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

[PATCH v1 0/2] ALSA: hda: cs35l41: Ignore errors when configuring interrupts to allow laptops with bad ACPI to play audio
by Stefan Binding 30 Apr '24

30 Apr '24

Some laptops have a bad _CRS defined for its interrupt. This errors out inside the driver probe for CS35L41 HDA. However, there is no fix in software for this bad configuration, and it is unlikely to get a new BIOS, therefore it is better to allow the laptop to continue probe, instead of erroring out inside the probe. The interrupt is only used for Error Detection and recovery, without the interrupt the driver will continue to function, but errors will not be detected, and recovery will require a reboot. Also add support for ASUS ROG 2024 laptops. These laptops were released without _DSD, so need to be added into the CS35L41 config table. Quirks for these laptops already exist. Stefan Binding (2): ALSA: hda: cs35l41: Ignore errors when configuring IRQs ALSA: hda: cs35l41: Add support for ASUS ROG 2024 Laptops sound/pci/hda/cs35l41_hda.c | 69 +++++++++++++++++++--------- sound/pci/hda/cs35l41_hda_property.c | 10 ++++ 2 files changed, 57 insertions(+), 22 deletions(-) -- 2.34.1

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[PATCH 0/2][next] firewire: Avoid -Wflex-array-member-not-at-end warnings
by Gustavo A. R. Silva 29 Apr '24

29 Apr '24

-Wflex-array-member-not-at-end is coming in GCC-14, and we are getting ready to enable it globally. So, we are deprecating flexible-array members in the middle of another struct. There are currently a couple of local structures (`u` and `template`) that are using a flexible `struct fw_iso_packet` as header for a couple of on-stack arrays. We make use of the `struct_group_tagged()` helper to separate the flexible array from the rest of the members in the flexible structure, and, with this, we can now declare objects of the type of the tagged struct, without embedding the flexible array in the middle of another struct. We also use `container_of()` whenever we need to retrieve a pointer to the flexible structure, through which the flexible-array member can be accessed. With these changes, we fix a couple of -Wflex-array-member-not-at-end warnings. Gustavo A. R. Silva (2): firewire: Avoid -Wflex-array-member-not-at-end warning ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning drivers/firewire/core-cdev.c | 9 +++++---- include/linux/firewire.h | 16 +++++++++------- sound/firewire/amdtp-stream.c | 8 +++++--- 3 files changed, 19 insertions(+), 14 deletions(-) -- 2.34.1

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[PATCH] ASoC: sunxi: DMIC: Add controls for adjusting the mic gains
by Joao Schim 29 Apr '24

29 Apr '24

The AllWinner H6 and later SoCs that sport a DMIC block contain a set of registers to control the gain (left + right) of each of the four supported channels. Add ASoC controls for changing each of the stereo channel gains using alsamixer and alike --- sound/soc/sunxi/sun50i-dmic.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index c76628bc86c6..f8613d8c3462 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -14,6 +14,7 @@ #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tlv.h> #define SUN50I_DMIC_EN_CTL (0x00) #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) @@ -43,6 +44,17 @@ #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) #define SUN50I_DMIC_CNT (0x2c) #define SUN50I_DMIC_CNT_N (1 << 0) +#define SUN50I_DMIC_D0D1_VOL_CTR (0x30) + #define SUN50I_DMIC_D0D1_VOL_CTR_0R (0) + #define SUN50I_DMIC_D0D1_VOL_CTR_0L (8) + #define SUN50I_DMIC_D0D1_VOL_CTR_1R (16) + #define SUN50I_DMIC_D0D1_VOL_CTR_1L (24) +#define SUN50I_DMIC_D2D3_VOL_CTR (0x34) + #define SUN50I_DMIC_D2D3_VOL_CTR_2R (0) + #define SUN50I_DMIC_D2D3_VOL_CTR_2L (8) + #define SUN50I_DMIC_D2D3_VOL_CTR_3R (16) + #define SUN50I_DMIC_D2D3_VOL_CTR_3L (24) + #define SUN50I_DMIC_HPF_CTRL (0x38) #define SUN50I_DMIC_VERSION (0x50) @@ -273,8 +285,30 @@ static const struct of_device_id sun50i_dmic_of_match[] = { }; MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); +static const DECLARE_TLV_DB_SCALE(sun50i_dmic_vol_scale, -12000, 75, 1); + +static const struct snd_kcontrol_new sun50i_dmic_controls[] = { + + SOC_DOUBLE_TLV("DMIC Channel 0 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_0L, SUN50I_DMIC_D0D1_VOL_CTR_0R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 1 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_1L, SUN50I_DMIC_D0D1_VOL_CTR_1R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 2 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_2L, SUN50I_DMIC_D2D3_VOL_CTR_2R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 3 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_3L, SUN50I_DMIC_D2D3_VOL_CTR_3R, + 0xFF, 0, sun50i_dmic_vol_scale), + + +}; + static const struct snd_soc_component_driver sun50i_dmic_component = { .name = "sun50i-dmic", + .controls = sun50i_dmic_controls, + .num_controls = ARRAY_SIZE(sun50i_dmic_controls), }; static int sun50i_dmic_runtime_suspend(struct device *dev) -- 2.25.1

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[PATCH 0/3] sysfs: Group visibility fixups
by Dan Williams 29 Apr '24

29 Apr '24

Hey Greg, Marc was able to get me a backtrace for that bootup hang scenario that Pierre noted here [1]. A Tested-by is still pending, but I am certain this is the issue, and it may impact more than just the one platform if that "empty_attrs" pattern has been repeated anywhere else in the kernel. I also took some time to document how to use the helpers a bit better, and introduce DEFINE_SIMPLE_SYSFS_GROUP_VISIBLE() for cases where all that matters is group visibility and not per attribute visibility. [1]: http://lore.kernel.org/r/b93ec9c2-23f5-486b-a3dc-ed9b960df359@linux.intel.c… --- Dan Williams (3): sysfs: Fix crash on empty group attributes array sysfs: Document new "group visible" helpers sysfs: Introduce DEFINE_SIMPLE_SYSFS_GROUP_VISIBLE() fs/sysfs/group.c | 4 +- include/linux/sysfs.h | 85 +++++++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 84 insertions(+), 5 deletions(-) base-commit: 70317fd24b419091aa0a6dc3ea3ec7bb50c37c32

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[PATCH] ALSA: hda: intel-sdw-acpi: fix usage of device_get_named_child_node()
by Pierre-Louis Bossart 29 Apr '24

29 Apr '24

The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid a leaked reference. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-sdw-acpi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index 5f60658c6051..d7417a40392b 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -45,6 +45,8 @@ static bool is_link_enabled(struct fwnode_handle *fw_node, u8 idx) "intel-quirk-mask", &quirk_mask); + fwnode_handle_put(link); + if (quirk_mask & SDW_INTEL_QUIRK_MASK_BUS_DISABLE) return false; -- 2.40.1

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[PATCH] ALSA: hda: intel-dsp-config: harden I2C/I2S codec detection
by Pierre-Louis Bossart 29 Apr '24

29 Apr '24

The SOF driver is selected whenever specific I2C/I2S HIDs are reported as 'present' in the ACPI DSDT. In some cases, an HID is reported but the hardware does not actually rely on I2C/I2S. This false positive leads to an invalid selection of the SOF driver and as a result an invalid topology is loaded. This patch hardens the detection with a check that the NHLT table is consistent with the report of an I2S-based codec in DSDT. This table should expose at least one SSP endpoint configured for an I2S-codec connection. Tested on Huawei Matebook D14 (NBLB-WAX9N) using an HDaudio codec with an invalid ES8336 ACPI HID reported: [ 7.858249] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380 [ 7.858312] snd_hda_intel 0000:00:1f.3: snd_intel_dsp_find_config: no valid SSP found for HID ESSX8336, skipped Reported-by: Mauro Carvalho Chehab <mchehab(a)kernel.org> Tested-by: Mauro Carvalho Chehab <mchehab(a)kernel.org> Closes: https://github.com/thesofproject/linux/issues/4934 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 27 +++++++++++++++++++++++++-- 1 file changed, 25 insertions(+), 2 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 6a384b922e4f..d1f6cdcf1866 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -557,9 +557,32 @@ static const struct config_entry *snd_intel_dsp_find_config if (table->codec_hid) { int i; - for (i = 0; i < table->codec_hid->num_codecs; i++) - if (acpi_dev_present(table->codec_hid->codecs[i], NULL, -1)) + for (i = 0; i < table->codec_hid->num_codecs; i++) { + struct nhlt_acpi_table *nhlt; + bool ssp_found = false; + + if (!acpi_dev_present(table->codec_hid->codecs[i], NULL, -1)) + continue; + + nhlt = intel_nhlt_init(&pci->dev); + if (!nhlt) { + dev_warn(&pci->dev, "%s: NHLT table not found, skipped HID %s\n", + __func__, table->codec_hid->codecs[i]); + continue; + } + + if (intel_nhlt_has_endpoint_type(nhlt, NHLT_LINK_SSP) && + intel_nhlt_ssp_endpoint_mask(nhlt, NHLT_DEVICE_I2S)) + ssp_found = true; + + intel_nhlt_free(nhlt); + + if (ssp_found) break; + + dev_warn(&pci->dev, "%s: no valid SSP found for HID %s, skipped\n", + __func__, table->codec_hid->codecs[i]); + } if (i == table->codec_hid->num_codecs) continue; } -- 2.40.1

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ALSA: Digital Microphone
by Muni Sekhar 29 Apr '24

29 Apr '24

Hi all, We have a Digital Microphone block on a Xilinx FPGA board connected to x86 host CPU through PCIe, it basically emulates a Microphone sending data to DUT(device under test) in PDM format. Input to the Digital Microphone block is Signed 32-bit PCM sample. Xilinx FPGA code has logic to convert PCM samples to PDM with a Sigma-Delta Converter. Could you kindly recommend any specific ALSA driver within the Linux kernel that matches with my hardware setup. Additionally, if there are any relevant documentation or community discussions that you could point me to, it would be immensely helpful. -- Thanks, Sekhar

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[PATCH] ALSA: usb-audio: Add sampling rates support for Mbox3
by ManuLinares 29 Apr '24

29 Apr '24

This adds support for all sample rates supported by the hardware, Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000} Fixes syncing clock issues that presented as pops. To test this, without this patch playing 440hz tone produces pops. Clock is now synced between playback and capture interfaces so no more latency drift issue when using pipewire pro-profile. (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900) Signed-off-by: ManuLinares <mbarriolinares(a)gmail.com> --- sound/usb/quirks-table.h | 37 +++++++++++------- sound/usb/quirks.c | 84 +++++++++++++++++++++++++++++++--------- 2 files changed, 88 insertions(+), 33 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5d72dc8441cb..5228afbc00e3 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3013,21 +3013,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = 0x00, - .endpoint = 0x01, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_OUT, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .sync_ep = USB_RECIP_INTERFACE | USB_DIR_IN, + .sync_iface = 3, + .sync_altsetting = 1, + .sync_ep_idx = 1, + .implicit_fb = 1, } }, { @@ -3035,22 +3042,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 3, .altsetting = 1, .altset_idx = 1, - .endpoint = 0x81, .attributes = 0x00, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_IN, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x009c, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e61c606..279b80874f6a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -984,21 +984,13 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } -static void mbox3_setup_48_24_magic(struct usb_device *dev) +static void mbox3_setup_defaults(struct usb_device *dev) { /* The Mbox 3 is "little endian" */ /* max volume is: 0x0000. */ /* min volume is: 0x0080 (shown in little endian form) */ - - /* Load 48000Hz rate into buffer */ - u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; - - /* Set 48000Hz sample rate */ - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + u8 com_buff[2]; /* Deactivate Tuner */ /* on = 0x01*/ @@ -1008,6 +1000,8 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); /* Set clock source to Internal (as opposed to S/PDIF) */ + /* Internal = 0x01*/ + /* S/PDIF = 0x02*/ com_buff[0] = 0x01; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, 0x21, 0x0100, 0x8001, &com_buff, 1); @@ -1113,9 +1107,11 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 1, 0x21, 0x0107, 0x4201, &com_buff, 2); /* Toggle allowing host control */ + /* Not needed com_buff[0] = 0x02; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + */ /* Do not dim fx returns */ com_buff[0] = 0x00; @@ -1259,26 +1255,27 @@ static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { - dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + dev_err(&dev->dev, "MBOX3: Invalid descriptor size=%d.\n", descriptor_size); return -ENODEV; } - dev_dbg(&dev->dev, "device initialised!\n"); + dev_dbg(&dev->dev, "MBOX3: device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; - if (err < 0) - dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + if (err < 0) + dev_dbg(&dev->dev, "MBOX3: error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); - if (err < 0) - dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); - dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + if (err < 0) + dev_dbg(&dev->dev, "MBOX3: error usb_reset_configuration: %d\n", err); + + dev_dbg(&dev->dev, "MBOX3: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); - mbox3_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + mbox3_setup_defaults(dev); + dev_info(&dev->dev, "MBOX3: Initialized."); return 0; /* Successful boot */ } @@ -1734,6 +1731,52 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, return 0; } +static void mbox3_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + // Set rate only for one interface + //u8 iface = subs->data_endpoint->iface; + //if (iface != 2) return; + + u8 buffer[4] = {0}; + u32 new_rate = subs->data_endpoint->cur_rate; + + // Get current rate from card and check if changing it is needed + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buffer, 4); + //u32 current_rate = le32_to_cpu(*(u32 *)buffer); + dev_dbg(&subs->dev->dev, + "MBOX3: Current configured sample rate: %d", le32_to_cpu(*(u32 *)buffer)); + if (le32_to_cpu(*(u32 *)buffer) == new_rate) { + dev_dbg(&subs->dev->dev, + "MBOX3: No change needed, current rate:%d == new rate:%d", + le32_to_cpu(*(u32 *)buffer), new_rate); + return; + } + + // Set new rate + dev_info(&subs->dev->dev, + "MBOX3: Changing sample rate to: %d", new_rate); + __le32 set_rate = cpu_to_le32(new_rate); + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &set_rate, 4); + + // Set clock source to Internal + buffer[0] = 0x01; + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8001, &buffer[0], 1); + + // Check whether the change was successful + buffer[0] = 0; buffer[1] = 0; buffer[2] = 0; buffer[3] = 0; + snd_usb_ctl_msg(subs->dev, + usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buffer, 4); + //set_rate = *(int *)buffer; + set_rate = le32_to_cpu(*(u32 *)buffer); + if (new_rate != set_rate) + dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate"); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1755,6 +1798,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; + case USB_ID(0x0dba, 0x5000): + mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */ + break; } } -- 2.44.0

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[PATCH v1] ALSA: ASoc/tas2781: fixed wrong loading calibrated data sequence
by Shenghao Ding 29 Apr '24

29 Apr '24

Calibrated data will be set to default after loading dsp config params, which will cause speaker protection work abnormally. Reload calibrated data after loading dsp config params. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v1: - Download calibrated data after loading the new dsp config params - call tasdevice_prmg_load instead of tasdevice_prmg_calibdata_load, it is unnecessary to load calibrated data after loading dsp program. Load it after loading dsp config params. - Remove tasdevice_prmg_calibdata_load, because it is unnecessary to load calibrated data after loading dsp program. - work in bypass-dsp mode even if no dsp fw loaded. --- include/sound/tas2781-dsp.h | 3 +- sound/soc/codecs/tas2781-fmwlib.c | 96 +++++++------------------------ sound/soc/codecs/tas2781-i2c.c | 4 +- 3 files changed, 23 insertions(+), 80 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..14a4a254ad2a 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe19523..c21bce6f09d6 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tasdevice-fmw.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -2210,21 +2210,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, for (i = 0; i < tas_priv->ndev; i++) { if (tas_priv->tasdevice[i].is_loaderr == true) continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration - *cal = cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) tas_priv->tasdevice[i].cur_prog = prm_no; - } } } @@ -2247,9 +2235,24 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, if (tas_priv->tasdevice[i].is_loaderr == true) { status |= 1 << (i + 4); continue; - } else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) + } + + if (tas_priv->tasdevice[i].is_loaderr == false + && tas_priv->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->tasdevice[i].cur_conf = cfg_no; + } } } else dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", @@ -2308,65 +2311,6 @@ int tasdevice_prmg_load(void *context, int prm_no) } EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB); -int tasdevice_prmg_calibdata_load(void *context, int prm_no) -{ - struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; - struct tasdevice_fw *tas_fmw = tas_priv->fmw; - struct tasdevice_prog *program; - int prog_status = 0; - int i; - - if (!tas_fmw) { - dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); - goto out; - } - - if (prm_no >= tas_fmw->nr_programs) { - dev_err(tas_priv->dev, - "%s: prm(%d) is not in range of Programs %u\n", - __func__, prm_no, tas_fmw->nr_programs); - goto out; - } - - for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) { - tas_priv->tasdevice[i].cur_conf = -1; - tas_priv->tasdevice[i].is_loading = true; - prog_status++; - } - tas_priv->tasdevice[i].is_loaderr = false; - } - - if (prog_status) { - program = &(tas_fmw->programs[prm_no]); - tasdevice_load_data(tas_priv, &(program->dev_data)); - for (i = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].is_loaderr == true) - continue; - else if (tas_priv->tasdevice[i].is_loaderr == false - && tas_priv->tasdevice[i].is_loading == true) { - struct tasdevice_fw *cal_fmw = - tas_priv->tasdevice[i].cali_data_fmw; - - if (cal_fmw) { - struct tasdevice_calibration *cal = - cal_fmw->calibrations; - - if (cal) - load_calib_data(tas_priv, - &(cal->dev_data)); - } - tas_priv->tasdevice[i].cur_prog = prm_no; - } - } - } - -out: - return prog_status; -} -EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load, - SND_SOC_TAS2781_FMWLIB); - void tasdevice_tuning_switch(void *context, int state) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b5abff230e43..9350972dfefe 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, __func__, tas_priv->cal_binaryname[i]); } - tasdevice_prmg_calibdata_load(tas_priv, 0); + tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { -- 2.34.1

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[PATCH] ASoC: intel: skylake: Add check for dma_set_max_seg_size
by Chen Ni 29 Apr '24

29 Apr '24

Add check for the return value of dma_set_max_seg_size() and return the error if it fails in order to catch the error. Fixes: c22a8086b384 ("ASoC: intel: skylake: Set max DMA segment size") Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/intel/skylake/skl.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 117125187793..8edc13795462 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -943,7 +943,9 @@ static int skl_first_init(struct hdac_bus *bus) /* allow 64bit DMA address if supported by H/W */ if (dma_set_mask_and_coherent(bus->dev, DMA_BIT_MASK(64))) dma_set_mask_and_coherent(bus->dev, DMA_BIT_MASK(32)); - dma_set_max_seg_size(bus->dev, UINT_MAX); + err = dma_set_max_seg_size(bus->dev, UINT_MAX); + if (err) + return err; /* initialize streams */ snd_hdac_ext_stream_init_all -- 2.25.1

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