- 12 Mar, 2024 1 commit
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Luca Ceresoli authored
The sample rates set by the rockchip_i2s_tdm driver in master mode are inaccurate up to 5% in several cases, due to the driver logic to configure clocks and a nasty interaction with the Common Clock Framework. To understand what happens, here is the relevant section of the clock tree (slightly simplified), along with the names used in the driver: vpll0 _OR_ vpll1 "mclk_root" clk_i2s2_8ch_tx_src "mclk_parent" clk_i2s2_8ch_tx_mux clk_i2s2_8ch_tx "mclk" or "mclk_tx" This is what happens when playing back e.g. at 192 kHz using audio-graph-card (when recording the same applies, only s/tx/rx/): 0. at probe, rockchip_i2s_tdm_set_sysclk() stores the passed frequency in i2s_tdm->mclk_tx_freq (*) which is 50176000, and that is never modified afterwards 1. when playback is started, rockchip_i2s_tdm_hw_params() is called and does the following two calls 2. rockchip_i2s_tdm_calibrate_mclk(): 2a. selects mclk_root0 (vpll0) as a parent for mclk_parent (mclk_tx_src), which is OK because the vpll0 rate is a good for 192000 (and sumbultiple) rates 2b. sets the mclk_root frequency based on ppm calibration computations 2c. sets mclk_tx_src to 49152000 (= 256 * 192000), which is also OK as it is a multiple of the required bit clock 3. rockchip_i2s_tdm_set_mclk() 3a. calls clk_set_rate() to set the rate of mclk_tx (clk_i2s2_8ch_tx) to the value of i2s_tdm->mclk_tx_freq (*), i.e. 50176000 which is not a multiple of the sampling frequency -- this is not OK 3a1. clk_set_rate() reacts by reparenting clk_i2s2_8ch_tx_src to vpll1 -- this is not OK because the default vpll1 rate can be divided to get 44.1 kHz and related rates, not 192 kHz The result is that the driver does a lot of ad-hoc decisions about clocks and ends up in using the wrong parent at an unoptimal rate. Step 0 is one part of the problem: unless the card driver calls set_sysclk at each stream start, whatever rate is set in mclk_tx_freq during boot will be taken and used until reboot. Moreover the driver does not care if its value is not a multiple of any audio frequency. Another part of the problem is that the whole reparenting and clock rate setting logic is conflicting with the CCF algorithms to achieve largely the same goal: selecting the best parent and setting the closest clock rate. And it turns out that only calling once clk_set_rate() on clk_i2s2_8ch_tx picks the correct vpll and sets the correct rate. The fix is based on removing the custom logic in the driver to select the parent and set the various clocks, and just let the Clock Framework do it all. As a side effect, the set_sysclk() op becomes useless because we now let the CCF compute the appropriate value for the sampling rate. It also implies that the whole calibration logic is now dead code and so it is removed along with the "PCM Clock Compensation in PPM" kcontrol, which has always been broken anyway. The handling of the 4 optional clocks also becomes dead code and is removed. The actual rates have been tested playing 30 seconds of audio at various sampling rates before and after this change using sox: time play -r <sample_rate> -n synth 30 sine 950 gain -3 The time reported in the table below is the 'real' value reported by the 'time' command in the above command line. rate before after --------- ------ ------ 8000 Hz 30.60s 30.63s 11025 Hz 30.45s 30.51s 16000 Hz 30.47s 30.50s 22050 Hz 30.78s 30.41s 32000 Hz 31.02s 30.43s 44100 Hz 30.78s 30.41s 48000 Hz 29.81s 30.45s 88200 Hz 30.78s 30.41s 96000 Hz 29.79s 30.42s 176400 Hz 27.40s 30.41s 192000 Hz 29.79s 30.42s While the tests are running the clock tree confirms that: * without the patch, vpll1 is always used and clk_i2s2_8ch_tx always produces 50176000 Hz, which cannot be divided for most audio rates except the slowest ones, generating inaccurate rates * with the patch: - for 192000 Hz vpll0 is used - for 176400 Hz vpll1 is used - clk_i2s2_8ch_tx always produces (256 * <rate>) Hz Tested on the RK3308 using the internal audio codec. Fixes: 081068fd ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by:Luca Ceresoli <luca.ceresoli@bootlin.com> Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-1-312acdbe628f@bootlin.comSigned-off-by:
Mark Brown <broonie@kernel.org>
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- 11 Mar, 2024 2 commits
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M Cooley authored
The ASUS M7600RE (Vivobook Pro 16X OLED) needs a quirks-table entry for the internal microphone to function properly. Signed-off-by:
Mitch Cooley <m.cooley.198@gmail.com> Link: https://msgid.link/r/CALijGznExWW4fujNWwMzmn_K=wo96sGzV_2VkT7NjvEUdkg7Gw@mail.gmail.comSigned-off-by:
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Uwe Kleine-König authored
Using __exit for the remove function results in the remove callback being discarded with SND_SOC_TLV320ADC3XXX=y. When such a device gets unbound (e.g. using sysfs or hotplug), the driver is just removed without the cleanup being performed. This results in resource leaks. Fix it by compiling in the remove callback unconditionally. This also fixes a W=1 modpost warning: WARNING: modpost: sound/soc/codecs/snd-soc-tlv320adc3xxx: section mismatch in reference: adc3xxx_i2c_driver+0x10 (section: .data) -> adc3xxx_i2c_remove (section: .exit.text) (which only happens with SND_SOC_TLV320ADC3XXX=m). Fixes: e9a3b57e ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by:
Uwe Kleine-König <u.kleine-koenig@pengutronix.de> Reviewed-by:
Geert Uytterhoeven <geert@linux-m68k.org> Link: https://msgid.link/r/20240310143852.397212-2-u.kleine-koenig@pengutronix.deSigned-off-by:
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- 07 Mar, 2024 6 commits
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Andy Shevchenko authored
PM constants for PCI devices are defined with bitwise annotation. When used as is, sparse complains about that: .../catpt/dsp.c:390:9: warning: restricted pci_power_t degrades to integer .../catpt/dsp.c:414:9: warning: restricted pci_power_t degrades to integer Force them to be u32 in the driver. Signed-off-by:
Andy Shevchenko <andriy.shevchenko@linux.intel.com> Acked-by:
Cezary Rojewski <cezary.rojewski@intel.com> Link: https://msgid.link/r/20240307163734.3852754-1-andriy.shevchenko@linux.intel.comSigned-off-by:
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Mark Brown authored
Merge series from Zhang Yi <zhangyi@everest-semi.com>: We found that using 0x45 as the default value for interrupt-clk would cause a headset detection error.So we took 0x00 as the default value for interrupt-clk and passed the test. We removed mic1-src and mic2-src, which were not used.
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Richard Fitzgerald authored
The kerneldoc for struct cs_dsp refers to a fw_file_name member but there's no such member. Signed-off-by:
Richard Fitzgerald <rf@opensource.cirrus.com> Link: https://msgid.link/r/20240307105516.40250-1-rf@opensource.cirrus.comSigned-off-by:
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Richard Fitzgerald authored
Initialize debugfs_root to -ENODEV so that if the client never sets a valid debugfs root the debugfs files will not be created. A NULL pointer passed to any of the debugfs_create_*() functions means "create in the root of debugfs". It doesn't mean "ignore". Signed-off-by:
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Zhang Yi authored
Removed mic1-src and mic2-src. and changed default value of interrupt-clk Signed-off-by:
Zhang Yi <zhangyi@everest-semi.com> Link: https://msgid.link/r/20240307051222.24010-3-zhangyi@everest-semi.comSigned-off-by:
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Zhang Yi authored
We don't use mic1_src and mic2_src.so we delete these two members. We changed the default value of interrupt-clk for headphone detection Signed-off-by:
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- 06 Mar, 2024 2 commits
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Luca Ceresoli authored
Add the event value to the snd_soc_dapm_start and snd_soc_dapm_done trace events to make them more informative. Trace before: aplay-229 [000] 250.140309: snd_soc_dapm_start: card=vscn-2046 aplay-229 [000] 250.167531: snd_soc_dapm_done: card=vscn-2046 aplay-229 [000] 251.169588: snd_soc_dapm_start: card=vscn-2046 aplay-229 [000] 251.195245: snd_soc_dapm_done: card=vscn-2046 Trace after: aplay-214 [000] 693.290612: snd_soc_dapm_start: card=vscn-2046 event=1 aplay-214 [000] 693.315508: snd_soc_dapm_done: card=vscn-2046 event=1 aplay-214 [000] 694.537349: snd_soc_dapm_start: card=vscn-2046 event=2 aplay-214 [000] 694.563241: snd_soc_dapm_done: card=vscn-2046 event=2 Signed-off-by: -
Luca Ceresoli authored
The snd_soc_bias_level_start and snd_soc_bias_level_done trace events currently look like: aplay-229 [000] 1250.140778: snd_soc_bias_level_start: card=vscn-2046 val=1 aplay-229 [000] 1250.140784: snd_soc_bias_level_done: card=vscn-2046 val=1 aplay-229 [000] 1250.140786: snd_soc_bias_level_start: card=vscn-2046 val=2 aplay-229 [000] 1250.140788: snd_soc_bias_level_done: card=vscn-2046 val=2 kworker/u8:1-21 [000] 1250.140871: snd_soc_bias_level_start: card=vscn-2046 val=1 kworker/u8:0-11 [000] 1250.140951: snd_soc_bias_level_start: card=vscn-2046 val=1 kworker/u8:0-11 [000] 1250.140956: snd_soc_bias_level_done: card=vscn-2046 val=1 kworker/u8:0-11 [000] 1250.140959: snd_soc_bias_level_start: card=vscn-2046 val=2 kworker/u8:0-11 [000] 1250.140961: snd_soc_bias_level_done: card=vscn-2046 val=2 kworker/u8:1-21 [000] 1250.167219: snd_soc_bias_level_done: card=vscn-2046 val=1 kworker/u8:1-21 [000] 1250.167222: snd_soc_bias_level_start: card=vscn-2046 val=2 kworker/u8:1-21 [000] 1250.167232: snd_soc_bias_level_done: card=vscn-2046 val=2 kworker/u8:0-11 [000] 1250.167440: snd_soc_bias_level_start: card=vscn-2046 val=3 kworker/u8:0-11 [000] 1250.167444: snd_soc_bias_level_done: card=vscn-2046 val=3 kworker/u8:1-21 [000] 1250.167497: snd_soc_bias_level_start: card=vscn-2046 val=3 kworker/u8:1-21 [000] 1250.167506: snd_soc_bias_level_done: card=vscn-2046 val=3 There are clearly multiple calls, one per component, but they cannot be discriminated from each other. Change the ftrace events to also print the component name, to make it clear which part of the code is involved. This requires changing the passed value from a struct snd_soc_card, where the DAPM context is not kwown, to a struct snd_soc_dapm_context where it is obviously known but the a card pointer is also available. With this change, the resulting trace becomes: aplay-247 [000] 1436.357332: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=1 aplay-247 [000] 1436.357338: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=1 aplay-247 [000] 1436.357340: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=2 aplay-247 [000] 1436.357343: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=2 kworker/u8:4-215 [000] 1436.357437: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=1 kworker/u8:5-231 [000] 1436.357518: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=1 kworker/u8:5-231 [000] 1436.357523: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=1 kworker/u8:5-231 [000] 1436.357526: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=2 kworker/u8:5-231 [000] 1436.357528: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=2 kworker/u8:4-215 [000] 1436.383217: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=1 kworker/u8:4-215 [000] 1436.383221: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=2 kworker/u8:4-215 [000] 1436.383231: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=2 kworker/u8:5-231 [000] 1436.383468: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=3 kworker/u8:5-231 [000] 1436.383472: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=3 kworker/u8:4-215 [000] 1436.383503: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=3 kworker/u8:4-215 [000] 1436.383513: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=3 Signed-off-by:
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- 05 Mar, 2024 2 commits
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Javier García authored
Convert the RT1015 Mono Class D Audio Amplifier to DT schema. Signed-off-by:
Javier García <javier.gar.tab@gmail.com> Reviewed-by:
Rob Herring <robh@kernel.org> Link: https://msgid.link/r/20240304142315.14522-1-javier.gar.tab@gmail.comSigned-off-by:
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Andy Shevchenko authored
intel-mid.h is providing some core parts of the South Complex PM, which are usually are not used by individual drivers. In particular, this driver doesn't use it, so simply remove the unused header. Signed-off-by:
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- 04 Mar, 2024 2 commits
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Chancel Liu authored
ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name for dlc (snd_soc_dai_link_component). In this function call dlc->dai_name is parsed via snd_soc_dai_name_get() (B). (A) int snd_soc_get_dlc(...) { ... (B) dlc->dai_name = snd_soc_dai_name_get(dai); ... } (B) has a priority to return dai->name as dlc->dai_name. In most cases card can probe successfully. However it has an issue that ASoC tries to rebind card. Here is a simplified flow for example: | a) Card probes successfully at first | b) One of the component bound to this card is removed for some | reason the component->dev is released | c) That component is re-registered v d) ASoC calls snd_soc_try_rebind_card() a) points dlc->dai_name to dai->name. b) releases all resource of the old DAI. c) creates new DAI structure. In result d) can not use dlc->dai_name to add new created DAI. So it's reasonable that prefer to return dai->driver->name in snd_soc_dai_name_get() because dai->driver is a pre-defined global variable. Also update snd_soc_is_matching_dai() for alignment. Signed-off-by:Chancel Liu <chancel.liu@nxp.com> Link: https://msgid.link/r/20240304072128.2845432-1-chancel.liu@nxp.comSigned-off-by:
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Richard Fitzgerald authored
Add a KUnit test for the cs-amp-lib library. This has test cases for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs(). A KUNIT_STATIC_STUB_REDIRECT() has been added to cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the KUnit test can redirect these to test harness functions. Much of the testing involves invoking the same function with different parameters, i.e. the number of amps and the amp index within the array. This uses parameterization rather than looping. The idea is to avoid looping over configurations within one test case as that has a higher chance of having a bug that doesn't actually test all the expected cases. Having the test run exactly one configuration, and then tear-down, is less prone to accidentally skipped configurations. Signed-off-by:
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- 27 Feb, 2024 2 commits
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Jerome Brunet authored
Use FIELD_GET() and FIELD_PREP() helpers instead of doing it manually. Signed-off-by:
Jerome Brunet <jbrunet@baylibre.com> Link: https://msgid.link/r/20240227150826.573581-1-jbrunet@baylibre.comSigned-off-by:
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Richard Fitzgerald authored
Cast u8 values to u32 when using them to build a 32-bit unsigned value that is then stored in a u64. This avoids the possibility of a bad sign extension where the u8 is implicitly extended to an int, thus changing it from an unsigned to a signed value. Whether this is a real problem is debatable, but it does no harm to ensure that the u8 are cast to a suitable type for shifting. Signed-off-by:
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- 26 Feb, 2024 9 commits
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Mark Brown authored
Merge series from Jerome Brunet <jbrunet@baylibre.com>: This are various fixes and clean up gathered while working on Amlogic audio support. These help better handle higher and unusual clock configuration for TDM, SPDIF or PDM.
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Mark Brown authored
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>: The patchset may not cover all codecs found in the codecs/ directory - noticed a possible improvement and grepped for similar pattern across C files found in the directory. Those addressed here seem pretty straightforward. Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed.
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Jerome Brunet authored
The rate of the stream does not matter for the fifos of the axg family. Fifos will just push or pull data to/from the DDR according to consumption or production of the downstream element, which is the DPCM backend. Drop the rate list and allow continuous rates. The lower and upper rate are set according what is known to work with the different backends This allows the PDM input backend to also use continuous rates. Signed-off-by:
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Jerome Brunet authored
Use maximum width between 2 edges to setup spdifin thresholds and detect the input sample rate. This comes from Amlogic SDK and seems to be marginally more reliable than minimum width. This is done to align with a future eARC support. No issue was reported with minimum width so far, this is considered to be an update so no Fixes tag is set. Signed-off-by:
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Jerome Brunet authored
ASoC stopped using CBS_CFS and CBM_CFM a few years ago but the traces in the amlogic tdm interface driver did not follow. Update this to match the new format names Signed-off-by:
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Jerome Brunet authored
According to Amlogic datasheets for the SoCs supported by this driver, the maximum bit clock rate is 100MHz. The tdm interface allows the rates listed by the DAI driver, regardless of the number slots or their width. However, these will impact the bit clock rate. Hitting the 100MHz limit is very unlikely for most use cases but it is possible. For example with 32 slots / 32 bits wide, the maximum rate is no longer 384kHz but ~96kHz. Add the constraint accordingly if the component is not already active. If it is active, the rate is already constrained by the first stream rate. Fixes: d60e4f1e ("ASoC: meson: add tdm interface driver") Signed-off-by:
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Jerome Brunet authored
By default, when mclk-fs is not provided, the tdm-interface driver requests an MCLK that is 4x the bit clock, SCLK. However there is no justification for this: * If the codec needs MCLK for its operation, mclk-fs is expected to be set according to the codec requirements. * If the codec does not need MCLK the minimum is 2 * SCLK, because this is minimum the divider between SCLK and MCLK can do. Multiplying by 4 may cause problems because the PLL limit may be reached sooner than it should, so use 2x instead. Fixes: d60e4f1e ("ASoC: meson: add tdm interface driver") Signed-off-by:
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Krzysztof Kozlowski authored
Starting with Qualcomm SM8350 SoC, so Low Power Audio SubSystem (LPASS) block version v9.2, the register responsible for TX SMIC MUXn muxes is different. In earlier LPASS versions this mux had bit fields for analogue (ADCn) and digital (SWR_DMICn) MICs. Choice of ADCn was selecting the analogue path in CDC_TX_TOP_CSR_SWR_DMICn_CTL register. With LPASS v9.2 and newer, the bit fields are integrated into just SWR_MICn and there is no distinction for analogue or digital MIC in the register. Fix support for LPASS v9.2+: 1. Add new set of widgets and audio routes for LPASS v9.2. 2. Do not choose analogue or digital in CDC_TX_TOP_CSR_SWR_DMICn_CTL based on value of the mux. 3. Replace all the input widgets (TX SWR_ADCn, TX SWR_DMICn) with TX SWR_INPUTn ones. The change is not backwards compatible with older DTBs and existing mixer settings, therefore it does not change handling of older platforms with working micrphones (SC8280xp) but only the ones with issues (SM8450, SM8550) which need the fix. Signed-off-by:
Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org> Link: https://msgid.link/r/20240226115925.53953-3-krzysztof.kozlowski@linaro.orgSigned-off-by:
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Krzysztof Kozlowski authored
TX macro codec differs slightly between different Qualcomm Low Power Audio SubSystem (LPASS) block versions. In LPASS version 9.2 the register responsible for TX SMIC MUXn muxes is different, thus to properly support it, the driver needs to register different widgets per different LPASS version. Prepare for supporting this register difference by refactoring existing code: 1. Move few widgets (TX SMIC MUXn, TX SWR_ADCn, TX SWR_DMICn) out of common 'tx_macro_dapm_widgets[]' array to a new per-variant specific array 'tx_macro_dapm_widgets_v9[]'. 2. Move also related audio routes into new array. 3. Store pointers to these variant-specific arrays in new variant-data structure 'tx_macro_data'. 4. Add variant-specific widgets and routes in component probe, instead of driver probe. The change should have no real impact, except re-shuffling code and registering some widgets and audio routes in component probe, instead of driver probe. Signed-off-by:
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- 24 Feb, 2024 1 commit
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Mark Brown authored
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>: Factory calibration of the speakers stores the calibration information into an EFI variable. This set of patches adds support for applying speaker calibration data from that EFI variable. The HDA patch (#5) depends on the ASoC patches #2 and #3
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- 23 Feb, 2024 7 commits
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Varshini Rajendran authored
Add sam9x7 compatible in the DT documentation. Signed-off-by:
Varshini Rajendran <varshini.rajendran@microchip.com> Acked-by:
Conor Dooley <conor.dooley@microchip.com> Link: https://lore.kernel.org/r/20240223172657.672439-1-varshini.rajendran@microchip.comSigned-off-by:
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Varshini Rajendran authored
Add sam9x7 compatible to DT bindings documentation. Signed-off-by:
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Richard Fitzgerald authored
If there are factory calibration settings in EFI, extract the settings and write them to the firmware calibration controls. Signed-off-by:
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Richard Fitzgerald authored
If there are factory calibration settings in EFI, extract the settings and write them to the firmware calibration controls. This must be done after any firmware or coefficients have been downloaded to the amp. Signed-off-by:
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Richard Fitzgerald authored
Adds some helper functions and data for applying amp calibration. 1. cs35l56_read_silicon_uid() to get the silicon ID that is used to search for the correct calibration data entry. 2. Add the registers for the silicon ID to the readable registers. 3. cs35l56_get_calibration() wrapper around cs_amp_get_efi_calibration_data() 4. cs35l56_calibration_controls() table of the firmware controls for calibration data. 5. Added members to struct cs35l56_base to store the calibration data. Signed-off-by:
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Richard Fitzgerald authored
Create a new library for code that is used by multiple Cirrus Logic amps. This initially implements extracting amp calibration data from EFI and writing it to firmware controls. During factory calibration of built-in speakers the firmware calibration constants are stored in an EFI file. The file contains an array of calibration constants for each of the speakers. cs_amp_get_calibration_data() searches for an entry matching the requested UID stamp, otherwise by array index. If the data is found in EFI the constants for that speaker are copied back to the caller. If EFI is not enabled, the cs_amp_get_calibration_data() implementation will compile to simply return -ENOENT and the linker can drop the code. The code to write calibration controls uses cs_dsp. Building of cs_dsp is not forced. Instead, the code will compile away the calls to cs_dsp if cs_dsp is not reachable. This strategy of conditional code allows cs-amp-lib to be shared by multiple drivers without forcing inclusion of other modules that might be unnecessary. The calls to efi.get_variable() and cs_dsp are in small wrapper functions. This is so that a KUNIT_STATIC_STUB_REDIRECT can be added in a future patch to redirect these calls to replacement functions for KUnit testing. Signed-off-by:
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Richard Fitzgerald authored
Separate the functionality of wm_adsp_event() into two exported functions wm_adsp_start() and wm_adsp_stop(). This allows the codec driver to start and stop the DSP outside of a DAPM widget. Signed-off-by:
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- 22 Feb, 2024 3 commits
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Kuninori Morimoto authored
snd_soc_is_matching_dai() checks DAI name, which is paired function with snd_soc_dai_name_get(). It checks dlc->dai_name and dai->name (A) or dai->driver_name (B) or dai->component->name (C) static int snd_soc_is_matching_dai(...) { ... if (strcmp(dlc->dai_name, dai->name) == 0) ~~~~~~~~~~~~~ ^^^^^^^^^(A) if (... strcmp(dai->driver->name, dlc->dai_name) == 0) (B)^^^^^^^^^^^^^^^^ ~~~~~~~~~~~~~ if (... strcmp(dlc->dai_name, dai->component->name) == 0) ~~~~~~~~~~~~~ ^^^^^^^^^^^^^^^^^^(C) ... } But (B) part order is different with (A) and (C) (= ^^^^ and ~~~~). This is not a big deal, but confusable to read. Fixup it. Signed-off-by:Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://msgid.link/r/87wmqxjbcg.wl-kuninori.morimoto.gx@renesas.comSigned-off-by:
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Vijendar Mukunda authored
Modify ACP63_REG_END macro to access all ACP registers. Signed-off-by:
Vijendar Mukunda <Vijendar.Mukunda@amd.com> Link: https://msgid.link/r/20240222102656.631144-2-Vijendar.Mukunda@amd.comSigned-off-by:
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Vijendar Mukunda authored
To align with AMD SoundWire manager driver license, update license as GPL-2.0-only for Pink Sardine ACP PCI driver and corresponding child drivers. Signed-off-by:
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- 21 Feb, 2024 3 commits
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Cezary Rojewski authored
Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by:
Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Signed-off-by:
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Cezary Rojewski authored
Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. rt5640_set_dai_sysclk() is an example of that - clk_set_rate() is not guarded by IS_ERR(). By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by:
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Cezary Rojewski authored
Most of clk_xxx() functions do check if provided clk-pointer is non-NULL. These do not check if the pointer is an error-pointer. Providing such to a clk_xxx() results in a panic. By utilizing _optional() variant of devm_clk_get() the driver code is both simplified and more robust. There is no need to remember about IS_ERR(clk) checks each time mclk is accessed. Reviewed-by:
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