When using right_j format and the codec is slave it can support 16bit
format, but only if slot_width == 16, in the same DAI mode the 24 bit
audio can work with 24 or 32 slot_width.
Because of this, the codec and CPU needs to be reconfigured when the sample
format changes.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190812095226.18870-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAC and ADC path of the codec is independent, have dedicated LRCK (FS)
and BCK for DAC/ADC.
They can be configured to use different format, TDM slots and slot_width if
needed.
Move these parameters under dedicated io_params structure and manage them
independently based on the dai.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190812095226.18870-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
hdac_hdmi_present_sense() calls the audio component to get ELD update,
then it reports the jack status change and updates DAPM graph
accordingly. This works when it's called from the normal code paths.
However, it may lead to a dead lock when it's called from the audio
component notifier. Namely, the DAPM update involves with the runtime
PM, and it eventually calls again the audio component get_power()
ops. Since i915 driver already takes a mutex around the audio
component ops calls, we'll eventually get the mutex doubly.
As a workaround, in this patch, only the jack state is updated in the
code path from hdac_hdmi_eld_notify_cb(), and the DAPM update is
deferred to a work so that it's processed in another context.
Reported-by: Imre Deak <imre.deak@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20190809151531.24359-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull sound fixes from Takashi Iwai:
"Lots of small fixes at this time since we've received the ASoC fix
batch now.
- Some coverage in ASoC core mostly for minor issues like NULL checks
for DPCM and proper error handling in DAI instantiation
- A collection of small device-specific changes in various ASoC codec
and platform drivers
- OF-tree refcount fixes in a few ASoC drivers
- Fixes of memory leaks in the error paths of various ASoC / ALSA
drivers
- A workaround for a long-standing issue on AMD HD-audio device
- Updates of MAINTAINERS, mail addresses, file permission fixups"
* tag 'sound-5.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (38 commits)
ALSA: firewire: fix a memory leak bug
sound: fix a memory leak bug
ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)
ALSA: hiface: fix multiple memory leak bugs
ALSA: hda - Don't override global PCM hw info flag
ALSA: usb-audio: fix a memory leak bug
ASoC: max98373: Remove executable bits
ASoC: amd: acp3x: use dma address for acp3x dma driver
ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver
ASoC: max98373: add 88200 and 96000 sampling rate support
ASoC: sun4i-i2s: Incorrect SR and WSS computation
MAINTAINERS: Update Intel ASoC drivers maintainers
ASoC: ti: davinci-mcasp: Correct slot_width posed constraint
ASoC: rockchip: Fix mono capture
ASoC: Intel: Fix some acpi vs apci typo in somme comments
ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode
ASoC: Fail card instantiation if DAI format setup fails
ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread
ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links
ASoC: dapm: fix a memory leak bug
...
In file included from ./include/sound/tlv.h:10:0,
from sound/soc/codecs/ml26124.c:19:
sound/soc/codecs/ml26124.c:59:35: warning: ngth defined but not used [-Wunused-const-variable=]
static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
^
./include/uapi/sound/tlv.h:64:15: note: in definition of macro SNDRV_CTL_TLVD_DECLARE_DB_SCALE
unsigned int name[] = { \
^~~~
sound/soc/codecs/ml26124.c:59:14: note: in expansion of macro DECLARE_TLV_DB_SCALE
static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
^~~~~~~~~~~~~~~~~~~~
It is never used, so can be removed.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20190809082440.67412-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Fixes for v5.3
A relatively large batch of mostly unremarkable fixes here, a couple of
small core fixes for fairly obscure issues, more comment/email updates
with no code impact than usual and a bunch of small driver fixes.
The support for new sample rates in the max98373 driver is a fix for the
fact that the driver declared support for those rates but would in fact
return an error if these rates were selected.
We don't need dev_err() messages when platform_get_irq() fails now that
platform_get_irq() prints an error message itself when something goes
wrong. Let's remove these prints with a simple semantic patch.
// <smpl>
@@
expression ret;
struct platform_device *E;
@@
ret =
(
platform_get_irq(E, ...)
|
platform_get_irq_byname(E, ...)
);
if ( \( ret < 0 \| ret <= 0 \) )
{
(
-if (ret != -EPROBE_DEFER)
-{ ...
-dev_err(...);
-... }
|
...
-dev_err(...);
)
...
}
// </smpl>
While we're here, remove braces on if statements that only have one
statement (manually).
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Link: https://lore.kernel.org/r/20190730181557.90391-50-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
VAG power control is improved to fit the manual [1]. This patch fixes as
minimum one bug: if customer muxes Headphone to Line-In right after boot,
the VAG power remains off that leads to poor sound quality from line-in.
I.e. after boot:
- Connect sound source to Line-In jack;
- Connect headphone to HP jack;
- Run following commands:
$ amixer set 'Headphone' 80%
$ amixer set 'Headphone Mux' LINE_IN
Change VAG power on/off control according to the following algorithm:
- turn VAG power ON on the 1st incoming event.
- keep it ON if there is any active VAG consumer (ADC/DAC/HP/Line-In).
- turn VAG power OFF when there is the latest consumer's pre-down event
come.
- always delay after VAG power OFF to avoid pop.
- delay after VAG power ON if the initiative consumer is Line-In, this
prevents pop during line-in muxing.
According to the data sheet [1], to avoid any pops/clicks,
the outputs should be muted during input/output
routing changes.
[1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf
Cc: stable@vger.kernel.org
Fixes: 9b34e6cc3b ("ASoC: Add Freescale SGTL5000 codec support")
Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com>
Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20190719100524.23300-3-oleksandr.suvorov@toradex.com
Signed-off-by: Mark Brown <broonie@kernel.org>
