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

[PATCH 2/2] ALSA: hda/realtek: Add quirks for Lenovo 16IAH7, 16IAX7 and 16ARHA7
by David Xu 13 Jul '23

13 Jul '23

Add quirks for Lenovo 16IAH7, 16IAX7 and 16ARHA7 that have dual cs35l41 amplifiers that drives their speakers. Please note that this patch depends on patch 1, for the added quirks to work for the specified models, proper CSC3551 configurtion fixups have to be applied. Signed-off-by: David Xu <xuwd1(a)hotmail.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2f8b608de82..cc10bb8b75d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9831,6 +9831,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3803, "Legion Slim 7i 16IAH7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), @@ -9846,6 +9847,10 @@ 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, 0x386e, "Legion Slim 7i 16IAH7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3874, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3877, "Legion 7 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), 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), -- 2.41.0

1 0

Re: [PATCH 2/4] ASoC: codecs: es8326: Fix power-up sequence
by Zhu Ning 13 Jul '23

13 Jul '23

> > Old power-up sequence causes large pop noise during start-up. > > Using a new sequence instead. > > > static const struct regmap_range es8326_volatile_ranges[] = { > > - regmap_reg_range(ES8326_HP_DETECT, ES8326_HP_DETECT), > > + regmap_reg_range(ES8326_HPDET_TYPE, ES8326_HPDET_TYPE), > > }; > > As well as not seeming to correspond to the patch description this seems > obviously buggy - even if _HPDET_TYPE was missed from the list of > volatile registers no software change should be able to make a volatile > register non-volatile, that's a property of the hardware. The changes > to read from HPDET_TYPE also seem unrelated to the main change here in > the resume function, this should probably be multiple commits. Just a naming space change, will create a new patch for it. > > static int es8326_resume(struct snd_soc_component *component) > > { > > struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); > > - unsigned int reg; > > > > regcache_cache_only(es8326->regmap, false); > > regcache_sync(es8326->regmap); > > > > + /* reset register value to default */ > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > + usleep_range(1000, 3000); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > This looks wrong, you're resyncing the cache and then start resetting > registers? It feels like the ordering is off here, and some of this > reset sequence might want to be done with the cache bypassed. Are you > sure that there's no corruption of user visible state resulting from the > power up sequence, especially around the HP driver? Basically the chip runs through the start-up sequence every time in the resume function. Will move the reset sequence to es8326_suspend

2 1

[PATCH v2 0/5] Few audio fixes on Tegra platforms
by Sameer Pujar 13 Jul '23

13 Jul '23

This series fixes some of the issues which were observed during an attempt to enhance automated test coverage on Jetson AGX Orin. Below is a short summary of the issues and fixes: * Sample rate coversion failures above 48kHz. * AMX and ADX test cases failures due to incorrect byte mask. * Atomic sleep in RT5640 codec which is present on Jetson AGX Orin. * AHUB clock fixes on Tegra234 and previous chips. * Minor cleanups in ASRC and AHUB driver. Changelog ========= v1 -> v2: --------- * Few patches got accepted in the original (v1) series. Now v2 addresses comments for remaining patches. * AMX/ADX byte map fix patch is updated with more details in the commit message and added TODO item in the driver to improve the logic. * For RT5640 codec patch, the threaded IRQ is used for only for rt5640_irq() and rt5640_jd_gpio_irq() is left untouched. Sameer Pujar (2): ASoC: rt5640: Fix sleep in atomic context arm64: tegra: Update AHUB clock parent and rate Sheetal (3): ASoC: tegra: Fix AMX byte map ASoC: tegra: Fix ADX byte map arm64: tegra: Update AHUB clock parent and rate on Tegra234 arch/arm64/boot/dts/nvidia/tegra186.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra210.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra234.dtsi | 3 ++- sound/soc/codecs/rt5640.c | 12 ++++++---- sound/soc/tegra/tegra210_adx.c | 34 +++++++++++++++++---------- sound/soc/tegra/tegra210_amx.c | 40 ++++++++++++++++++-------------- 7 files changed, 59 insertions(+), 39 deletions(-) -- 2.7.4

4 11

[PATCH 1/3] ALSA: emu10k1: fix return value of snd_emu1010_adc_pads_put()
by Oswald Buddenhagen 13 Jul '23

13 Jul '23

It returned zero even if the value had changed. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- this should have been in cc766807a2 (fix return value of snd_emu1010_dac_pads_put()), but, well. --- sound/pci/emu10k1/emumixer.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index f9500cd50a4b..573e1c7c5e50 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -770,18 +770,21 @@ static int snd_emu1010_adc_pads_put(struct snd_kcontrol *kcontrol, struct snd_ct struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); unsigned int mask = snd_emu1010_adc_pad_regs[kcontrol->private_value]; unsigned int val, cache; + int change; + val = ucontrol->value.integer.value[0]; cache = emu->emu1010.adc_pads; if (val == 1) cache = cache | mask; else cache = cache & ~mask; - if (cache != emu->emu1010.adc_pads) { + change = (cache != emu->emu1010.adc_pads); + if (change) { snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, cache ); emu->emu1010.adc_pads = cache; } - return 0; + return change; } static const struct snd_kcontrol_new emu1010_adc_pads_ctl = { -- 2.40.0.152.g15d061e6df

2 14

[PATCH] ALSA: pcmtest: Extend error injection, decrease buffer filling overhead
by Ivan Orlov 13 Jul '23

13 Jul '23

Extend 'pcmtest' virtual driver with 'open' callback error injection functionality, as it already can inject errors into other PCM callbacks. Fix the driver to use already defined variables where it is possible. Additionally, decrease the buffer filling overhead with conditional reminder calculation in the 'inc_buf_pos' inline function. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/drivers/pcmtest.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 291e7fe47893..08e14b5eb772 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -65,6 +65,7 @@ static int inject_delay; static bool inject_hwpars_err; static bool inject_prepare_err; static bool inject_trigger_err; +static bool inject_open_err; static short fill_mode = FILL_MODE_PAT; @@ -88,6 +89,9 @@ module_param(inject_prepare_err, bool, 0600); MODULE_PARM_DESC(inject_prepare_err, "Inject EINVAL error in the 'prepare' callback"); module_param(inject_trigger_err, bool, 0600); MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callback"); +module_param(inject_open_err, bool, 0600); +MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); + struct pcmtst { struct snd_pcm *pcm; @@ -140,7 +144,8 @@ static inline void inc_buf_pos(struct pcmtst_buf_iter *v_iter, size_t by, size_t { v_iter->total_bytes += by; v_iter->buf_pos += by; - v_iter->buf_pos %= bytes; + if (v_iter->buf_pos >= bytes) + v_iter->buf_pos %= bytes; } /* @@ -196,10 +201,10 @@ static void check_buf_block_ni(struct pcmtst_buf_iter *v_iter, struct snd_pcm_ru u8 current_byte; for (i = 0; i < v_iter->b_rw; i++) { - current_byte = runtime->dma_area[buf_pos_n(v_iter, channels, i % channels)]; + ch_num = i % channels; + current_byte = runtime->dma_area[buf_pos_n(v_iter, channels, ch_num)]; if (!current_byte) break; - ch_num = i % channels; if (current_byte != patt_bufs[ch_num].buf[(v_iter->total_bytes / channels) % patt_bufs[ch_num].len]) { v_iter->is_buf_corrupted = true; @@ -239,7 +244,7 @@ static void fill_block_pattern_n(struct pcmtst_buf_iter *v_iter, struct snd_pcm_ for (i = 0; i < v_iter->b_rw; i++) { ch_num = i % channels; - runtime->dma_area[buf_pos_n(v_iter, channels, i % channels)] = + runtime->dma_area[buf_pos_n(v_iter, channels, ch_num)] = patt_bufs[ch_num].buf[(v_iter->total_bytes / channels) % patt_bufs[ch_num].len]; inc_buf_pos(v_iter, 1, runtime->dma_bytes); @@ -364,6 +369,9 @@ static int snd_pcmtst_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct pcmtst_buf_iter *v_iter; + if (inject_open_err) + return -EBUSY; + v_iter = kzalloc(sizeof(*v_iter), GFP_KERNEL); if (!v_iter) return -ENOMEM; -- 2.34.1

2 2

[PATCH][next] selftests: ALSA: Fix fclose on an already fclosed file pointer
by Colin Ian King 13 Jul '23

13 Jul '23

In the case where a sysfs file cannot be opened the error return path fcloses file pointer fpl, however, fpl has already been closed in the previous stanza. Fix the double fclose by removing it. Fixes: 10b98a4db11a ("selftests: ALSA: Add test for the 'pcmtest' driver") Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- tools/testing/selftests/alsa/test-pcmtest-driver.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/tools/testing/selftests/alsa/test-pcmtest-driver.c b/tools/testing/selftests/alsa/test-pcmtest-driver.c index 71931b240a83..357adc722cba 100644 --- a/tools/testing/selftests/alsa/test-pcmtest-driver.c +++ b/tools/testing/selftests/alsa/test-pcmtest-driver.c @@ -47,10 +47,8 @@ static int read_patterns(void) sprintf(pf, "/sys/kernel/debug/pcmtest/fill_pattern%d", i); fp = fopen(pf, "r"); - if (!fp) { - fclose(fpl); + if (!fp) return -1; - } fread(patterns[i].buf, 1, patterns[i].len, fp); fclose(fp); } -- 2.39.2

3 2

[PATCH v4] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 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/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 80 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..b4f07bdcf8b4 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 */ @@ -305,6 +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 extra_jack_data; int ret; /* need to enable ASRC function for 24MHz mclk rate */ @@ -315,7 +318,16 @@ 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 +377,12 @@ 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_jack_data = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra_jack_data); + } 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 +419,8 @@ 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 +441,8 @@ 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 +452,9 @@ 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 +583,30 @@ 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 +657,17 @@ 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 +706,9 @@ 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 +861,11 @@ 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 +952,12 @@ 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 +1246,14 @@ 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..f56bd7d656e9 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.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 2

[PATCH 00/13] PCI: Define Intel PCI IDs and use them in drivers
by Amadeusz Sławiński 12 Jul '23

12 Jul '23

PCI IDs for Intel HDA are duplicated across quite a few drivers, due to various configurations and historical reasons. Currently almost all uses of HDA PCI IDs have corresponding comment telling which platform it is. Additionally there are some inconsistencies between drivers about which ID corresponds to which device. Simplify things, by adding PCI IDs to global header and make use of them in drivers. This allows for removal of comments by having IDs themselves being self explanatory. Additionally it allows for removal of existing inconsistencies by having one source of truth. Changes from RFC: - Sort Intel PCI IDs before adding new ones - Fix ordering of new PCI IDs (Andy) - Define all used Intel IDs (Andy) - Add macros for controller type detection (Andy/Bjorn) - Add set of patches changing to use above macro (Andy/Bjorn) - Use PCI_DEVICE_DATA for Intel IDs in sound/pci/hda/hda_intel.c (Andy) - Commit message wording (Andy) - Remove unnecessary tabs (Andy) Amadeusz Sławiński (13): PCI: Sort Intel PCI IDs by number PCI: Add Intel Audio DSP devices to pci_ids.h ALSA: hda: Add controller matching macros ALSA: hda: Use global PCI match macro ALSA: hda/i915: Use global PCI match macro ASoC: Intel: Skylake: Use global PCI match macro ALSA: intel-dsp-config: Convert to PCI device IDs defines ALSA: hda: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: Skylake: Convert to PCI device IDs defines ASoC: SOF: Intel: Convert to PCI device IDs defines ASoC: Intel: sst: Convert to PCI device IDs defines include/linux/pci_ids.h | 104 +++++-- include/sound/hda_codec.h | 3 - include/sound/hdaudio.h | 27 ++ sound/hda/hdac_i915.c | 7 +- sound/hda/intel-dsp-config.c | 119 ++++---- sound/pci/hda/hda_intel.c | 373 ++++++++++--------------- sound/soc/intel/atom/sst/sst.c | 3 +- sound/soc/intel/atom/sst/sst.h | 1 - sound/soc/intel/atom/sst/sst_pci.c | 4 +- sound/soc/intel/avs/board_selection.c | 10 +- sound/soc/intel/avs/core.c | 16 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl.c | 36 +-- sound/soc/sof/intel/pci-apl.c | 9 +- sound/soc/sof/intel/pci-cnl.c | 15 +- sound/soc/sof/intel/pci-icl.c | 12 +- sound/soc/sof/intel/pci-mtl.c | 3 +- sound/soc/sof/intel/pci-skl.c | 6 +- sound/soc/sof/intel/pci-tgl.c | 45 +-- sound/soc/sof/intel/pci-tng.c | 3 +- 21 files changed, 384 insertions(+), 431 deletions(-) -- 2.34.1

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[PATCH v2 0/4] ASoC: rt5677: Refactor GPIO and use device_get_match_data()
by Andy Shevchenko 12 Jul '23

12 Jul '23

The code can be simplified with refactored GPIO parts and with use of device_get_match_data(). Besides that couple of additional changes, one for maintenance and one for making IRQ domain agnostic (not being pinned to OF). Changelog v2: - refactored GPIO code in (a new) patch 1 - fixed compilation error in patch 2 (LKP) Andy Shevchenko (4): ASoC: rt5677: Refactor GPIO support code ASoC: rt5677: Use agnostic irq_domain_create_linear() ASoC: rt5677: Use device_get_match_data() ASoC: rt5677: Sort headers alphabetically sound/soc/codecs/rt5677.c | 117 +++++++++++++------------------------- sound/soc/codecs/rt5677.h | 92 +++++------------------------- 2 files changed, 53 insertions(+), 156 deletions(-) -- 2.40.0.1.gaa8946217a0b

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[PATCH] ASoC: tlv320aic32x4: Fix the divide by zero
by Guiting Shen 12 Jul '23

12 Jul '23

The value of register(NDAC,MDAC,NADC,MADC,BCLKN) maybe zero lead to divide by zero in clk_aic32x4_div_recalc_rate().And the rate should be divide by 128 if the value was zero in this function according to the datasheet. Add the macro AIC32X4_DIV_MAX to present the 128 and return 0 if failing to read the value of register. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4-clk.c | 16 +++++++++++----- sound/soc/codecs/tlv320aic32x4.h | 5 +++-- 2 files changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index c116e82f712d..5c0a76a4a106 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -321,7 +321,7 @@ static int clk_aic32x4_div_set_rate(struct clk_hw *hw, unsigned long rate, u8 divisor; divisor = DIV_ROUND_UP(parent_rate, rate); - if (divisor > 128) + if (divisor > AIC32X4_DIV_MAX) return -EINVAL; return regmap_update_bits(div->regmap, div->reg, @@ -334,7 +334,7 @@ static int clk_aic32x4_div_determine_rate(struct clk_hw *hw, unsigned long divisor; divisor = DIV_ROUND_UP(req->best_parent_rate, req->rate); - if (divisor > 128) + if (divisor > AIC32X4_DIV_MAX) return -EINVAL; req->rate = DIV_ROUND_UP(req->best_parent_rate, divisor); @@ -345,12 +345,18 @@ static unsigned long clk_aic32x4_div_recalc_rate(struct clk_hw *hw, unsigned long parent_rate) { struct clk_aic32x4 *div = to_clk_aic32x4(hw); - unsigned int val; + int err; + + err = regmap_read(div->regmap, div->reg, &val); + if (err) + return 0; - regmap_read(div->regmap, div->reg, &val); + val &= AIC32X4_DIV_MASK; + if (!val) + val = AIC32X4_DIV_MAX; - return DIV_ROUND_UP(parent_rate, val & AIC32X4_DIV_MASK); + return DIV_ROUND_UP(parent_rate, val); } static const struct clk_ops aic32x4_div_ops = { diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 4de5bd9e8cc5..d6101ce73f80 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -223,8 +223,9 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_REFPOWERUP_120MS 0x07 /* Common mask and enable for all of the dividers */ -#define AIC32X4_DIVEN BIT(7) -#define AIC32X4_DIV_MASK GENMASK(6, 0) +#define AIC32X4_DIVEN BIT(7) +#define AIC32X4_DIV_MASK GENMASK(6, 0) +#define AIC32X4_DIV_MAX 128 /* Clock Limits */ #define AIC32X4_MAX_DOSR_FREQ 6200000 -- 2.25.1

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