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[RFC PATCH v2] ALSA: pcm: Introduce MSBITS subformat API extension
by Jaroslav Kysela 13 Sep '23

13 Sep '23

Improve granularity of format selection for linear formats by adding constants representing MAX, 20, 24 most significant bits. The MAX means the maximum number of significant bits which can the physical format hold. For 32-bit formats, MAX is related to 32 bits. For 8-bit formats, MAX is related to 8 bits etc. The drivers may use snd_pcm_hw_constraint_subformats with a simple format -> subformats table. Original proposal - https://lore.kernel.org/alsa-devel/20230811164853.1797547-1-cezary.rojewski… Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- v2: - added link to the original proposal (RFC) - reshuffle declarations in snd_pcm_hw_rule_subformats - remove nmemd from description - snd_pcm_hw_constraint_subformats - more strict handling for SNDRV_PCM_SUBFORMAT_MSBITS_MAX in fixup_unreferenced_params - fixed sparse warning (unsigned int -> snd_pcm_format_t) --- include/sound/pcm.h | 17 +++++++++ include/uapi/sound/asound.h | 7 ++-- sound/core/pcm_lib.c | 59 +++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 25 ++++++++++--- tools/include/uapi/sound/asound.h | 7 ++-- 5 files changed, 107 insertions(+), 8 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2a815373dac1..59ad45b42e03 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -217,6 +217,12 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE #endif +#define _SNDRV_PCM_SUBFMTBIT(fmt) BIT((__force int)SNDRV_PCM_SUBFORMAT_##fmt) +#define SNDRV_PCM_SUBFMTBIT_STD _SNDRV_PCM_SUBFMTBIT(STD) +#define SNDRV_PCM_SUBFMTBIT_MSBITS_MAX _SNDRV_PCM_SUBFMTBIT(MSBITS_MAX) +#define SNDRV_PCM_SUBFMTBIT_MSBITS_20 _SNDRV_PCM_SUBFMTBIT(MSBITS_20) +#define SNDRV_PCM_SUBFMTBIT_MSBITS_24 _SNDRV_PCM_SUBFMTBIT(MSBITS_24) + struct snd_pcm_file { struct snd_pcm_substream *substream; int no_compat_mmap; @@ -290,6 +296,13 @@ struct snd_pcm_hw_constraint_ranges { unsigned int mask; }; +#define SNDRV_PCM_FORMAT_CONSTRAINT_END (~0) + +struct snd_pcm_hw_constraint_subformat { + snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */ + u32 subformats; /* SNDRV_PCM_SUBFMTBIT_* */ +}; + /* * userspace-provided audio timestamp config to kernel, * structure is for internal use only and filled with dedicated unpack routine @@ -375,6 +388,7 @@ struct snd_pcm_runtime { unsigned int rate_num; unsigned int rate_den; unsigned int no_period_wakeup: 1; + unsigned int subformat_constraint: 1; /* -- SW params; see struct snd_pcm_sw_params for comments -- */ int tstamp_mode; @@ -1068,6 +1082,9 @@ int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratdens *r); +int snd_pcm_hw_constraint_subformats(struct snd_pcm_runtime *runtime, + unsigned int cond, + struct snd_pcm_hw_constraint_subformat *subformats); int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, unsigned int cond, unsigned int width, diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index f9939da41122..d5b9cfbd9cea 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -142,7 +142,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 15) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 16) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,7 +267,10 @@ typedef int __bitwise snd_pcm_format_t; typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_SUBFORMAT_STD ((__force snd_pcm_subformat_t) 0) -#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_STD +#define SNDRV_PCM_SUBFORMAT_MSBITS_MAX ((__force snd_pcm_subformat_t) 1) +#define SNDRV_PCM_SUBFORMAT_MSBITS_20 ((__force snd_pcm_subformat_t) 2) +#define SNDRV_PCM_SUBFORMAT_MSBITS_24 ((__force snd_pcm_subformat_t) 3) +#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_MSBITS_24 #define SNDRV_PCM_INFO_MMAP 0x00000001 /* hardware supports mmap */ #define SNDRV_PCM_INFO_MMAP_VALID 0x00000002 /* period data are valid during transfer */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a11cd7d6295f..cb376e428f59 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1404,6 +1404,65 @@ int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, } EXPORT_SYMBOL(snd_pcm_hw_constraint_ratdens); +static int snd_pcm_hw_rule_subformats(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + const struct snd_pcm_hw_constraint_subformat *sf; + snd_pcm_format_t k; + struct snd_mask m; + struct snd_mask *fmask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_SUBFORMAT); + bool found; + + snd_mask_none(&m); + snd_mask_set(&m, (__force unsigned)SNDRV_PCM_SUBFORMAT_STD); + pcm_for_each_format(k) { + if (!snd_mask_test(fmask, k)) + continue; + found = false; + for (sf = rule->private; sf && sf->format != SNDRV_PCM_FORMAT_CONSTRAINT_END; sf++) { + if (sf->format != k) + continue; + found = true; + m.bits[0] |= sf->subformats; + break; + } + if (!found && snd_pcm_format_linear(k)) + snd_mask_set(&m, (__force unsigned)SNDRV_PCM_SUBFORMAT_MSBITS_MAX); + } + return snd_mask_refine(mask, &m); +} + +/** + * snd_pcm_hw_constraint_subformats - add a hw constraint subformats rule + * @runtime: PCM runtime instance + * @cond: condition bits + * @subformats: array with struct snd_pcm_subformat elements + * + * This constraint will set relation between format and subformats. + * The STD and MAX subformats are handled automatically. If the driver + * does not set this constraint, only STD and MAX subformats are handled. + * + * Return: Zero if successful, or a negative error code on failure. + */ +int snd_pcm_hw_constraint_subformats(struct snd_pcm_runtime *runtime, + unsigned int cond, + struct snd_pcm_hw_constraint_subformat *subformats) +{ + int ret; + + ret = snd_pcm_hw_rule_add(runtime, cond, -1, + snd_pcm_hw_rule_subformats, + (void*) subformats, + SNDRV_PCM_HW_PARAM_SUBFORMAT, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret < 0) + return ret; + runtime->subformat_constraint = 1; + return 0; +} +EXPORT_SYMBOL(snd_pcm_hw_constraint_subformats); + static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index bd9ddf412b46..d4442b458c3c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -479,6 +479,7 @@ static int fixup_unreferenced_params(struct snd_pcm_substream *substream, { const struct snd_interval *i; const struct snd_mask *m; + struct snd_mask *m_rw; int err; if (!params->msbits) { @@ -487,6 +488,21 @@ static int fixup_unreferenced_params(struct snd_pcm_substream *substream, params->msbits = snd_interval_value(i); } + if (params->msbits) { + m = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); + if (snd_mask_single(m)) { + snd_pcm_format_t format = (__force snd_pcm_format_t)snd_mask_min(m); + if (snd_pcm_format_linear(format) && + snd_pcm_format_width(format) != params->msbits) { + m_rw = hw_param_mask(params, SNDRV_PCM_HW_PARAM_SUBFORMAT); + snd_mask_reset(m_rw, + (__force unsigned)SNDRV_PCM_SUBFORMAT_MSBITS_MAX); + if (snd_mask_empty(m_rw)) + return -EINVAL; + } + } + } + if (!params->rate_den) { i = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); if (snd_interval_single(i)) { @@ -2634,10 +2650,11 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; - err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, - PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD)); - if (err < 0) - return err; + if (!runtime->subformat_constraint) { + err = snd_pcm_hw_constraint_subformats(runtime, 0, NULL); + if (err < 0) + return err; + } err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, hw->channels_min, hw->channels_max); diff --git a/tools/include/uapi/sound/asound.h b/tools/include/uapi/sound/asound.h index f9939da41122..d5b9cfbd9cea 100644 --- a/tools/include/uapi/sound/asound.h +++ b/tools/include/uapi/sound/asound.h @@ -142,7 +142,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 15) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 16) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,7 +267,10 @@ typedef int __bitwise snd_pcm_format_t; typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_SUBFORMAT_STD ((__force snd_pcm_subformat_t) 0) -#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_STD +#define SNDRV_PCM_SUBFORMAT_MSBITS_MAX ((__force snd_pcm_subformat_t) 1) +#define SNDRV_PCM_SUBFORMAT_MSBITS_20 ((__force snd_pcm_subformat_t) 2) +#define SNDRV_PCM_SUBFORMAT_MSBITS_24 ((__force snd_pcm_subformat_t) 3) +#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_MSBITS_24 #define SNDRV_PCM_INFO_MMAP 0x00000001 /* hardware supports mmap */ #define SNDRV_PCM_INFO_MMAP_VALID 0x00000002 /* period data are valid during transfer */ -- 2.41.0

1 0

[PATCH] ALSA: usb-audio: mixer: Remove temporary string use in parse_clock_source_unit
by Peter Ujfalusi 13 Sep '23

13 Sep '23

The kctl->id.name can be directly passed to snd_usb_copy_string_desc() and if the string has been fetched the suffix can be appended with the append_ctl_name() call. The temporary name string becomes redundant and can be removed. This change will also fixes the following compiler warning/error (W=1): sound/usb/mixer.c: In function ‘parse_audio_unit’: sound/usb/mixer.c:1972:29: error: ‘ Validity’ directive output may be truncated writing 9 bytes into a region of size between 1 and 44 [-Werror=format-truncation=] 1972 | "%s Validity", name); | ^~~~~~~~~ In function ‘parse_clock_source_unit’, inlined from ‘parse_audio_unit’ at sound/usb/mixer.c:2892:10: sound/usb/mixer.c:1971:17: note: ‘snprintf’ output between 10 and 53 bytes into a destination of size 44 1971 | snprintf(kctl->id.name, sizeof(kctl->id.name), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 1972 | "%s Validity", name); | ~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors The warnings got brought to light by a recent patch upstream: commit 6d4ab2e97dcf ("extrawarn: enable format and stringop overflow warnings in W=1") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/usb/mixer.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9105ec623120..985b1aea9cdc 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1929,7 +1929,6 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, struct uac_clock_source_descriptor *hdr = _ftr; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ret; if (state->mixer->protocol != UAC_VERSION_2) @@ -1966,10 +1965,9 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, kctl->private_free = snd_usb_mixer_elem_free; ret = snd_usb_copy_string_desc(state->chip, hdr->iClockSource, - name, sizeof(name)); + kctl->id.name, sizeof(kctl->id.name)); if (ret > 0) - snprintf(kctl->id.name, sizeof(kctl->id.name), - "%s Validity", name); + append_ctl_name(kctl, " Validity"); else snprintf(kctl->id.name, sizeof(kctl->id.name), "Clock Source %d Validity", hdr->bClockID); -- 2.42.0

2 1

[PATCH V2 0/5] ASoC: codecs: Add aw87390 amplifier driver
by wangweidong.a＠awinic.com 13 Sep '23

13 Sep '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The awinic aw87390 is a new high efficiency, low noise, constant large volume, 6th Smart K audio amplifier. Add a DT schema for describing awinic aw87390 audio amplifiers. They are controlled using I2C. v1 -> v2 : Added properties in awinic,aw88395 file Remove fade-enable because the functionality is duplicated in the aw88395 driver Modify dev_err_probe usage Delete unused header files Change the i2c driver name Modify the way a function returns a value Weidong Wang (5): ASoC: dt-bindings: Adds properties to "awinic,aw88395" ASoC: dt-bindings: Add schema for "awinic,aw87390" ASoC: codecs: Add code for bin parsing compatible with aw87390 ASoC: codecs: Change the aw88261 variable name and i2c driver name ASoC: codecs: Add aw87390 amplifier driver .../bindings/sound/awinic,aw87390.yaml | 58 +++ .../bindings/sound/awinic,aw88395.yaml | 19 + sound/soc/codecs/Kconfig | 15 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw87390.c | 462 ++++++++++++++++++ sound/soc/codecs/aw87390.h | 85 ++++ sound/soc/codecs/aw88261.c | 23 +- sound/soc/codecs/aw88261.h | 2 +- sound/soc/codecs/aw88395/aw88395.c | 9 +- sound/soc/codecs/aw88395/aw88395.h | 2 +- sound/soc/codecs/aw88395/aw88395_device.c | 47 +- sound/soc/codecs/aw88395/aw88395_device.h | 6 +- sound/soc/codecs/aw88395/aw88395_lib.c | 23 +- sound/soc/codecs/aw88395/aw88395_reg.h | 1 + 14 files changed, 685 insertions(+), 69 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw87390.yaml create mode 100644 sound/soc/codecs/aw87390.c create mode 100644 sound/soc/codecs/aw87390.h base-commit: 0bb80ecc33a8fb5a682236443c1e740d5c917d1d -- 2.41.0

3 12

[RFC PATCH 00/17] ALSA/ASoC: hda: Address format selection limitations and ambiguity
by Cezary Rojewski 13 Sep '23

13 Sep '23

Patchset aims to address format selection restrictions present currently in the HDAudio library. Formats which we are concerned about are 20 and 24 valid bits per sample within 32 bit depth container. One may identify them as S20_LE and S24_LE except that those, according to comments found in include/uapi/sound/asound.h, are for LSB-aligned scenarios. HDAudio streams expect MSB-aligned data, no matter if we are speaking of HOST (SDxFMT) or LINK (PPLCxFMT) side - chapter 4.5.1 of the public HDAudio specification. In short, S20_LE and S24_LE are invalid options. Right now, given the implementation of snd_hdac_query_supported_pcm() within sound/hda/hdac_device.c, even if a codec responds with: "I support all possible formats specified within HDAudio specification", there will be no option to open a 20/32 or 24/32 stream. The kernel will force the stream to be opened using the highest available bit depth. After discussing subject initially with Jaroslav and Takashi, suggestion was made to utilize 'subformat' option to address the problem. The eye-opening discussion begun much earlier though, in 2019 [1]. Paired with PRs for alsa-utils [2] and alsa-lib [3]. Flow of changes: The very first patch adds MSBITS subformat options to allow for granular 20/32, 24/32 and 32/32 format selection. The next three make sure subformat is actually honored during runtime. Most of that code is based on format-related API. Follow up is upgrade to the hda stream-format interface - several functions are added to make the granular format selection simple in the HDAudio world. Core of the implementation is based on the existing snd_hdac_calc_stream_format(). The next ten patches are straightforward switch from one interface to another with cleanup of now-unsed function as a finishing touch. Last but not least - the avs-driver, on which the problem analyzed and debugged, is updated to no longer acknowledge S24_LE as a valid format option. Note: No testing done on other drivers yet. First results from snd_hda_intel and snd_soc_skl will be here on Monday. Will trigger SOF cycle through github PR. [1]: https://lore.kernel.org/alsa-devel/20190905053302.9262-1-pawel.harlozinski@… [2]: https://github.com/alsa-project/alsa-utils/pull/228 [3]: https://github.com/alsa-project/alsa-lib/pull/342 Cezary Rojewski (17): ALSA: pcm: Introduce MSBITS subformat interface ALSA: pcm: Honor subformat when configuring substream ALSA: hda: Honor subformat when querying PCMs ASoC: pcm: Honor subformat when configuring runtime ALSA: hda: Upgrade stream-format infrastructure ALSA: hda: Switch to new stream-format interface ALSA: hda/hdmi: Switch to new stream-format interface ALSA: hda/ca0132: Switch to new stream-format interface ASoC: codecs: hda: Switch to new stream-format interface ASoC: codecs: hdac_hda: Switch to new stream-format interface ASoC: codecs: hdac_hdmi: Switch to new stream-format interface ASoC: Intel Skylake: Switch to new stream-format interface ASoC: SOF: Intel: Switch to new stream-format interface ASoC: Intel: avs: Switch to new stream-format interface ALSA: hda: Drop snd_hdac_calc_stream_format() ASoC: Intel: avs: Unhardcode HDAudio BE DAI drivers description ASoC: Intel: avs: Kill S24_LE in HDAudio streaming include/sound/hda_codec.h | 5 +- include/sound/hdaudio.h | 12 +-- include/sound/pcm.h | 8 ++ include/sound/pcm_params.h | 2 + include/sound/soc.h | 1 + include/uapi/sound/asound.h | 5 +- sound/core/pcm_lib.c | 30 ++++++ sound/core/pcm_misc.c | 23 +++++ sound/core/pcm_native.c | 5 +- sound/hda/hdac_device.c | 153 +++++++++++++++++++++--------- sound/pci/hda/hda_codec.c | 2 + sound/pci/hda/hda_controller.c | 10 +- sound/pci/hda/patch_ca0132.c | 3 +- sound/pci/hda/patch_hdmi.c | 6 +- sound/soc/codecs/hda-dai.c | 6 +- sound/soc/codecs/hda.c | 2 + sound/soc/codecs/hdac_hda.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 10 +- sound/soc/intel/avs/loader.c | 4 +- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/pcm.c | 60 ++++++------ sound/soc/intel/avs/probes.c | 3 +- sound/soc/intel/avs/topology.c | 9 +- sound/soc/intel/skylake/skl-pcm.c | 11 ++- sound/soc/soc-pcm.c | 17 ++++ sound/soc/sof/intel/hda-dai-ops.c | 5 +- tools/include/uapi/sound/asound.h | 5 +- 27 files changed, 290 insertions(+), 117 deletions(-) -- 2.25.1

6 52

[PATCH] ALSA: hda: intel-sdw-acpi: Use u8 type for link index
by Peter Ujfalusi 13 Sep '23

13 Sep '23

Use consistently u8 for sdw link index. The id is limited to 4, u8 is adequate in size to store it. This change will also fixes the following compiler warning/error (W=1): sound/hda/intel-sdw-acpi.c: In function ‘sdw_intel_acpi_scan’: sound/hda/intel-sdw-acpi.c:34:35: error: ‘-subproperties’ directive output may be truncated writing 14 bytes into a region of size between 7 and 17 [-Werror=format-truncation=] 34 | "mipi-sdw-link-%d-subproperties", i); | ^~~~~~~~~~~~~~ In function ‘is_link_enabled’, inlined from ‘sdw_intel_scan_controller’ at sound/hda/intel-sdw-acpi.c:106:8, inlined from ‘sdw_intel_acpi_scan’ at sound/hda/intel-sdw-acpi.c:180:9: sound/hda/intel-sdw-acpi.c:33:9: note: ‘snprintf’ output between 30 and 40 bytes into a destination of size 32 33 | snprintf(name, sizeof(name), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ 34 | "mipi-sdw-link-%d-subproperties", i); | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors The warnings got brought to light by a recent patch upstream: commit 6d4ab2e97dcf ("extrawarn: enable format and stringop overflow warnings in W=1") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/hda/intel-sdw-acpi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index 5cb92f7ccbca..b57d72ea4503 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -23,7 +23,7 @@ static int ctrl_link_mask; module_param_named(sdw_link_mask, ctrl_link_mask, int, 0444); MODULE_PARM_DESC(sdw_link_mask, "Intel link mask (one bit per link)"); -static bool is_link_enabled(struct fwnode_handle *fw_node, int i) +static bool is_link_enabled(struct fwnode_handle *fw_node, u8 idx) { struct fwnode_handle *link; char name[32]; @@ -31,7 +31,7 @@ static bool is_link_enabled(struct fwnode_handle *fw_node, int i) /* Find master handle */ snprintf(name, sizeof(name), - "mipi-sdw-link-%d-subproperties", i); + "mipi-sdw-link-%hhu-subproperties", idx); link = fwnode_get_named_child_node(fw_node, name); if (!link) @@ -51,8 +51,8 @@ static int sdw_intel_scan_controller(struct sdw_intel_acpi_info *info) { struct acpi_device *adev = acpi_fetch_acpi_dev(info->handle); - int ret, i; - u8 count; + u8 count, i; + int ret; if (!adev) return -EINVAL; -- 2.42.0

3 2

[PATCH] ALSA: hda: cs35l56: Disable low-power hibernation mode
by Richard Fitzgerald 13 Sep '23

13 Sep '23

Do not allow the CS35L56 to be put into its lowest power "hibernation" mode. This only affects I2C because "hibernation" is already disabled on SPI. Recent firmwares need a different wake-up sequence. Until that sequence has been specified, the chip "hibernation" mode must be disabled otherwise it can intermittently fail to wake. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda_i2c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 83e4acdd89ac..757a4d193e0f 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -21,7 +21,6 @@ static int cs35l56_hda_i2c_probe(struct i2c_client *clt) return -ENOMEM; cs35l56->base.dev = &clt->dev; - cs35l56->base.can_hibernate = true; cs35l56->base.regmap = devm_regmap_init_i2c(clt, &cs35l56_regmap_i2c); if (IS_ERR(cs35l56->base.regmap)) { ret = PTR_ERR(cs35l56->base.regmap); -- 2.30.2

2 1

[PATCH] ASoC: dwc: Add Single DMA mode support
by Myunguk Kim 13 Sep '23

13 Sep '23

There is a SoC between dwc and DMA block (ie. PL330) that does not have a burst signal and supports only single. So write not-support-burst property on dts, it support single DMA mode. Signed-off-by: Myunguk Kim <mwkim(a)gaonchips.com> --- .../bindings/sound/snps,designware-i2s.yaml | 3 +++ include/sound/designware_i2s.h | 1 + sound/soc/dwc/dwc-i2s.c | 12 ++++++++++-- 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml index a48d040b0a4f..43a46ba2a70c 100644 --- a/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml @@ -86,6 +86,9 @@ properties: The phandle to System Register Controller syscon node and the I2S-rx(ADC) enabled control offset and mask of SYS_SYSCONSAIF__SYSCFG register. + no-burst: + description: Use single transaction only between DesignWare I2S and DMA IP. + allOf: - $ref: dai-common.yaml# - if: diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index f6803205a9fb..f0207f21f09d 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -36,6 +36,7 @@ struct i2s_platform_data { #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) #define DW_I2S_QUIRK_16BIT_IDX_OVERRIDE (1 << 2) + #define DW_I2S_QUIRK_NO_BURST (1 << 3) unsigned int quirks; unsigned int i2s_reg_comp1; unsigned int i2s_reg_comp2; diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 5ab1b3eb2d28..71ff894035a4 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -713,7 +713,10 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; dev->play_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2]) >> 8; - dev->play_dma_data.dt.maxburst = 16; + if (dev->quirks & DW_I2S_QUIRK_NO_BURST) + dev->play_dma_data.dt.maxburst = 1; + else + dev->play_dma_data.dt.maxburst = 16; } if (COMP1_RX_ENABLED(comp1)) { idx2 = COMP2_RX_WORDSIZE_0(comp2); @@ -722,7 +725,10 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; dev->capture_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2] >> 8); - dev->capture_dma_data.dt.maxburst = 16; + if (dev->quirks & DW_I2S_QUIRK_NO_BURST) + dev->capture_dma_data.dt.maxburst = 1; + else + dev->capture_dma_data.dt.maxburst = 16; } return 0; @@ -979,6 +985,8 @@ static int dw_i2s_probe(struct platform_device *pdev) ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata); } else { clk_id = "i2sclk"; + if (of_get_property(pdev->dev.of_node, "no-burst", NULL)) + dev->quirks |= DW_I2S_QUIRK_NO_BURST; ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res); } if (ret < 0) -- 2.34.1

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AIF2 ADC Stereo Capture Route is uninitialised in alsa-ucm-conf
by GitHub issues - edited 13 Sep '23

13 Sep '23

alsa-project/alsa-ucm-conf issue #351 was edited from broukema: DESCRIPTION: `AIF2 ADC Stereo Capture Route` is not initialised in `alsa-ucm-conf`. This implies that if its setting is significant, any accidental change by the user (using any of the various `alsa` tools) to a wrong setting will not be corrected by any of the three `PinePhone/*.conf` files, leading, in particular, to audio errors during a phone call. This was discovered by testing variations on the Mobian version of the PinePhone alsa-ucm-conf files. CODING ARGUMENT: All parameters that can significantly affect the audio system should be either initialised or overridden in configuration files that the user expects to necessarily work, such as those for phone calls. In `alsa-ucm-conf` commit b68aa52 (one beyond v1.2.10): ```` $ grep -ri "AIF2 ADC Stereo Capture" . $ ```` shows that `AIF2 ADC Stereo Capture Route` is not set for any device at all (PinePhone or other). HARDWARE ARGUMENT: For the PinePhone (PP), Megi recommends in https://xnux.eu/devices/feature/audio-pp.html that `AIF2 ADC Stereo Capture Route` is set to `Mix Mono`. TESTING ARGUMENT: Experimentation with different stored alsa settings (from ```` sudo alsactl store -f - > YYYYMMDD_example_alsa_settings ```` ) showed that settings files whose only substantial difference was in the setting of `AIF2 ADC Stereo Capture Route` had the effect that audio from the PP was heard on the remote phone when the value was `Mix Mono` and was not heard with the value `Stereo`. Testing was done with the equivalent of https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… with the settings file https://codeberg.org/boud/pinephone_hacks/src/commit/7c7b6660a4446a24747f2c… that has the `Mix Mono` value selected. RECOMMENDATIONS: * `AIF2 ADC Stereo Capture Route` should be initialised or set to *something* in one of the .conf files (in the PP case, most likely in `VoiceCall.conf`) to prevent users who experiment and have difficulty remembering which of the 50 PP alsa parameters should be set in which way from accidentally sabotaging their otherwise working audio calls; * Based on Megi's 2020 page and my testing, it seems that the correct value should be `Mix Mono`. COMMENT: While setting all 50 of the alsa settings ```` $ sudo alsactl store -f - |grep "name " |wc 50 258 1788 ```` in the .conf files is most likely unnecessary, there could be others that are necessary too. Ideally, someone who really understands how the PP audio system works (including Megi's two diagrams at https://xnux.eu/devices/feature/audio-pp.html) should be able to make these recommendations. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/351 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: dwc: fix typo in comment
by Myunguk Kim 13 Sep '23

13 Sep '23

This fixes typo in comment Signed-off-by: Myunguk Kim <mwkim(a)gaonchips.com> --- sound/soc/dwc/dwc-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 71ff894035a4..45e55e550924 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -138,7 +138,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) irq_valid = true; } - /* Error Handling: TX */ + /* Error Handling: RX */ if (isr[i] & ISR_RXFO) { dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i); irq_valid = true; -- 2.34.1

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[PATCH] ASoC: imx-rpmsg: Set ignore_pmdown_time for dai_link
by Chancel Liu 13 Sep '23

13 Sep '23

i.MX rpmsg sound cards work on codec slave mode. MCLK will be disabled by CPU DAI driver in hw_free(). Some codec requires MCLK present at power up/down sequence. So need to set ignore_pmdown_time to power down codec immediately before MCLK is turned off. Take WM8962 as an example, if MCLK is disabled before DAPM power down playback stream, FIFO error will arise in WM8962 which will have bad impact on playback next. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/imx-rpmsg.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 3c7b95db2eac..d2ecd5e821b1 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -89,6 +89,13 @@ static int imx_rpmsg_probe(struct platform_device *pdev) SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + /* i.MX rpmsg sound cards work on codec slave mode. MCLK will be + * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK + * present at power up/down sequence. So need to set ignore_pmdown_time + * to power down codec immediately before MCLK is turned off. + */ + data->dai.ignore_pmdown_time = 1; + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { -- 2.25.1

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