September 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: tas2781: Fix ROG ALLY X audio
by Jonathan LoBue 11 Oct '24

11 Oct '24

This patch enables the TI TAS2781 amplifier SoC for the ASUS ROG ALLY X. This is a design change from the original ASUS ROG ALLY, creating the need for this patch. All other Realtek Codec settings seem to be re-used from the original ROG ALLY design (on the ROG ALLY X). This patch maintains the previous settings for the Realtek codec portion, but enables the I2C binding for the TI TAS2781 amplifier (instead of the Cirrus CS35L41 amp used on the original ASUS ROG ALLY). One other requirement must be met for audio to work on the ASUS ROG ALLY X. A proper firmware file in the correct location with a proper symlink. We had reached out to TI engineers and confirmed that the firmware found in the Windows' driver package has a GPL license. Bazzite Github is hosting this firmware file for now until proper linux-firmware upstreaming can occur. https://github.com/ublue-os/bazzite This firmware file should be placed in /usr/lib/firmware/ti/tas2781/TAS2XXX1EB3.bin with a symlink to it from /usr/lib/firmware/TAS2XXX1EB3.bin Co-developed by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Signed-off-by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Co-developed by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Antheas Kapenekakis <lkml(a)antheas.dev> Tested-by: Richard Alvarez <alvarez.richard(a)gmail.com> Tested-by: Miles Montierth <cyber_dopamine(a)intheblackmedia.com> Signed-off-by: Jonathan LoBue <jlobue10(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 480e82df7a4c..9bc39c2ee6b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7410,6 +7410,7 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_X, ALC294_FIXUP_ASUS_ALLY_PINS, ALC294_FIXUP_ASUS_ALLY_VERBS, ALC294_FIXUP_ASUS_ALLY_SPEAKER, @@ -8877,6 +8878,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS }, + [ALC294_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, [ALC294_FIXUP_ASUS_ALLY_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10313,6 +10320,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.46.0

4 5

Add stacked and parallel memories support in spi-nor
by Mahapatra, Amit Kumar 10 Oct '24

10 Oct '24

Hello Everyone, As https://lore.kernel.org/all/20231125092137.2948-8-amit.kumar-mahapatra@amd.… thread discussing the addition of stacked and parallel mode support in SPI-NOR has become too long and difficult to follow. Therefore, I'm starting a new email thread to focus on the DT binding changes and the overall implementation architecture for adding stacked and parallel (multi-CS) support in the Kernel. Please share your input on this approach. Background: The AMD QSPI controller supports two advanced connection modes(Stacked and Dual Parallel) which allow the controller to treat two different flashes as one storage. Stacked: Flashes share the same SPI bus, but different CS line, controller driver asserts the CS of the flash to which it needs to communicate. Stacked mode is a software abstraction rather than a controller feature or capability. At any given time, the controller communicates with one of the two connected flash devices, as determined by the requested address and data length. If an operation starts on one flash and ends on the other, the core needs to split it into two separate operations and adjust the data length accordingly. Dual Parallel: Both the flashes have their separate SPI bus, CS of both the flashes will be asserted/de-asserted at the same time. In this mode data will be split across both the flashes by enabling the STRIPE setting in the controller. Parallel mode is a controller feature where if the STRIPE bit is set then the controller internally handles the data split during data write to the flashes and while reading data from the flash the controller internally merges data from both the flashes before writing to the controller FIFO. If STRIPE is not enabled, then same data will be sent to both the devices. In parallel mode both the flashes should be identical. For more information on the modes please feel free to go through the controller flash interface below https://docs.amd.com/r/en-US/am011-versal-acap-trm/QSPI-Flash-Device-Interf… Mirochip QSPI controller also supports "Dual Parallel 8-bit IO mode", but they call it "Twin Quad Mode". https://ww1.microchip.com/downloads/aemDocuments/documents/MPU32/ProductDoc… Initially in [1] [2] [3] Miquel had tried to extend MTD-CONCAT driver to support Stacked mode, but the bindings were not accepted. [1] https://lore.kernel.org/all/20191113171505.26128-4-miquel.raynal@bootlin.co… [2] https://lore.kernel.org/all/20191127105522.31445-5-miquel.raynal@bootlin.co… [3]https://lore.kernel.org/all/20211112152411.818321-1-miquel.raynal@bootlin… So, the MTD-CONCAT approach was dropped and the DT bindings that got accepted [4] [5] [6] describe the two flash devices as being one. [4] https://github.com/torvalds/linux/commit/f89504300e94524d5d5846ff8b728592ac… [5] https://github.com/torvalds/linux/commit/eba5368503b4291db7819512600fa014ea… [6] https://github.com/torvalds/linux/commit/e2edd1b64f1c79e8abda365149ed62a2a9… SPI core changes to support the above bindings were added through the following commit: https://github.com/torvalds/linux/commit/4d8ff6b0991d5e86b17b235fc46ec62e91… While adding the support in SPI-NOR we received some inputs/suggestions from Tudor, based on that I am planning to add a new layer between the SPI-NOR and MTD layers to support stacked and parallel configurations. This new layer will be part of the spi-nor and located in mtd/spi-nor/. This layer would perform the following tasks: - During probing, store information from all the connected flashes, whether in stacked or parallel mode, and present it as a single device to the MTD layer. - Register callbacks and manage MTD device registration within the new layer instead of in spi-nor/core.c - There will very minimal change in spinor/core.c as the stacked & parallel handling will be managed by the new layer on top of SPI-NOR. - In stacked mode, the new layer will select the appropriate spi-nor flash based on the address provided by the MTD layer during flash operations. - Manage flash crossover operations in stacked mode. - Handle odd byte count requests from the MTD layer during flash operations in parallel mode. For implementing this the current DT binding[1] [2] [3] need to be updated as follows. stacked-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each stacked flash will have its own flash node. This approach allows flashes of different makes and sizes to be stacked together, as each flash will be probed individually. - Each of the flash node will have its own "reg" property that will contain its physical CS. - The stacked-memories DT bindings will contain the phandles of the flash nodes connected in stacked mode. - The first flash node will contain the mtd partition that would have the cross over memory staring at a memory location in the first flash and ending at some memory location of the 2nd flash - The new layer will update the mtd->size and other mtd_info parameters after both the flashes are probed and will call mtd_device_register with the combined information. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00>; stacked-memories = <&flash@0 &flash@1>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; }; } flash@1 { compatible = "jedec,spi-nor" reg = <0x01>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-1"; reg = <0x0 0x800000>; }; } } parallel-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each flash connected in parallel mode should be identical and will have one flash node for both the flash devices. - The "reg" prop will contain the physical CS number for both the connected flashes. - The new layer will double the mtd-> size and register it with the mtd layer. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00 0x01>; parallel-memories; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; } }; } Regards, Amit

2 6

[PATCH v1] ASoC: codecs: lpass-macro: Add support for channel map mixer control
by Mohammad Rafi Shaik 08 Oct '24

08 Oct '24

From: Rohit kumar <quic_rohkumar(a)quicinc.com> Add channel map mixer control for lpass macro codec DAIs. Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> Co-developed-by: Mohammad Rafi Shaik <quic_mohs(a)quicinc.com> Signed-off-by: Mohammad Rafi Shaik <quic_mohs(a)quicinc.com> --- sound/soc/codecs/lpass-macro-common.c | 55 +++++++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 2 + sound/soc/codecs/lpass-rx-macro.c | 12 ++++++ sound/soc/codecs/lpass-tx-macro.c | 9 +++++ sound/soc/codecs/lpass-va-macro.c | 9 +++++ sound/soc/codecs/lpass-wsa-macro.c | 12 ++++++ 6 files changed, 99 insertions(+) diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c index 6e3b8d0897dd..777af4885245 100644 --- a/sound/soc/codecs/lpass-macro-common.c +++ b/sound/soc/codecs/lpass-macro-common.c @@ -8,12 +8,67 @@ #include <linux/platform_device.h> #include <linux/pm_domain.h> #include <linux/pm_runtime.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> #include "lpass-macro-common.h" static DEFINE_MUTEX(lpass_codec_mutex); static enum lpass_codec_version lpass_codec_version; +static int lpass_macro_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct snd_soc_dai *dai = info->private_data; + u32 *chmap_data = NULL; + u32 rx_ch_cnt = 0; + u32 tx_ch_cnt = 0; + u32 rx_ch, tx_ch; + + chmap_data = kzalloc(sizeof(u32) * 2, GFP_KERNEL); + if (!chmap_data) + return -ENOMEM; + + snd_soc_dai_get_channel_map(dai, &tx_ch_cnt, &tx_ch, &rx_ch_cnt, &rx_ch); + if (rx_ch_cnt) { + chmap_data[0] = rx_ch_cnt; + chmap_data[1] = rx_ch; + } else if (tx_ch_cnt) { + chmap_data[0] = tx_ch_cnt; + chmap_data[1] = tx_ch; + } + memcpy(ucontrol->value.bytes.data, chmap_data, sizeof(u32) * 2); + + kfree(chmap_data); + return 0; +} + +int lpass_macro_add_chmap_ctls(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai, int dir) +{ + struct snd_pcm_chmap *info; + int ret; + + info = kzalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, dir, NULL, + 2 * sizeof(u32), 0, &info); + if (ret < 0) { + kfree(info); + return ret; + } + + /* override handlers */ + info->private_data = dai; + info->kctl->get = lpass_macro_chmap_ctl_get; + return 0; +} +EXPORT_SYMBOL_GPL(lpass_macro_add_chmap_ctls); + struct lpass_macro *lpass_macro_pds_init(struct device *dev) { struct lpass_macro *l_pds; diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index fb4b96cb2b23..23ed6836addf 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -40,6 +40,8 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev); void lpass_macro_pds_exit(struct lpass_macro *pds); void lpass_macro_set_codec_version(enum lpass_codec_version version); enum lpass_codec_version lpass_macro_get_codec_version(void); +int lpass_macro_add_chmap_ctls(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai, int dir); static inline void lpass_macro_pds_exit_action(void *pds) { diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 71e0d3bffd3f..05bd0f8d126b 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -1954,10 +1954,22 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int rx_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + if (dai->id == RX_MACRO_AIF_ECHO) + dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops rx_macro_dai_ops = { .hw_params = rx_macro_hw_params, .get_channel_map = rx_macro_get_channel_map, .mute_stream = rx_macro_digital_mute, + .pcm_new = rx_macro_pcm_new, }; static struct snd_soc_dai_driver rx_macro_dai[] = { diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index a134584acf90..c7a8e2694e36 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1211,10 +1211,19 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int tx_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops tx_macro_dai_ops = { .hw_params = tx_macro_hw_params, .get_channel_map = tx_macro_get_channel_map, .mute_stream = tx_macro_digital_mute, + .pcm_new = tx_macro_pcm_new, }; static struct snd_soc_dai_driver tx_macro_dai[] = { diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index c781da476240..1a13f472ed3a 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -939,10 +939,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int va_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops va_macro_dai_ops = { .hw_params = va_macro_hw_params, .get_channel_map = va_macro_get_channel_map, .mute_stream = va_macro_digital_mute, + .pcm_new = va_macro_pcm_new, }; static struct snd_soc_dai_driver va_macro_dais[] = { diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index c989d82d1d3c..7c4fd4bf2668 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1344,9 +1344,21 @@ static int wsa_macro_get_channel_map(const struct snd_soc_dai *dai, return 0; } +static int wsa_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + if (dai->id == WSA_MACRO_AIF_VI || dai->id == WSA_MACRO_AIF_ECHO) + dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops wsa_macro_dai_ops = { .hw_params = wsa_macro_hw_params, .get_channel_map = wsa_macro_get_channel_map, + .pcm_new = wsa_macro_pcm_new, }; static struct snd_soc_dai_driver wsa_macro_dai[] = { -- 2.25.1

2 1

[PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param()
by (학생) 장인규 (컴퓨터공학과) 05 Oct '24

05 Oct '24

From 791716bf359b8540c519810848fd1f8006d7c3c5 Mon Sep 17 00:00:00 2001 From: Ingyu Jang <ingyujang25(a)unist.ac.kr> Date: Thu, 26 Sep 2024 19:40:04 +0900 Subject: [PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param() The aio_src_set_param() function did not previously check the return values of regmap_write() and regmap_update_bits(). If these functions fail, it could lead to silent failures when configuring the sample rate converter (SRC), causing improper behavior in audio processing without any indication of an error. This patch modifies aio_src_set_param to check the return values of regmap_write() and regmap_update_bits(). If either function returns an error, the error code is propagated back to the caller to ensure proper error handling. This change aligns with the existing error-handling behavior in functions like uniphier_aio_prepare(), where a failure in a sub-function should result in an immediate return of the error. Signed-off-by: Ingyu Jang <ingyujang25(a)unist.ac.kr> --- sound/soc/uniphier/aio-core.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 0eba60758134..2c4e8b873253 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -921,16 +921,19 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, { struct regmap *r = sub->aio->chip->regmap; u32 v; + int ret; if (sub->swm->dir != PORT_DIR_OUTPUT) return 0; - regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), + ret = regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), OPORTMXSRC1CTR_THMODE_SRC | OPORTMXSRC1CTR_SRCPATH_CALC | OPORTMXSRC1CTR_SYNC_ASYNC | OPORTMXSRC1CTR_FSIIPSEL_INNER | OPORTMXSRC1CTR_FSISEL_ACLK); + if (ret) + return ret; switch (params_rate(params)) { default: @@ -951,12 +954,18 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, break; } - regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), + + ret = regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), v | OPORTMXRATE_I_ACLKSRC_APLL | OPORTMXRATE_I_LRCKSTP_STOP); - regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), + if (ret) + return ret; + + ret = regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), OPORTMXRATE_I_LRCKSTP_MASK, OPORTMXRATE_I_LRCKSTP_START); + if (ret) + return ret; return 0; } -- 2.34.1

2 2

[PATCH v2] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 04 Oct '24

04 Oct '24

The headset push button cannot work properly in case of the ALC257. This patch reverted the previous commit to correct the side effect. Fixes: ef9718b3d54e ("ALSA: hda/realtek: Fix noise from speakers on Lenovo IdeaPad 3 15IAU7") Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f787ff4182d4..86db1a4ec24c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: case 0x10ec0283: case 0x10ec0285: -- 2.34.1

3 2

[PATCH v1] soundwire: Correct some typos in comments
by Shen Lichuan 03 Oct '24

03 Oct '24

Fixed some confusing spelling errors that were currently identified, the details are as follows: -in the code comments: drivers/soundwire/bus.c:115: enumuration==> enumeration drivers/soundwire/qcom.c:1176: wil ==> will drivers/soundwire/sysfs_slave.c:218: highligh ==> highlight Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- drivers/soundwire/bus.c | 2 +- drivers/soundwire/qcom.c | 2 +- drivers/soundwire/sysfs_slave.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 263ca32f0c5c..d1dc62c34f1c 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -112,7 +112,7 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* Set higher order bits */ *bus->assigned = ~GENMASK(SDW_BROADCAST_DEV_NUM, SDW_ENUM_DEV_NUM); - /* Set enumuration device number and broadcast device number */ + /* Set enumeration device number and broadcast device number */ set_bit(SDW_ENUM_DEV_NUM, bus->assigned); set_bit(SDW_BROADCAST_DEV_NUM, bus->assigned); diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2b403b14066c 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1173,7 +1173,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, else sconfig.direction = SDW_DATA_DIR_RX; - /* hw parameters wil be ignored as we only support PDM */ + /* hw parameters will be ignored as we only support PDM */ sconfig.ch_count = 1; sconfig.frame_rate = params_rate(params); sconfig.type = stream->type; diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index f4259710dd0f..c5c22d1708ec 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -215,7 +215,7 @@ const struct attribute_group *sdw_attr_groups[] = { /* * the status is shown in capital letters for UNATTACHED and RESERVED - * on purpose, to highligh users to the fact that these status values + * on purpose, to highlight users to the fact that these status values * are not expected. */ static const char *const slave_status[] = { -- 2.17.1

2 1

[PATCH 0/4] refactor amd soundwire manager driver for acp 6.3 platform
by Vijendar Mukunda 03 Oct '24

03 Oct '24

This patch series refactors the amd soundwire manager driver to support acp 6.3 platform based on acp pci revision id. This patch series prepared on top of asoc tree for-next branch. This patch series should be applied at one go. @Vinod: Please help to review soundwire patches and provide ack-by for this patch series. Vijendar Mukunda (4): soundwire: amd: pass acp pci revision id as resource data soundwire: amd: refactor existing code for acp 6.3 platform ASoC: SOF: amd: pass acp_rev as soundwire resource data ASoC: amd: ps: pass acp pci revision id as soundwire resource data drivers/soundwire/amd_init.c | 1 + drivers/soundwire/amd_manager.c | 99 ++++++++++++++++++++----------- drivers/soundwire/amd_manager.h | 16 ++--- include/linux/soundwire/sdw_amd.h | 6 ++ sound/soc/amd/ps/acp63.h | 2 + sound/soc/amd/ps/pci-ps.c | 2 + sound/soc/sof/amd/acp.c | 1 + 7 files changed, 84 insertions(+), 43 deletions(-) -- 2.34.1

3 7

[PATCH] ASoC: intel: sof_sdw: Add check devm_kasprintf() returned value
by Charles Han 02 Oct '24

02 Oct '24

devm_kasprintf() can return a NULL pointer on failure but this returned value is not checked. Fixes: b359760d95ee ("ASoC: intel: sof_sdw: Add simple DAI link creation helper") Signed-off-by: Charles Han <hanchunchao(a)inspur.com> --- sound/soc/intel/boards/sof_sdw.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d258728d64cf..26917f6f15cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -919,6 +919,9 @@ static int create_ssp_dailinks(struct snd_soc_card *card, char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); char *codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", ssp_info->acpi_id, j++); + if (!name || !cpu_dai_name || !codec_name) + return -ENOMEM; + int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; @@ -985,6 +988,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, for (i = 0; i < hdmi_num; i++) { char *name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1); + if (!name || !cpu_dai_name) + return -ENOMEM; + char *codec_name, *codec_dai_name; if (intel_ctx->hdmi.idisp_codec) { @@ -996,6 +1002,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, codec_dai_name = "snd-soc-dummy-dai"; } + if (!codec_dai_name) + return -ENOMEM; + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, 1, 0, // HDMI only supports playback cpu_dai_name, platform_component->name, @@ -1019,6 +1028,9 @@ static int create_bt_dailinks(struct snd_soc_card *card, SOF_BT_OFFLOAD_SSP_SHIFT; char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); + if (!name || !cpu_dai_name) + return -ENOMEM; + int ret; ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, -- 2.31.1

3 2

[RFC PATCH v3 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 02 Oct '24

02 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

3 9

[PATCH] Fix possible NULL Pointer Dereference in 'asoc_qcom_lpass_cpu_platform_probe'
by Gax-c 01 Oct '24

01 Oct '24

A 'devm_kzalloc' in 'asoc_qcom_lpass_cpu_platform_probe' could possibly return NULL pointer. NULL Pointer Dereference may be triggerred in 'asoc_qcom_lpass_cpu_platform_probe' without addtional check. Add a null check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zijie Zhao <zzjas98(a)gmail.com> Reported-by: Chenyuan Yang <chenyuan0y(a)gmail.com> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..a8e56f47f237 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1243,6 +1243,9 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; + /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, drvdata->lpaif_map); -- 2.25.1

5 7

[PATCH 0/3] ASoC: fsl_micfil: fix and improvement
by Shengjiu Wang 01 Oct '24

01 Oct '24

Fix the usage of regmap_write_bits(). Move mclk clock enablement to late stage. Enable the micfil error interrupt. Shengjiu Wang (3): ASoC: fsl_micfil: fix regmap_write_bits usage ASoC: fsl_micfil: Add mclk enable flag ASoC: fsl_micfil: Enable micfil error interrupt sound/soc/fsl/fsl_micfil.c | 49 ++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 10 deletions(-) -- 2.34.1

3 5

[PATCH -next 0/5] sound/soc: fix some macro definitions and usages
by Hongbo Li 01 Oct '24

01 Oct '24

Many variable in macro are not used as we used macro_check script to detect and mamually check, let us address these issues. Hongbo Li (5): sound/soc: fix macro definition on TWL4030_OUTPUT_PGA sound/soc remove unused substream in macro soc_component_mark_pop sound/soc remove unused substream in macro soc_dai_mark_pop sound/soc remove unused substream in macro soc_link_mark_pop sound/soc: fix macro definition on STM_SAI_HAS_EXT_SYNC sound/soc/codecs/twl4030.c | 12 ++++++------ sound/soc/soc-component.c | 14 +++++++------- sound/soc/soc-dai.c | 10 +++++----- sound/soc/soc-link.c | 10 +++++----- sound/soc/stm/stm32_sai_sub.c | 2 +- 5 files changed, 24 insertions(+), 24 deletions(-) -- 2.34.1

2 7

[PATCH v2 0/2] {ASoC,media}: constify snd_soc_component_driver when used as read-only
by Javier Carrasco 01 Oct '24

01 Oct '24

Declare `snd_soc_component_driver` as const to move their declarations to read-only sections for the drivers that do not modify the struct after its declaration. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. To: Tim Harvey <tharvey(a)gateworks.com> To: Mauro Carvalho Chehab <mchehab(a)kernel.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> To: Ray Jui <rjui(a)broadcom.com> To: Scott Branden <sbranden(a)broadcom.com> To: Broadcom internal kernel review list <bcm-kernel-feedback-list(a)broadcom.com> To: Shenghao Ding <shenghao-ding(a)ti.com> To: Kevin Lu <kevin-lu(a)ti.com> To: Baojun Xu <baojun.xu(a)ti.com> To: Olivier Moysan <olivier.moysan(a)foss.st.com> To: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> To: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> To: Alexandre Torgue <alexandre.torgue(a)foss.st.com> To: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> To: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: linux-media(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> Changes in v2: - drop cs43130 and sti-sas, as they modifiy the struct in the probe function. - Link to v1: https://lore.kernel.org/r/20240725-const_snd_soc_component_driver-v1-0-3d7e… --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 12 files changed, 13 insertions(+), 13 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

2 3

[PATCH V2 0/9] ASoC: amd: acp: refactor acp version differentiation logic
by Vijendar Mukunda 01 Oct '24

01 Oct '24

Currently different logics being used in the code for acp version differentiation. This patch series refactors the code to use acp pci revision id for handling acp version specific code. Changes since v1: - Add patch to update mach_params subsystem_rev field in machine select logic. Vijendar Mukunda (9): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: update mach_params subsystem_rev field ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver sound/soc/amd/acp/acp-i2s.c | 38 +++++++++++++-------------- sound/soc/amd/acp/acp-legacy-common.c | 24 ++++++++--------- sound/soc/amd/acp/acp-legacy-mach.c | 8 +++--- sound/soc/amd/acp/acp-mach-common.c | 38 +++++++++++++++++++-------- sound/soc/amd/acp/acp-mach.h | 12 +++------ sound/soc/amd/acp/acp-pci.c | 7 +---- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 15 ++++++----- sound/soc/amd/acp/acp-rembrandt.c | 4 +-- sound/soc/amd/acp/acp-renoir.c | 4 +-- sound/soc/amd/acp/acp-sof-mach.c | 6 ++--- sound/soc/amd/acp/acp63.c | 4 +-- sound/soc/amd/acp/acp70.c | 14 ++++------ sound/soc/amd/acp/acp_common.h | 19 ++++++++++++++ sound/soc/amd/acp/amd.h | 9 ++----- 15 files changed, 109 insertions(+), 95 deletions(-) create mode 100644 sound/soc/amd/acp/acp_common.h -- 2.34.1

2 10

[PATCH v2 0/2] Updates for Atmel SSC DAI
by Andrei Simion 01 Oct '24

01 Oct '24

This patch set includes two updates for the Atmel SSC DAI driver: - Address the limitation with the S24_LE format. - Add stream names for DPCM and future use-cases. Codrin Ciubotariu (2): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) base-commit: 7083504315d64199a329de322fce989e1e10f4f7 -- 2.34.1

3 4

[PATCH 0/2] Updates for mchp-spdif(tx/rx)
by Andrei Simion 01 Oct '24

01 Oct '24

This patch set includes two updates for the MCHP SPDIF RX and TX drivers. The patches remove the interface name from the stream_name, allowing the interface name and index to be set in the Device Tree (DT) using the sound-name-prefix string property. Codrin Ciubotariu (2): ASoC: atmel: mchp-spdiftx: Remove interface name from stream_name ASoC: atmel: mchp-spdifrx: Remove interface name from stream_name sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) base-commit: 5acd9952f95fb4b7da6d09a3be39195a80845eb6 -- 2.34.1

2 3

[PATCH 0/8] ASoC: amd: acp: refactor acp version differentiation logic
by Vijendar Mukunda 01 Oct '24

01 Oct '24

Currently different logics being used in the code for acp version differentiation. This patch series refactors the code to use acp pci revision id for handling acp version specific code. Vijendar Mukunda (8): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver sound/soc/amd/acp/acp-i2s.c | 38 +++++++++++++-------------- sound/soc/amd/acp/acp-legacy-common.c | 24 ++++++++--------- sound/soc/amd/acp/acp-legacy-mach.c | 8 +++--- sound/soc/amd/acp/acp-mach-common.c | 38 +++++++++++++++++++-------- sound/soc/amd/acp/acp-mach.h | 12 +++------ sound/soc/amd/acp/acp-pci.c | 7 +---- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 14 +++++----- sound/soc/amd/acp/acp-rembrandt.c | 4 +-- sound/soc/amd/acp/acp-renoir.c | 4 +-- sound/soc/amd/acp/acp-sof-mach.c | 6 ++--- sound/soc/amd/acp/acp63.c | 4 +-- sound/soc/amd/acp/acp70.c | 14 ++++------ sound/soc/amd/acp/acp_common.h | 19 ++++++++++++++ sound/soc/amd/acp/amd.h | 9 ++----- 15 files changed, 108 insertions(+), 95 deletions(-) create mode 100644 sound/soc/amd/acp/acp_common.h -- 2.34.1

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[PATCH v28 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 01 Oct '24

01 Oct '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v28: - Updated comments and commit log in the stop endpoint sync patch. Clarified that the default stop endpoint completion routine won't fully run as expected since it has a completion associated w/ the command. - Added a null check for sb->xhci within xhci_sideband_create_interrupter(). This is to just ensure that caller has registered sideband before calling create interrupter. Changes in v27: - Added some comments and notes about the offload design. Enforcing the q6routing to only allow one USB mixer (PCM device) to be enabled at a time. - Modified SND_JACK_USB notifications for all USB audio offloadable devices plugged in - Rebased on latest XHCI secondary interrupter IMOD changes upstream. Modified the change in this series to allow for XHCI sideband to set the IMOD for sideband clients. - Updated documentation on how USB SND kcontrols are involved in the overall design. - Remove mutex locking from suspend/resume platform ops, as USB core ensures that the interface and device are in the RPM_ACTIVE state while disconnect is handled. Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: xhci-sideband: Set IMOD for xHCI sideband clients usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 456 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 31 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 47 +- drivers/usb/host/xhci.h | 16 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 70 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 139 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 32 +- sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 342 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 106 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5870 insertions(+), 55 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

3 41

[PATCH] ASoC: fsl_sai: Enable 'FIFO continue on error' FCONT bit
by Shengjiu Wang 01 Oct '24

01 Oct '24

FCONT=1 means On FIFO error, the SAI will continue from the same word that caused the FIFO error to set after the FIFO warning flag has been cleared. Set FCONT bit in control register to avoid the channel swap issue after SAI xrun. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 5 ++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ab58a4461073..634168d2bb6e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -613,6 +613,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Set to avoid channel swap */ + val_cr4 |= FSL_SAI_CR4_FCONT; + /* Set to output mode to avoid tri-stated data pins */ if (tx) val_cr4 |= FSL_SAI_CR4_CHMOD; @@ -699,7 +702,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | - FSL_SAI_CR4_CHMOD_MASK, + FSL_SAI_CR4_CHMOD_MASK | FSL_SAI_CR4_FCONT_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index dadbd16ee394..9c4d19fe22c6 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -137,6 +137,7 @@ /* SAI Transmit and Receive Configuration 4 Register */ +#define FSL_SAI_CR4_FCONT_MASK BIT(28) #define FSL_SAI_CR4_FCONT BIT(28) #define FSL_SAI_CR4_FCOMB_SHIFT BIT(26) #define FSL_SAI_CR4_FCOMB_SOFT BIT(27) -- 2.34.1

2 1

[PATCH alsa-ucm-conf 1/2] sof-soundwire: Change mapped control names to make them unique
by Maciej Strozek 01 Oct '24

01 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found as substrings in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Fixes: e87dde51d689 ("sof-soundwire: Add basic support for cs42l43") Fixes: e4277a204cd6 ("sof-soundwire: Add support for cs42l43/cs35l56 bridge configuration") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/codecs/cs42l43/init.conf | 4 ++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..cf61833 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..7c36655 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Combined Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Combined Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Combined" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } -- 2.39.5

3 5

[PATCH v2 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 01 Oct '24

01 Oct '24

Hi Mark This is v2 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

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[PATCH] sound: soc: codecs: tlv320aic3x: Fix codec gpio-reset
by Benjamin Hahn 30 Sep '24

30 Sep '24

The TLV320AIC3007 requires a hardware reset after power-up for proper operation. After all power supplies are at their specified values, the RESET pin must be driven low for at least 10 ns. Even though the datasheet sais min 10ns, I needed more than 10ns when testing this out. 15ns worked for me. Signed-off-by: Benjamin Hahn <B.Hahn(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 56e795a00e22..d002fc0b99b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1367,8 +1367,11 @@ static int aic3x_regulator_event(struct notifier_block *nb, * Put codec to reset and require cache sync as at least one * of the supplies was disabled */ - if (aic3x->gpio_reset) + if (aic3x->gpio_reset) { gpiod_set_value(aic3x->gpio_reset, 1); + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + } regcache_mark_dirty(aic3x->regmap); } @@ -1813,6 +1816,10 @@ int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver gpiod_set_consumer_name(aic3x->gpio_reset, "tlv320aic3x reset"); + /* CODEC datasheet says minimum of 10ns */ + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; --- base-commit: ad46e8f95e931e113cb98253daf6d443ac244cde change-id: 20240930-wip-bhahn-tlv320aic3x_fix_reset-cca918a4889d Best regards, -- Benjamin Hahn <B.Hahn(a)phytec.de>

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[PATCH] ASoC: fsl: Use maple tree register cache
by Mark Brown 30 Sep '24

30 Sep '24

Several of the NXP drivers use regmaps with a rbtree register cache. Since the maple tree cache is uisng a generally more modern data structure which makes implementation choices more suitable for modern systems let's convert these drivers to it. This should have no practical impact. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..910b4ce7671c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -169,7 +169,7 @@ static const struct regmap_config fsl_aud2htx_regmap_config = { .readable_reg = fsl_aud2htx_readable_reg, .volatile_reg = fsl_aud2htx_volatile_reg, .writeable_reg = fsl_aud2htx_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static const struct of_device_id fsl_aud2htx_dt_ids[] = { diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..5149e732f763 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1748,7 +1748,7 @@ static const struct regmap_config fsl_easrc_regmap_config = { .rd_table = &fsl_easrc_readable_table, .wr_table = &fsl_easrc_writeable_table, .volatile_table = &fsl_easrc_volatileable_table, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; #ifdef DEBUG diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..c0dc9bebb77b 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -955,7 +955,7 @@ static const struct regmap_config fsl_micfil_regmap_config = { .readable_reg = fsl_micfil_readable_reg, .volatile_reg = fsl_micfil_volatile_reg, .writeable_reg = fsl_micfil_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; /* END OF REGMAP */ --- base-commit: 98f7e32f20d28ec452afb208f9cffc08448a2652 change-id: 20240924-asoc-imx-maple-abb62f013533 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH] ASoC: meson: axg-iface: set continuous rates
by Jerome Brunet 30 Sep '24

30 Sep '24

The axg TDM HW does not depend on a selected set of rates. The hardware itself, just takes an input clock and work with it, regardless of its rate. In this way, the rates TDM can take are continuous. What might force the use of specific rate are the PLL available as clock and/or the codecs facing the TDM HW. Either way, this constraint does not belong in the TDM interface driver. Allow any rate as far as TDM is concerned by setting SNDRV_PCM_RATE_CONTINUOUS with an interval it has been tested with. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 12 +++++++++--- sound/soc/meson/axg-tdm.h | 2 -- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 62057c71f742..09103eef2a97 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -442,14 +442,18 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Playback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_PAD, @@ -461,7 +465,9 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Loopback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_LOOPBACK, diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 1a17f546ce6e..acfcd48f8a00 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -15,8 +15,6 @@ #define AXG_TDM_NUM_LANES 4 #define AXG_TDM_CHANNEL_MAX 128 -#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_768000) #define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ --- base-commit: cd87a98b53518e44cf3c1a7c1c07c869ce33bf83 change-id: 20240920-asoc-axg-iface-continuous-b94cfa234787 Best regards, -- Jerome

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[PATCH] ASoC: tas2781: Fix redundant parameter assignment
by Tang Bin 30 Sep '24

30 Sep '24

In these functions, the variable 'rc' is redundant, thus remove it. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 35 ++++++++-------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a8d97dd7..68887799e 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -650,7 +650,6 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; @@ -659,9 +658,7 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, @@ -673,7 +670,6 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG; @@ -681,9 +677,7 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, reg = TAS2563_RUNTIME_RE_REG; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, @@ -696,7 +690,6 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); @@ -707,9 +700,7 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, else return -1; dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, @@ -721,13 +712,10 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, @@ -739,13 +727,10 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_nop_get( @@ -1115,14 +1100,12 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); int dev_id = ucontrol->value.integer.value[0]; - int max = tas_priv->ndev - 1, rc; + int max = tas_priv->ndev - 1; dev_id = clamp(dev_id, 0, max); guard(mutex)(&tas_priv->codec_lock); - rc = tasdev_chn_switch(tas_priv, dev_id); - - return rc; + return tasdev_chn_switch(tas_priv, dev_id); } static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) @@ -1339,10 +1322,8 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) i++; } - rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + return snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls < i ? nctrls : i); - - return rc; } static void tasdevice_fw_ready(const struct firmware *fmw, -- 2.33.0

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