init_module definition error.
Instead of having both imx-pcm-dma and imx-pcm-fiq register their
own platform_driver, we should do only once in imx-pcm.c. Using
platform_device_id to distinguish between imx-pcm-dma and imx-pcm-fiq,
we can run-time call imx-pcm-dma/fiq specific initialization in .probe
hook to have module snd-soc-imx-pcm work for both cases.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 25b8d31488.
While the commit fixes multiple init_module definition error with
module build, it breaks build when both imx-pcm-fiq and imx-pcm-dma
are built in as below.
LD sound/soc/fsl/snd-soc-fsl-ssi.o
LD sound/soc/fsl/snd-soc-fsl-utils.o
LD sound/soc/fsl/snd-soc-imx-ssi.o
LD sound/soc/fsl/snd-soc-imx-audmux.o
LD sound/soc/fsl/snd-soc-imx-pcm-fiq.o
LD sound/soc/fsl/snd-soc-imx-pcm-dma.o
LD sound/soc/fsl/snd-soc-eukrea-tlv320.o
LD sound/soc/fsl/snd-soc-imx-sgtl5000.o
LD sound/soc/fsl/snd-soc-imx-mc13783.o
LD sound/soc/fsl/built-in.o
sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_free':
imx-pcm.c:(.text+0x464): multiple definition of `imx_pcm_free'
sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x1a8): first defined here
sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `snd_imx_pcm_mmap':
imx-pcm.c:(.text+0x35c): multiple definition of `snd_imx_pcm_mmap'
sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0xa0): first defined here
sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_new':
imx-pcm.c:(.text+0x3dc): multiple definition of `imx_pcm_new'
sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x120): first defined here
make[4]: *** [sound/soc/fsl/built-in.o] Error 1
Let's revert the commit and find a proper fix for multiple init_module
definition error later.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 23caaf19b1 (ALSA: usb-mixer: Add support for Audio Class v2.0)
forgot to adjust the length check for UAC 2.0 feature unit descriptors.
This would make the code abort on encountering a feature unit without
per-channel controls, and thus prevented the driver to work with any
device having such a unit, such as the RME Babyface or Fireface UCX.
Reported-by: Florian Hanisch <fhanisch@uni-potsdam.de>
Tested-by: Matthew Robbetts <wingfeathera@gmail.com>
Tested-by: Michael Beer <beerml@sigma6audio.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: 2.6.35+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usual set of driver updates, nothing too thrilling in here - one
core change for the regulator bypass mode which was just not doing the
right thing at all and a bunch of driver specifics.
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Merge tag 'asoc-3.8-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.8-rc4
The usual set of driver updates, nothing too thrilling in here - one
core change for the regulator bypass mode which was just not doing the
right thing at all and a bunch of driver specifics.
u32 rotate = (32 - word_length) / 4;
This implementation is wrong, but it works only for 16, or 32 bit audio data.
(rotation for 16 or 32 bit is same as in code I present) Mcasp rotated data in
4 bits (max value 0x7)and then masks them . That data are sended to i2s bus.
For 24 bit or 20 bit or other data formats, this code rotates data badly and
you hear somethink like noise. You need to use
u32 rotate = (word_length / 4) & 0x7;
to proper data rotation.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There were missing break statements so everything used
TWO_CHANNEL_SUPPORT.
Also I added a return statement to silence a GCC warning:
sound/soc/dwc/designware_i2s.c: In function ‘dw_i2s_hw_params’:
sound/soc/dwc/designware_i2s.c:236:32: warning: ‘ch_reg’ may be
used uninitialized in this function
[-Wmaybe-uninitialized]
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
