mirror of
https://github.com/raspberrypi/linux.git
synced 2025-12-24 19:12:14 +00:00
e48c5462e65b5b901b81b9a3ec23565c67da8bbb
21 Commits
| Author | SHA1 | Message | Date | |
|---|---|---|---|---|
Florian Meier
|
e34d08fd1a |
Add support for all the downstream rpi sound card drivers
ASoC: Add support for Rpi-DAC ASoC: Add prompt for ICS43432 codec Without a prompt string, a config setting can't be included in a defconfig. Give CONFIG_SND_SOC_ICS43432 a prompt so that Pi soundcards can use the driver. Signed-off-by: Phil Elwell <phil@raspberrypi.org> Add IQaudIO Sound Card support for Raspberry Pi Set a limit of 0dB on Digital Volume Control The main volume control in the PCM512x DAC has a range up to +24dB. This is dangerously loud and can potentially cause massive clipping in the output stages. Therefore this sets a sensible limit of 0dB for this control. Allow up to 24dB digital gain to be applied when using IQAudIO DAC+ 24db_digital_gain DT param can be used to specify that PCM512x codec "Digital" volume control should not be limited to 0dB gain, and if specified will allow the full 24dB gain. Modify IQAudIO DAC+ ASoC driver to set card/dai config from dt Add the ability to set the card name, dai name and dai stream name, from dt config. Signed-off-by: DigitalDreamtime <clive.messer@digitaldreamtime.co.uk> IQaudIO: auto-mute for AMP+ and DigiAMP+ IQAudIO amplifier mute via GPIO22. Add dt params for "one-shot" unmute and auto mute. Revision 2, auto mute implementing HiassofT suggestion to mute/unmute using set_bias_level, rather than startup/shutdown.... "By default DAPM waits 5 seconds (pmdown_time) before shutting down playback streams so a close/stop immediately followed by open/start doesn't trigger an amp mute+unmute." Tested on both AMP+ (via DAC+) and DigiAMP+, with both options... dtoverlay=iqaudio-dacplus,unmute_amp "one-shot" unmute when kernel module loads. dtoverlay=iqaudio-dacplus,auto_mute_amp Unmute amp when ALSA device opened by a client. Mute, with 5 second delay when ALSA device closed. (Re-opening the device within the 5 second close window, will cancel mute.) Revision 4, using gpiod. Revision 5, clean-up formatting before adding mute code. - Convert tab plus 4 space formatting to 2x tab - Remove '// NOT USED' commented code Revision 6, don't attempt to "one-shot" unmute amp, unless card is successfully registered. Signed-off-by: DigitalDreamtime <clive.messer@digitaldreamtime.co.uk> ASoC: iqaudio-dac: fix S24_LE format Remove set_bclk_ratio call so 24-bit data is transmitted in 24 bclk cycles. Signed-off-by: Matthias Reichl <hias@horus.com> ASoC: iqaudio-dac: use modern dai_link style Signed-off-by: Matthias Reichl <hias@horus.com> Added support for HiFiBerry DAC+ The driver is based on the HiFiBerry DAC driver. However HiFiBerry DAC+ uses a different codec chip (PCM5122), therefore a new driver is necessary. Add support for the HiFiBerry DAC+ Pro. The HiFiBerry DAC+ and DAC+ Pro products both use the existing bcm sound driver with the DAC+ Pro having a special clock device driver representing the two high precision oscillators. An addition bug fix is included for the PCM512x codec where by the physical size of the sample frame is used in the calculation of the LRCK divisor as it was found to be wrong when using 24-bit depth sample contained in a little endian 4-byte sample frame. Limit PCM512x "Digital" gain to 0dB by default with HiFiBerry DAC+ 24db_digital_gain DT param can be used to specify that PCM512x codec "Digital" volume control should not be limited to 0dB gain, and if specified will allow the full 24dB gain. Add dt param to force HiFiBerry DAC+ Pro into slave mode "dtoverlay=hifiberry-dacplus,slave" Add 'slave' param to use HiFiBerry DAC+ Pro in slave mode, with Pi as master for bit and frame clock. Signed-off-by: DigitalDreamtime <clive.messer@digitaldreamtime.co.uk> Fixed a bug when using 352.8kHz sample rate Signed-off-by: Daniel Matuschek <daniel@hifiberry.com> ASoC: pcm512x: revert downstream changes This partially reverts commit |
||
Kuninori Morimoto
|
03c0f1b5e1 |
ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87r1tlwiwe.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
||
Shengjiu Wang
|
b429ca4940 |
ASoC: cs42xx8: Force suspend/resume during system suspend/resume
Use force_suspend/resume to make sure clocks are disabled/enabled accordingly during system suspend/resume. Signed-off-by: Dong Aisheng <aisheng.dong@nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com> Link: https://lore.kernel.org/r/1566944026-18113-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown <broonie@kernel.org> |
||
|
Shengjiu Wang
|
48dfd37a0f |
ASoC: cs42xx8: Fix MFREQ selection issue for async mode
When sample rate of TX is different with sample rate of RX in
async mode, the MFreq selection will be wrong.
For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz.
Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX
and RX instance, the correct value of MFreq is 2 for both TX and RX.
But original method will cause MFreq = 0 for TX, MFreq = 2 for RX.
If TX is started after RX, RX will be impacted, RX work abnormal with
MFreq = 0.
This patch is to select proper MFreq value according to TX rate and
RX rate.
Fixes:
|
||
|
Shengjiu Wang
|
7cda622350 |
ASoC: cs42xx8: Fix build error with CONFIG_GPIOLIB is not set
config: x86_64-randconfig-x000201921-201921
compiler: gcc-7 (Debian 7.3.0-1) 7.3.0
reproduce:
make ARCH=x86_64
sound/soc/codecs/cs42xx8.c: In function ‘cs42xx8_probe’:
sound/soc/codecs/cs42xx8.c:472:25: error: implicit declaration of function ‘devm_gpiod_get_optional’; did you mean ‘devm_clk_get_optional’? [-Werror=implicit-function-declaration]
cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset",
^~~~~~~~~~~~~~~~~~~~~~~
devm_clk_get_optional
sound/soc/codecs/cs42xx8.c:473:8: error: ‘GPIOD_OUT_HIGH’ undeclared (first use in this function); did you mean ‘GPIOF_INIT_HIGH’?
GPIOD_OUT_HIGH);
^~~~~~~~~~~~~~
GPIOF_INIT_HIGH
sound/soc/codecs/cs42xx8.c:473:8: note: each undeclared identifier is reported only once for each function it appears in
sound/soc/codecs/cs42xx8.c:477:2: error: implicit declaration of function ‘gpiod_set_value_cansleep’; did you mean ‘gpio_set_value_cansleep’? [-Werror=implicit-function-declaration]
gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0);
^~~~~~~~~~~~~~~~~~~~~~~~
gpio_set_value_cansleep
Fixes:
|
||
Mark Brown
|
a41016e403 | Merge branch 'for-5.2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-5.3 | ||
|
S.j. Wang
|
bfe95dfa4d |
ASoC: cs42xx8: Add reset gpio handling
Handle the reset GPIO and reset the device every time we start it. Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
|
S.j. Wang
|
ad6eecbfc0 |
ASoC: cs42xx8: Add regcache mask dirty
Add regcache_mark_dirty before regcache_sync for power
of codec may be lost at suspend, then all the register
need to be reconfigured.
Fixes:
|
||
|
Kuninori Morimoto
|
99a9f45209 |
ASoC: cs42xx8: replace codec to component
Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 1 -> .idle_bias_on = 0 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
Charles Keepax
|
03d2ec460f |
ASoC: cs42xx8: Mark chip ID as volatile and remove cache bypass
Rather than manually enabling cache bypass when reading the ID registers simply remove the default which will cause the first read to go to the hardware. The old code worked this is simply the more standard way to implement this. There is a comment included in the code that claims the chip ID register also contains the right input volume, however this is clearly not the case from the rest of the driver. Further investigation reveals exactly the same comment in the wm8962 driver, where this is the case, so this is almost certainly a copy and paste error from when the driver was created. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
|
Kuninori Morimoto
|
4d7ee73707 |
ASoC: codec duplicated callback function goes to component on cs42xx8
codec driver and component driver has duplicated callback functions, and codec side functions are just copied to component side when register timing. This was quick-hack, but no longer needed. This patch moves these functions from codec driver to component driver. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
Zidan Wang
|
1f1e60c9cd |
ASoC: cs42xx8: fix the noise in the right dac channel with mono playback
When playback mono wav with record in background, there will be some nosie in the right dac channel. It seems that the ADC data has been routed to the dac channel. The cs42888 have 8 dac channels, it's appropriate to mute the unused dac channels, and the noise will disappear. Steps to reproduce this issue: arecord -D hw:0,0 -f S16_LE -r 48000 -c 1 a.wav & aplay -Dhw:0,0 audio48k16M.wav Signed-off-by: Zidan Wang <zidan.wang@freescale.com> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
Axel Lin
|
5e4cb7b608 |
ASoC: cs42xx8: Setup of_match_table
Setup of_match_table and since cs42xx8_of_match is exported and used in cs42xx8-i2c.c, it cannot be static. Signed-off-by: Axel Lin <axel.lin@ingics.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
|
Axel Lin
|
d375d0abcd |
ASoC: cs42xx8: Move the code checking *regmap argument earlier
Slightly improve the readability by moving the code checking *regmap argument earlier. Also move the assignment of of_id close to the place testing it. Signed-off-by: Axel Lin <axel.lin@ingics.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
Lars-Peter Clausen
|
02b8c59ade |
ASoC: cs42xx8: Replace direct snd_soc_codec dapm field access
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm(). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org> |
||
Rafael J. Wysocki
|
641d334b29 |
sound / PM: Replace CONFIG_PM_RUNTIME with CONFIG_PM
After commit
|
||
Shengjiu Wang
|
689dc64385 |
ASoC: cs42xx8: Add SND_SOC_DAIFMT_DSP_A support
According to the spec, the definition of TDM and ONELINE_24 for CS42XX8_INTF_DAC and CS42XX8_INTF_ADC is wrong. correct them and enable SND_SOC_DAIFMT_DSP_A support. Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@linaro.org> |
||
|
Lars-Peter Clausen
|
afb7bb45bb |
ASoC: cs42xx8: Make of match table static
The cs42xx8_of_match table is not used outside of the driver, hence it can and should be made static. Fixes the following warning from sparse: sound/soc/codecs/cs42xx8.c:425:27: warning: symbol 'cs42xx8_of_match' was not declared. Should it be static? Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@linaro.org> |
||
|
Lars-Peter Clausen
|
5958de23ed |
ASoC: cs42xx8: Do not use rtd->codec
rtd->codec does not necessarily point to the CODEC instance for which the callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use dai->codec instead. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@linaro.org> |
||
|
Axel Lin
|
06b4b81305 |
ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value
Fix checking return value of regmap_read(). Also fix reporting the chip_id value. CS42XX8_CHIPID_CHIP_ID_MASK is 0xF0, so the chip_id value is (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4). Signed-off-by: Axel Lin <axel.lin@ingics.com> Acked-by: Paul Handrigan <paul.handrigan@cirrus.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org> |
||
Nicolin Chen
|
0c516b4ff8 |
ASoC: cs42xx8: Add codec driver support for CS42448/CS42888
This patch adds support for the Cirrus Logic CS42448/CS42888 Audio CODEC that has six/four 24-bit AD and eight 24-bit DA converters. [ CS42448/CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com> Signed-off-by: Mark Brown <broonie@linaro.org> |