July 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 1/3] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 10 Jul '23

10 Jul '23

Integrate tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one audio device. The code support realtek as the primary codec. Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of the certain one. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v2: - simplify the check of vendor id with Lenovo - ThinkPad is one of Lenovo's brands, they suggested me to use ALC269_FIXUP_THINKPAD_ACPI. - Add comments on ACARD_SINGLE_RANGE_EXT_TLV - Add the range check for tas_priv->tasdevice[] in tas2781_acpi_get_i2c_res. - remove acpi_subsystem_id - Issue in Laptop 0x17aa38be ACPI talbe caused codec->bus->pci->subsystem_device is not equal to (codec->core.subsystem_id & 0xffff) in snd_hda_pick_fixup. The former is 0x3802 and the latter is 0x38be leads to getting the wrong fixup_id and enter into the wrong entry. Although, this issue has been raised to the laptop manufacturer, but the ACPI table is locked, cannot be changed any more. Correct the wrong entry in the code. - Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of one certain codec. - Ignore the checkpatch complaints in alc269_fixup_tbl - Drop the hunk which is irrelevant with my code in alc_fixup_headset_mode_alc255_no_hp_mic - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. - delete the white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic --- sound/pci/hda/patch_realtek.c | 104 +++++++++++++++++++++++++++++++++- 1 file changed, 101 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2f8b608de82..d730497af57d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6705,7 +6705,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6714,7 +6714,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6730,12 +6730,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6763,6 +6783,41 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = 0; + spec->comps[0].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + case HDA_FIXUP_ACT_FREE: + component_master_del(dev, &comp_master_ops); + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6790,6 +6845,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7213,6 +7274,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9283,6 +9345,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9846,6 +9914,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -10761,6 +10843,22 @@ static int patch_alc269(struct hda_codec *codec) codec->fixup_id = HDA_FIXUP_ID_NOT_SET; } + /* FIXME: Issue in Laptop 0x17aa38be ACPI talbe caused + * codec->bus->pci->subsystem_device is not equal to + * (codec->core.subsystem_id & 0xffff) in snd_hda_pick_fixup. + * The former is 0x3802 and the latter is 0x38be leads to getting the + * wrong fixup_id and enter into the wrong entry. Although, this issue + * has been raised to the laptop manufacturer, but the ACPI table is + * locked, cannot be changed any more. Correct the wrong entry in the + * code. + */ + if (codec->fixup_id == ALC287_FIXUP_YOGA7_14ITL_SPEAKERS && + codec->core.vendor_id == 0x10ec0287 && + codec->core.subsystem_id == 0x17aa38be) { + codec_dbg(codec, "Clear wrong fixup for 17aa38be\n"); + codec->fixup_id = ALC287_FIXUP_TAS2781_I2C; + } + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, -- 2.34.1

2 4

[PATCH v1 0/8] spi: Header and core clean up and refactoring
by Andy Shevchenko 10 Jul '23

10 Jul '23

Various cleanups and refactorings of the SPI header and core parts united in a single series. Patches 1 & 2, 5 & 6 & 8 are dependent inside each group. No functional change intended. Andy Shevchenko (8): spi: Remove unneeded OF node NULL checks spi: Drop duplicate IDR allocation code in spi_register_controller() spi: Use sysfs_emit() to instead of s*printf() spi: Get rid of old SPI_MASTER_NO_.X and SPI_MASTER_MUST_.X spi: Sort headers alphabetically spi: Clean up headers spi: Fix spelling typos and acronyms capitalization spi: Use struct_size() helper drivers/spi/spi-at91-usart.c | 2 +- drivers/spi/spi-atmel.c | 2 +- drivers/spi/spi-bitbang-txrx.h | 16 +-- drivers/spi/spi-bitbang.c | 2 +- drivers/spi/spi-davinci.c | 2 +- drivers/spi/spi-fsl-lpspi.c | 2 +- drivers/spi/spi-gpio.c | 8 +- drivers/spi/spi-lp8841-rtc.c | 8 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-mt65xx.c | 2 +- drivers/spi/spi-pci1xxxx.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-rb4xx.c | 2 +- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-stm32.c | 2 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 102 ++++++++--------- include/linux/spi/spi.h | 188 ++++++++++++++++++-------------- 18 files changed, 183 insertions(+), 165 deletions(-) -- 2.40.0.1.gaa8946217a0b

2 18

[PATCH] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 10 Jul '23

10 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 81 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 2 files changed, 90 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..53451c1b67f8 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,7 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; - int ret; + int ret, extra; /* need to enable ASRC function for 24MHz mclk rate */ if ((sof_rt5682_quirk & SOF_RT5682_MCLK_EN) && @@ -315,7 +317,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +376,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +418,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +440,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +451,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +582,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +656,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +705,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +860,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +951,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1245,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..36ea2bacbe7d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

1 0

[PATCH v2] soundwire: amd: Fix a check for errors in probe()
by Dan Carpenter 10 Jul '23

10 Jul '23

This code has two problems: 1) The devm_ioremap() function returns NULL, not error pointers. 2) It's checking the wrong variable. ->mmio instead of ->acp_mmio. Fixes: a673a8dfc214 ("soundwire: amd: Add support for AMD Manager driver") Suggested-by: "Mukunda,Vijendar" <vijendar.mukunda(a)amd.com> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- v2: Change the variable. My original fix would have prevented the driver from loading so many thanks to Vijendar for pointing that out. drivers/soundwire/amd_manager.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 08aeb7ed00e1..3a99f6dcdfaf 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -910,9 +910,9 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return -ENOMEM; amd_manager->acp_mmio = devm_ioremap(dev, res->start, resource_size(res)); - if (IS_ERR(amd_manager->mmio)) { + if (!amd_manager->acp_mmio) { dev_err(dev, "mmio not found\n"); - return PTR_ERR(amd_manager->mmio); + return -ENOMEM; } amd_manager->instance = pdata->instance; amd_manager->mmio = amd_manager->acp_mmio + -- 2.39.2

2 1

[PATCH v2] sound: seq: fix typo in system_2p_ev_to_ump_midi1()
by Minjie Du 10 Jul '23

10 Jul '23

Fix data->system.parm2 typo. Signed-off-by: Minjie Du <duminjie(a)vivo.com> --- sound/core/seq/seq_ump_convert.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index eb1d86ff6..7cc84e137 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -714,7 +714,7 @@ static int system_2p_ev_to_ump_midi1(const struct snd_seq_event *event, { data->system.status = status; data->system.parm1 = (event->data.control.value >> 7) & 0x7f; - data->system.parm1 = event->data.control.value & 0x7f; + data->system.parm2 = event->data.control.value & 0x7f; return 1; } -- 2.39.0

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[PATCH] ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR
by Srinivas Kandagatla 10 Jul '23

10 Jul '23

dB range for HPHL and HPHR gains are from +6dB to -30dB in steps of 1.5dB with register values range from 0 to 24. Current code maps these dB ranges incorrectly, fix them to allow proper volume setting. Fixes: e8ba1e05bdc0("ASoC: codecs: wcd938x: add basic controls") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd938x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index faa15a5ed2c8..3a3360711f8f 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -210,7 +210,7 @@ struct wcd938x_priv { }; static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(ear_pa_gain, 600, -1800); -static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(line_gain, 600, -3000); +static const DECLARE_TLV_DB_SCALE(line_gain, -3000, 150, -3000); static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(analog_gain, 0, 3000); struct wcd938x_mbhc_zdet_param { @@ -2655,8 +2655,8 @@ static const struct snd_kcontrol_new wcd938x_snd_controls[] = { wcd938x_get_swr_port, wcd938x_set_swr_port), SOC_SINGLE_EXT("DSD_R Switch", WCD938X_DSD_R, 0, 1, 0, wcd938x_get_swr_port, wcd938x_set_swr_port), - SOC_SINGLE_TLV("HPHL Volume", WCD938X_HPH_L_EN, 0, 0x18, 0, line_gain), - SOC_SINGLE_TLV("HPHR Volume", WCD938X_HPH_R_EN, 0, 0x18, 0, line_gain), + SOC_SINGLE_TLV("HPHL Volume", WCD938X_HPH_L_EN, 0, 0x18, 1, line_gain), + SOC_SINGLE_TLV("HPHR Volume", WCD938X_HPH_R_EN, 0, 0x18, 1, line_gain), WCD938X_EAR_PA_GAIN_TLV("EAR_PA Volume", WCD938X_ANA_EAR_COMPANDER_CTL, 2, 0x10, 0, ear_pa_gain), SOC_SINGLE_EXT("ADC1 Switch", WCD938X_ADC1, 1, 1, 0, -- 2.25.1

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[PATCH v2 0/4] soundwire: qcom: stablity fixes
by Srinivas Kandagatla 10 Jul '23

10 Jul '23

During x13s audio testing we hit few corner cases due to issues in codec drivers and some obvious code bugs. Here are the fixes for those issues, mostly the issues are around devices loosing the sync in between runtime pm suspend resume path. With codec fixes along with these fixes, audio on x13s is pretty stable. Thanks, Srini Changes since v1: - dropped runtime pm changes patch as unable to reproduced it anymore - fixed clk stop flag as suggested by Pierre - rebased on top of linux-next Srinivas Kandagatla (4): soundwire: qcom: update status correctly with mask soundwire: qcom: wait for fifo to be empty before suspend soundwire: qcom: add software workaround for bus clash interrupt assertion soundwire: qcom: set clk stop need reset flag at runtime drivers/soundwire/qcom.c | 100 ++++++++++++++++++++++++++------------- 1 file changed, 68 insertions(+), 32 deletions(-) -- 2.21.0

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on (not) breaking user space
by Oswald Buddenhagen 09 Jul '23

09 Jul '23

hi linus, the alsa maintainers seem to be applying a reading of "we don't break user space" that i find counter-productive. so i would appreciate you clarifying your position on that matter (once more ...) to be clear, we're not arguing over changing any abi, but simply exercising one somewhat more than before. the claim is that this would likely have unacceptable consequences. i disagree. context: > Due to hardware constraints, changing the clock multiplier (CM) > changes the available audio ports and the number of available > channels. [...] Therefore, we do dynamic reconfiguration (DR) of the > mixer in response to changing the CM. arguments: > DR is somewhat controversial, as it has the potential to crash poorly > programmed applications. > But that in itself isn't a very convincing argument against it, as by > that logic we'd have to ban all hot-plugging. (1) > Such crashes would also not really qualify as regressions, as the > D.A.S. mode is a new opt-in feature, and therefore no previously > stable setups would be impacted. (2) the full text is at https://lore.kernel.org/all/20230630144542.664190-7-oswald.buddenhagen@gmx.… the preceding discussion is at https://lore.kernel.org/all/87v8fren1k.wl-tiwai@suse.de/T/#u regards, ossi

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[Feature Request] New hdajackretask ncurses TUI
by GitHub issues - opened 09 Jul '23

09 Jul '23

alsa-project/alsa-tools issue #16 was opened from HalanoSiblee: Hi Why i need gtk GUI when I do such a simple retaskable jack input Please make hdajackretask with ncurses like alsamixer to work in tty and terminal's at this point it will be so useful less bloat ,and blazingly fast Thanks. Issue URL : https://github.com/alsa-project/alsa-tools/issues/16 Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH 1/2] ALSA: pcmtest: Convert to platform remove callback returning void
by Uwe Kleine-König 07 Jul '23

07 Jul '23

The .remove() callback for a platform driver returns an int which makes many driver authors wrongly assume it's possible to do error handling by returning an error code. However the value returned is ignored (apart from emitting a warning) and this typically results in resource leaks. To improve here there is a quest to make the remove callback return void. In the first step of this quest all drivers are converted to .remove_new() which already returns void. Eventually after all drivers are converted, .remove_new() is renamed to .remove(). Trivially convert this driver from always returning zero in the remove callback to the void returning variant. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/drivers/pcmtest.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 2ae912a64d16..1fff85feaf82 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -576,10 +576,9 @@ static int pcmtst_probe(struct platform_device *pdev) return 0; } -static int pdev_remove(struct platform_device *dev) +static void pdev_remove(struct platform_device *dev) { snd_pcmtst_free(pcmtst); - return 0; } static struct platform_device pcmtst_pdev = { @@ -589,7 +588,7 @@ static struct platform_device pcmtst_pdev = { static struct platform_driver pcmtst_pdrv = { .probe = pcmtst_probe, - .remove = pdev_remove, + .remove_new = pdev_remove, .driver = { .name = "pcmtest", }, base-commit: 5133c9e51de41bfa902153888e11add3342ede18 -- 2.39.2

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