From 0c6498a59fbbcbf3d0a58c282dd6f0bca0eed92a Mon Sep 17 00:00:00 2001 From: Matus Malych Date: Sun, 12 Nov 2023 17:54:04 +0100 Subject: [PATCH 01/82] ASoC: amd: yc: Add HP 255 G10 into quirk table HP 255 G10's internal microphone array can be made to work by adding it to the quirk table. Signed-off-by: Matus Malych Link: https://lore.kernel.org/r/20231112165403.3221-1-matus@malych.org Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 15a864dcd7bd..e2a510443bf1 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -367,6 +367,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8A3E"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8B2F"), + } + }, { .driver_data = &acp6x_card, .matches = { From 37e6fd0cebf0b9f71afb38fd95b10408799d1f0b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 13 Nov 2023 15:59:16 +0000 Subject: [PATCH 02/82] ASoC: wm8974: Correct boost mixer inputs Bit 6 of INPPGA (INPPGAMUTE) does not control the Aux path, it controls the input PGA path, as can been seen from Figure 8 Input Boost Stage in the datasheet. Update the naming of things in the driver to match this and update the routing to also reflect this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20231113155916.1741027-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 044b6f604c09..260bac695b20 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0), /* Boost mixer */ static const struct snd_kcontrol_new wm8974_boost_mixer[] = { -SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1), +SOC_DAPM_SINGLE("PGA Switch", WM8974_INPPGA, 6, 1, 1), }; /* Input PGA */ @@ -246,8 +246,8 @@ static const struct snd_soc_dapm_route wm8974_dapm_routes[] = { /* Boost Mixer */ {"ADC", NULL, "Boost Mixer"}, - {"Boost Mixer", "Aux Switch", "Aux Input"}, - {"Boost Mixer", NULL, "Input PGA"}, + {"Boost Mixer", NULL, "Aux Input"}, + {"Boost Mixer", "PGA Switch", "Input PGA"}, {"Boost Mixer", NULL, "MICP"}, /* Input PGA */ From d5c65be34df73fa01ed05611aafb73b440d89e29 Mon Sep 17 00:00:00 2001 From: Kamil Duljas Date: Thu, 16 Nov 2023 13:51:50 +0100 Subject: [PATCH 03/82] ASoC: Intel: Skylake: Fix mem leak in few functions MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The resources should be freed when function return error. Signed-off-by: Kamil Duljas Reviewed-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20231116125150.1436-1-kamil.duljas@gmail.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-pcm.c | 4 +++- sound/soc/intel/skylake/skl-sst-ipc.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index d0c02e8a6785..18866bc415a5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -240,8 +240,10 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream); mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - if (!mconfig) + if (!mconfig) { + kfree(dma_params); return -EINVAL; + } skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 7a425271b08b..fd9624ad5f72 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -1003,8 +1003,10 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK; buf = krealloc(reply.data, reply.size, GFP_KERNEL); - if (!buf) + if (!buf) { + kfree(reply.data); return -ENOMEM; + } *payload = buf; *bytes = reply.size; From c1501f2597dd08601acd42256a4b0a0fc36bf302 Mon Sep 17 00:00:00 2001 From: David Lin Date: Fri, 17 Nov 2023 12:30:12 +0800 Subject: [PATCH 04/82] ASoC: nau8822: Fix incorrect type in assignment and cast to restricted __be16 This issue is reproduced when W=1 build in compiler gcc-12. The following are sparse warnings: sound/soc/codecs/nau8822.c:199:25: sparse: sparse: incorrect type in assignment sound/soc/codecs/nau8822.c:199:25: sparse: expected unsigned short sound/soc/codecs/nau8822.c:199:25: sparse: got restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16 Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202311122320.T1opZVkP-lkp@intel.com/ Signed-off-by: David Lin Link: https://lore.kernel.org/r/20231117043011.1747594-1-CTLIN0@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8822.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index ff3024899f45..7199d734c79f 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -184,6 +184,7 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *params = (void *)kcontrol->private_value; int i, reg; u16 reg_val, *val; + __be16 tmp; val = (u16 *)ucontrol->value.bytes.data; reg = NAU8822_REG_EQ1; @@ -192,8 +193,8 @@ static int nau8822_eq_get(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - reg_val = cpu_to_be16(reg_val); - memcpy(val + i, ®_val, sizeof(reg_val)); + tmp = cpu_to_be16(reg_val); + memcpy(val + i, &tmp, sizeof(tmp)); } return 0; @@ -216,6 +217,7 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol, void *data; u16 *val, value; int i, reg, ret; + __be16 *tmp; data = kmemdup(ucontrol->value.bytes.data, params->max, GFP_KERNEL | GFP_DMA); @@ -228,7 +230,8 @@ static int nau8822_eq_put(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - value = be16_to_cpu(*(val + i)); + tmp = (__be16 *)(val + i); + value = be16_to_cpup(tmp); ret = snd_soc_component_write(component, reg + i, value); if (ret) { dev_err(component->dev, From 31e721fbd194d5723722eaa21df1d14cee7e12b5 Mon Sep 17 00:00:00 2001 From: Kamil Duljas Date: Thu, 16 Nov 2023 22:39:17 +0100 Subject: [PATCH 05/82] ASoC: SOF: topology: Fix mem leak in sof_dai_load() The function has multiple return points at which it is not released previously allocated memory. Signed-off-by: Kamil Duljas Acked-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231116213926.2034-2-kamil.duljas@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index a3a3af252259..37ec671a2d76 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1736,8 +1736,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, /* perform pcm set op */ if (ipc_pcm_ops && ipc_pcm_ops->pcm_setup) { ret = ipc_pcm_ops->pcm_setup(sdev, spcm); - if (ret < 0) + if (ret < 0) { + kfree(spcm); return ret; + } } dai_drv->dobj.private = spcm; From f8ba14b780273fd290ddf7ee0d7d7decb44cc365 Mon Sep 17 00:00:00 2001 From: Kamil Duljas Date: Thu, 16 Nov 2023 23:41:13 +0100 Subject: [PATCH 06/82] ASoC: Intel: Skylake: mem leak in skl register function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit skl_platform_register() uses krealloc. When krealloc is fail, then previous memory is not freed. The leak is also when soc component registration failed. Signed-off-by: Kamil Duljas Reviewed-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20231116224112.2209-2-kamil.duljas@gmail.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 18866bc415a5..174aae6e0398 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1464,6 +1464,7 @@ int skl_platform_register(struct device *dev) dais = krealloc(skl->dais, sizeof(skl_fe_dai) + sizeof(skl_platform_dai), GFP_KERNEL); if (!dais) { + kfree(skl->dais); ret = -ENOMEM; goto err; } @@ -1476,8 +1477,10 @@ int skl_platform_register(struct device *dev) ret = devm_snd_soc_register_component(dev, &skl_component, skl->dais, num_dais); - if (ret) + if (ret) { + kfree(skl->dais); dev_err(dev, "soc component registration failed %d\n", ret); + } err: return ret; } From e7f289a59e76a5890a57bc27b198f69f175f75d9 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 17 Nov 2023 14:13:38 +0000 Subject: [PATCH 07/82] ASoC: cs43130: Fix the position of const qualifier Signed-off-by: Maciej Strozek Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231117141344.64320-2-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 0b40fdfb1825..20f06679b8f7 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1682,7 +1682,7 @@ static ssize_t hpload_dc_r_show(struct device *dev, return cs43130_show_dc(dev, buf, HP_RIGHT); } -static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = { +static const u16 cs43130_ac_freq[CS43130_AC_FREQ] = { 24, 43, 93, @@ -2362,7 +2362,7 @@ static const struct regmap_config cs43130_regmap = { .use_single_write = true, }; -static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { +static const u16 cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { 50, 120, }; From aa7e8e5e4011571022dc06e4d7a2f108feb53d1a Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 17 Nov 2023 14:13:39 +0000 Subject: [PATCH 08/82] ASoC: cs43130: Fix incorrect frame delay configuration Signed-off-by: Maciej Strozek Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231117141344.64320-3-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 20f06679b8f7..d8ec325b9cc9 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -578,7 +578,7 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, break; case SND_SOC_DAIFMT_LEFT_J: hi_size = bitwidth_sclk; - frm_delay = 2; + frm_delay = 0; frm_phase = 1; break; case SND_SOC_DAIFMT_DSP_A: From 14e8442e0789598514f3c9de014950de9feda7a4 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 20 Nov 2023 18:05:35 +0800 Subject: [PATCH 09/82] ASoC: fsl_sai: Fix no frame sync clock issue on i.MX8MP On i.MX8MP, when the TERE and FSD_MSTR enabled before configuring the word width, there will be no frame sync clock issue, because old word width impact the generation of frame sync. TERE enabled earlier only for i.MX8MP case for the hardware limitation, So need to disable FSD_MSTR before configuring word width, then enable FSD_MSTR bit for this specific case. Fixes: 3e4a82612998 ("ASoC: fsl_sai: MCLK bind with TX/RX enable bit") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1700474735-3863-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 79e7c6b98a75..32bbe5056a63 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -673,6 +673,20 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE((dl_cfg[dl_cfg_idx].mask[tx] & trce_mask))); + /* + * When the TERE and FSD_MSTR enabled before configuring the word width + * There will be no frame sync clock issue, because word width impact + * the generation of frame sync clock. + * + * TERE enabled earlier only for i.MX8MP case for the hardware limitation, + * We need to disable FSD_MSTR before configuring word width, then enable + * FSD_MSTR bit for this specific case. + */ + if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output && + !sai->is_consumer_mode) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FSD_MSTR, 0); + 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, @@ -680,6 +694,13 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); + + /* Enable FSD_MSTR after configuring word width */ + if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output && + !sai->is_consumer_mode) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FSD_MSTR, FSL_SAI_CR4_FSD_MSTR); + regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << min(channels, slots)) - 1)); From c33fd110424dfcb544cf55a1b312f43fe1918235 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 22 Nov 2023 09:42:53 +0800 Subject: [PATCH 10/82] ASoC: fsl_xcvr: Enable 2 * TX bit clock for spdif only case The bit 10 in TX_DPTH_CTRL register controls the TX clock rate. If this bit is set, TX datapath clock should be = 2* TX bit rate. If this bit is not set, TX datapath clock should be 10* TX bit rate. As the spdif only case, we always use 2 * TX bit clock, so this bit need to be set. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1700617373-6472-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index fa0a15263c66..77f8e2394bf9 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -414,6 +414,16 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, switch (xcvr->mode) { case FSL_XCVR_MODE_SPDIF: + if (xcvr->soc_data->spdif_only && tx) { + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_TX_DPTH_CTRL_SET, + FSL_XCVR_TX_DPTH_CTRL_BYPASS_FEM, + FSL_XCVR_TX_DPTH_CTRL_BYPASS_FEM); + if (ret < 0) { + dev_err(dai->dev, "Failed to set bypass fem: %d\n", ret); + return ret; + } + } + fallthrough; case FSL_XCVR_MODE_ARC: if (tx) { ret = fsl_xcvr_en_aud_pll(xcvr, fout); From 732c678eb021dbc514a699be1815e194692fdd5c Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 21 Nov 2023 15:44:19 +0000 Subject: [PATCH 11/82] ALSA: hda: cs35l56: Enable low-power hibernation mode on SPI SPI hibernation is now supported with the latest hibernation/wake sequences in the shared ASoC code. This has a functional dependency on two commits: commit 3df761bdbc8b ("ASoC: cs35l56: Wake transactions need to be issued twice") commit a47cf4dac7dc ("ASoC: cs35l56: Change hibernate sequence to use allow auto hibernate") To protect against this, enabling hibernation is conditional on CS35L56_WAKE_HOLD_TIME_US being defined, which indicates that the new hibernation sequences are available. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20231121154419.19435-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda_spi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index 756aec342eab..27d7fbc56b4c 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -21,6 +21,10 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi) return -ENOMEM; cs35l56->base.dev = &spi->dev; + +#ifdef CS35L56_WAKE_HOLD_TIME_US + cs35l56->base.can_hibernate = true; +#endif cs35l56->base.regmap = devm_regmap_init_spi(spi, &cs35l56_regmap_spi); if (IS_ERR(cs35l56->base.regmap)) { ret = PTR_ERR(cs35l56->base.regmap); From cdba4301adda7c60a2064bf808e48fccd352aaa9 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Wed, 22 Nov 2023 18:01:23 +0800 Subject: [PATCH 12/82] ASoC: rt5650: add mutex to avoid the jack detection failure This patch adds the jd_mutex to protect the jack detection control flow. And only the headset type could check the button status. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20231122100123.2831753-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7938b52d741d..a0d01d71d8b5 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -448,6 +448,7 @@ struct rt5645_priv { struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; struct rt5645_eq_param_s *eq_param; struct timer_list btn_check_timer; + struct mutex jd_mutex; int codec_type; int sysclk; @@ -3193,6 +3194,8 @@ static int rt5645_jack_detect(struct snd_soc_component *component, int jack_inse rt5645_enable_push_button_irq(component, true); } } else { + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(component, false); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; @@ -3295,6 +3298,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!rt5645->component) return; + mutex_lock(&rt5645->jd_mutex); + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { @@ -3321,7 +3326,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (!val && (rt5645->jack_type == 0)) { /* jack in */ report = rt5645_jack_detect(rt5645->component, 1); - } else if (!val && rt5645->jack_type != 0) { + } else if (!val && rt5645->jack_type == SND_JACK_HEADSET) { /* for push button and jack out */ btn_type = 0; if (snd_soc_component_read(rt5645->component, RT5645_INT_IRQ_ST) & 0x4) { @@ -3377,6 +3382,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) rt5645_jack_detect(rt5645->component, 0); } + mutex_unlock(&rt5645->jd_mutex); + snd_soc_jack_report(rt5645->hp_jack, report, SND_JACK_HEADPHONE); snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); if (rt5645->en_button_func) @@ -4150,6 +4157,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) } timer_setup(&rt5645->btn_check_timer, rt5645_btn_check_callback, 0); + mutex_init(&rt5645->jd_mutex); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); From 3841d8a563a7473ceb7415ecfe577e20b2a66d37 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Thu, 23 Nov 2023 10:18:15 +0100 Subject: [PATCH 13/82] ASoC: soc-pcm: fix up bad merge A recent change to address pops and clicks with codecs like WSA883X touched the same code paths as a fix for clearing DAI parameters and resulted in a bad merge. Specifically, commit f0220575e65a ("ASoC: soc-dai: add flag to mute and unmute stream during trigger") made mute at stream close conditional, while commit 3efcb471f871 ("ASoC: soc-pcm.c: Make sure DAI parameters cleared if the DAI becomes inactive") moved that same mute call back to soc_pcm_hw_clean(). Fix up the bad merge by dropping the second mute call from soc_pcm_clean() and making sure that the call in soc_pcm_hw_clean() is conditional as intended. Fixes: bdb7e1922052 ("ASoC: Merge up workaround for CODECs that play noise on stopped stream") Signed-off-by: Johan Hovold Link: https://lore.kernel.org/r/20231123091815.21933-1-johan+linaro@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 323e4d7b6adf..f6d1b2e11795 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -704,11 +704,6 @@ static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd, if (snd_soc_dai_active(dai) == 0 && (dai->rate || dai->channels || dai->sample_bits)) soc_pcm_set_dai_params(dai, NULL); - - if (snd_soc_dai_stream_active(dai, substream->stream) == 0) { - if (dai->driver->ops && !dai->driver->ops->mute_unmute_on_trigger) - snd_soc_dai_digital_mute(dai, 1, substream->stream); - } } } @@ -947,8 +942,10 @@ static int soc_pcm_hw_clean(struct snd_soc_pcm_runtime *rtd, if (snd_soc_dai_active(dai) == 1) soc_pcm_set_dai_params(dai, NULL); - if (snd_soc_dai_stream_active(dai, substream->stream) == 1) - snd_soc_dai_digital_mute(dai, 1, substream->stream); + if (snd_soc_dai_stream_active(dai, substream->stream) == 1) { + if (dai->driver->ops && !dai->driver->ops->mute_unmute_on_trigger) + snd_soc_dai_digital_mute(dai, 1, substream->stream); + } } /* run the stream event */ From 505c83212da5bfca95109421b8f5d9f8c6cdfef2 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Thu, 23 Nov 2023 09:44:54 +0100 Subject: [PATCH 14/82] ASoC: SOF: mediatek: mt8186: Add Google Steelix topology compatible Add the machine compatible and topology filename for the Google Steelix MT8186 Chromebook to load the correct SOF topology file. Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20231123084454.20471-1-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8186/mt8186.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index b69fa788b16f..e0d88e7aa8ca 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -597,6 +597,9 @@ static struct snd_sof_dsp_ops sof_mt8186_ops = { static struct snd_sof_of_mach sof_mt8186_machs[] = { { + .compatible = "google,steelix", + .sof_tplg_filename = "sof-mt8186-google-steelix.tplg" + }, { .compatible = "mediatek,mt8186", .sof_tplg_filename = "sof-mt8186.tplg", }, From 347ecf29a68cc8958fbcbd26ef410d07fe9d82f4 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 23 Nov 2023 09:14:53 +0800 Subject: [PATCH 15/82] ASoC: fsl_xcvr: refine the requested phy clock frequency As the input phy clock frequency will divided by 2 by default on i.MX8MP with the implementation of clk-imx8mp-audiomix driver, So the requested frequency need to be updated. The relation of phy clock is: sai_pll_ref_sel sai_pll sai_pll_bypass sai_pll_out sai_pll_out_div2 earc_phy_cg Signed-off-by: Shengjiu Wang Reviewed-by: Iuliana Prodan Link: https://lore.kernel.org/r/1700702093-8008-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_xcvr.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 77f8e2394bf9..f0fb33d719c2 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -358,7 +358,7 @@ static int fsl_xcvr_en_aud_pll(struct fsl_xcvr *xcvr, u32 freq) struct device *dev = &xcvr->pdev->dev; int ret; - freq = xcvr->soc_data->spdif_only ? freq / 10 : freq; + freq = xcvr->soc_data->spdif_only ? freq / 5 : freq; clk_disable_unprepare(xcvr->phy_clk); ret = clk_set_rate(xcvr->phy_clk, freq); if (ret < 0) { @@ -409,7 +409,7 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 m_ctl = 0, v_ctl = 0; u32 r = substream->runtime->rate, ch = substream->runtime->channels; - u32 fout = 32 * r * ch * 10 * 2; + u32 fout = 32 * r * ch * 10; int ret = 0; switch (xcvr->mode) { From 3d1dc8b1030df8ca0fdfd4905c88ee10db943bf8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 24 Nov 2023 14:40:15 +0200 Subject: [PATCH 16/82] ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available When the HDMI is not present due to disabled display support we will use dummy codec and the HDMI routes will refer to non existent DAPM widgets. Trim the route list from the HDMI routes to be able to probe the card even if the HDMI dais are not registered. Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231124124015.15878-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 6c6ef63cd5d9..6e172719c979 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -154,6 +154,8 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); if (!ctx->idisp_codec) { + card->dapm_routes = &skl_hda_map[IDISP_ROUTE_COUNT]; + num_route -= IDISP_ROUTE_COUNT; for (i = 0; i < IDISP_DAI_COUNT; i++) { skl_hda_be_dai_links[i].codecs = &snd_soc_dummy_dlc; skl_hda_be_dai_links[i].num_codecs = 1; From fba293488ccb1902e715da328e71aa868dd561f6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 24 Nov 2023 14:40:32 +0200 Subject: [PATCH 17/82] ASoC: Intel: sof_sdw: Always register the HDMI dai links The topology files for SDW devices require HDMI dai links to be present and this is granted under normal conditions but in case of special use cases the display (i915) driver might not be enabled due to deny-listing, booting with nomodeset or just not compiled at all. This should not block the non HDMI audio to be usable so register the dai links unconditionally. The code has been prepared for this and in case of no HDMI audio the link is created with dummy codec. Closes: https://github.com/thesofproject/linux/issues/4594 Closes: https://github.com/thesofproject/linux/issues/4648 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231124124032.15946-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3312ad8a563b..4e4284729773 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1546,7 +1546,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) { struct device *dev = card->dev; struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); - int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, hdmi_num = 0, bt_num = 0; + int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, bt_num = 0; struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; @@ -1564,6 +1564,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) char *codec_name, *codec_dai_name; int i, j, be_id = 0; int codec_index; + int hdmi_num; int ret; ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); @@ -1584,14 +1585,13 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) ssp_num = hweight_long(ssp_mask); } - if (mach_params->codec_mask & IDISP_CODEC_MASK) { + if (mach_params->codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) - hdmi_num = SOF_TGL_HDMI_COUNT; - else - hdmi_num = SOF_PRE_TGL_HDMI_COUNT; - } + if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) + hdmi_num = SOF_TGL_HDMI_COUNT; + else + hdmi_num = SOF_PRE_TGL_HDMI_COUNT; /* enable dmic01 & dmic16k */ if (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) @@ -1601,7 +1601,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) bt_num = 1; dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d, bt: %d\n", - sdw_be_num, ssp_num, dmic_num, hdmi_num, bt_num); + sdw_be_num, ssp_num, dmic_num, + ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num); /* allocate BE dailinks */ num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num; From 0376b995bb7a65fb0c056f3adc5e9695ad0c1805 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 24 Nov 2023 15:57:42 +0200 Subject: [PATCH 18/82] ASoC: SOF: ipc4-topology: Add core_mask in struct snd_sof_pipeline MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With IPC4, a pipeline may contain multiple modules in the data processing domain and they can be scheduled to run on different cores. Add a new field in struct snd_sof_pipeline to keep track of all the cores that are associated with the modules in the pipeline. Set the pipeline core mask for IPC3 when initializing the pipeline widget IPC structure. For IPC4, set the core mark when initializing the pipeline widget and initializing processing modules in the data processing domain. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231124135743.24674-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 2 ++ sound/soc/sof/ipc4-topology.c | 9 +++++++++ sound/soc/sof/sof-audio.h | 2 ++ 3 files changed, 13 insertions(+) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index ba4ef290b634..2c7a5e7a364c 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -493,6 +493,7 @@ static int sof_ipc3_widget_setup_comp_mixer(struct snd_sof_widget *swidget) static int sof_ipc3_widget_setup_comp_pipeline(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_pipeline *spipe = swidget->spipe; struct sof_ipc_pipe_new *pipeline; struct snd_sof_widget *comp_swidget; int ret; @@ -545,6 +546,7 @@ static int sof_ipc3_widget_setup_comp_pipeline(struct snd_sof_widget *swidget) swidget->dynamic_pipeline_widget); swidget->core = pipeline->core; + spipe->core_mask |= BIT(pipeline->core); return 0; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index b24a64377f68..19f36db30979 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -656,6 +656,7 @@ static int sof_ipc4_widget_setup_comp_pipeline(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; struct sof_ipc4_pipeline *pipeline; + struct snd_sof_pipeline *spipe = swidget->spipe; int ret; pipeline = kzalloc(sizeof(*pipeline), GFP_KERNEL); @@ -670,6 +671,7 @@ static int sof_ipc4_widget_setup_comp_pipeline(struct snd_sof_widget *swidget) } swidget->core = pipeline->core_id; + spipe->core_mask |= BIT(pipeline->core_id); if (pipeline->use_chain_dma) { dev_dbg(scomp->dev, "Set up chain DMA for %s\n", swidget->widget->name); @@ -797,6 +799,7 @@ static int sof_ipc4_widget_setup_comp_mixer(struct snd_sof_widget *swidget) static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; + struct snd_sof_pipeline *spipe = swidget->spipe; struct sof_ipc4_src *src; int ret; @@ -819,6 +822,8 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget) goto err; } + spipe->core_mask |= BIT(swidget->core); + dev_dbg(scomp->dev, "SRC sink rate %d\n", src->sink_rate); ret = sof_ipc4_widget_setup_msg(swidget, &src->msg); @@ -864,6 +869,7 @@ static int sof_ipc4_widget_setup_comp_process(struct snd_sof_widget *swidget) { struct snd_soc_component *scomp = swidget->scomp; struct sof_ipc4_fw_module *fw_module; + struct snd_sof_pipeline *spipe = swidget->spipe; struct sof_ipc4_process *process; void *cfg; int ret; @@ -920,6 +926,9 @@ static int sof_ipc4_widget_setup_comp_process(struct snd_sof_widget *swidget) sof_ipc4_widget_update_kcontrol_module_id(swidget); + /* set pipeline core mask to keep track of the core the module is scheduled to run on */ + spipe->core_mask |= BIT(swidget->core); + return 0; free_base_cfg_ext: kfree(process->base_config_ext); diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 5d5eeb1a1a6f..a6d6bcd00cee 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -480,6 +480,7 @@ struct snd_sof_widget { * @paused_count: Count of number of PCM's that have started and have currently paused this pipeline * @complete: flag used to indicate that pipeline set up is complete. + * @core_mask: Mask containing target cores for all modules in the pipeline * @list: List item in sdev pipeline_list */ struct snd_sof_pipeline { @@ -487,6 +488,7 @@ struct snd_sof_pipeline { int started_count; int paused_count; int complete; + unsigned long core_mask; struct list_head list; }; From 31ed8da1c8e5e504710bb36863700e3389f8fc81 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 24 Nov 2023 15:57:43 +0200 Subject: [PATCH 19/82] ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In the current code, we enable a widget core when it is set up and disable it when it is freed. This is problematic with IPC4 because widget free is essentially a NOP and all widgets are freed in the firmware when the pipeline is deleted. This results in a crash during pipeline deletion when one of it's widgets is scheduled to run on a secondary core and is powered off when widget is freed. So, change the logic to enable all cores needed by all the modules in a pipeline when the pipeline widget is set up and disable them after the pipeline widget is freed. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231124135743.24674-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 65 ++++++++++++++++++++++++--------------- 1 file changed, 41 insertions(+), 24 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 563fe6f7789f..77cc64ac7113 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -46,6 +46,7 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) { const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg); + struct snd_sof_pipeline *spipe = swidget->spipe; struct snd_sof_widget *pipe_widget; int err = 0; int ret; @@ -87,15 +88,22 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, } /* - * disable widget core. continue to route setup status and complete flag - * even if this fails and return the appropriate error + * decrement ref count for cores associated with all modules in the pipeline and clear + * the complete flag */ - ret = snd_sof_dsp_core_put(sdev, swidget->core); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to disable target core: %d for widget %s\n", - swidget->core, swidget->widget->name); - if (!err) - err = ret; + if (swidget->id == snd_soc_dapm_scheduler) { + int i; + + for_each_set_bit(i, &spipe->core_mask, sdev->num_cores) { + ret = snd_sof_dsp_core_put(sdev, i); + if (ret < 0) { + dev_err(sdev->dev, "failed to disable target core: %d for pipeline %s\n", + i, swidget->widget->name); + if (!err) + err = ret; + } + } + swidget->spipe->complete = 0; } /* @@ -108,10 +116,6 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, err = ret; } - /* clear pipeline complete */ - if (swidget->id == snd_soc_dapm_scheduler) - swidget->spipe->complete = 0; - if (!err) dev_dbg(sdev->dev, "widget %s freed\n", swidget->widget->name); @@ -134,8 +138,10 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) { const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg); + struct snd_sof_pipeline *spipe = swidget->spipe; bool use_count_decremented = false; int ret; + int i; /* skip if there is no private data */ if (!swidget->private) @@ -166,19 +172,23 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev, goto use_count_dec; } - /* enable widget core */ - ret = snd_sof_dsp_core_get(sdev, swidget->core); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to enable target core for widget %s\n", - swidget->widget->name); - goto pipe_widget_free; + /* update ref count for cores associated with all modules in the pipeline */ + if (swidget->id == snd_soc_dapm_scheduler) { + for_each_set_bit(i, &spipe->core_mask, sdev->num_cores) { + ret = snd_sof_dsp_core_get(sdev, i); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable target core %d for pipeline %s\n", + i, swidget->widget->name); + goto pipe_widget_free; + } + } } /* setup widget in the DSP */ if (tplg_ops && tplg_ops->widget_setup) { ret = tplg_ops->widget_setup(sdev, swidget); if (ret < 0) - goto core_put; + goto pipe_widget_free; } /* send config for DAI components */ @@ -208,15 +218,22 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev, return 0; widget_free: - /* widget use_count and core ref_count will both be decremented by sof_widget_free() */ + /* widget use_count will be decremented by sof_widget_free() */ sof_widget_free_unlocked(sdev, swidget); use_count_decremented = true; -core_put: - if (!use_count_decremented) - snd_sof_dsp_core_put(sdev, swidget->core); pipe_widget_free: - if (swidget->id != snd_soc_dapm_scheduler) + if (swidget->id != snd_soc_dapm_scheduler) { sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget); + } else { + int j; + + /* decrement ref count for all cores that were updated previously */ + for_each_set_bit(j, &spipe->core_mask, sdev->num_cores) { + if (j >= i) + break; + snd_sof_dsp_core_put(sdev, j); + } + } use_count_dec: if (!use_count_decremented) swidget->use_count--; From f83d38def6b1b00c9bb17173837045b41df7e7d7 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Sat, 25 Nov 2023 14:53:00 +0800 Subject: [PATCH 20/82] ASoC: imx-rpmsg: SND_SOC_IMX_RPMSG should depend on OF and I2C SND_SOC_IMX_RPMSG should depend on OF and I2C. It fixes the following error reported by kernel test robot: ld: sound/soc/fsl/imx-rpmsg.o: in function `imx_rpmsg_late_probe': imx-rpmsg.c:(.text+0x4f): undefined reference to `i2c_find_device_by_fwnode' Fixes: 5d9f746ca64c ("ASoC: imx-rpmsg: Force codec power on in low power audio mode") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202311230506.DPF9vvYY-lkp@intel.com/ Signed-off-by: Chancel Liu Link: https://lore.kernel.org/r/20231125065300.6385-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 725c530a3636..be342ee03fb9 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -360,6 +360,7 @@ config SND_SOC_IMX_HDMI config SND_SOC_IMX_RPMSG tristate "SoC Audio support for i.MX boards with rpmsg" depends on RPMSG + depends on OF && I2C select SND_SOC_IMX_PCM_RPMSG select SND_SOC_IMX_AUDIO_RPMSG help From 7b4c93a50a2ebbbaf656cc4fa6aca74a6166d85b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 27 Nov 2023 13:16:58 +0200 Subject: [PATCH 21/82] ALSA: hda: intel-nhlt: Ignore vbps when looking for DMIC 32 bps format When looking up DMIC blob from the NHLT table and the format is 32 bits, ignore the vbps matching for 32 bps for DMIC since some NHLT table have the vbps as 24, some have it as 32. The DMIC hardware supports only one type of 32 bit sample size, which is 24 bit sampling on the MSB side and bits[1:0] is used for indicating the channel number. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20231127111658.17275-1-peter.ujfalusi@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-nhlt.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 2c4dfc0b7e34..696a958d93e9 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -238,7 +238,7 @@ EXPORT_SYMBOL(intel_nhlt_ssp_mclk_mask); static struct nhlt_specific_cfg * nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch, - u32 rate, u8 vbps, u8 bps) + u32 rate, u8 vbps, u8 bps, bool ignore_vbps) { struct nhlt_fmt_cfg *cfg = fmt->fmt_config; struct wav_fmt *wfmt; @@ -255,8 +255,12 @@ nhlt_get_specific_cfg(struct device *dev, struct nhlt_fmt *fmt, u8 num_ch, dev_dbg(dev, "Endpoint format: ch=%d fmt=%d/%d rate=%d\n", wfmt->channels, _vbps, _bps, wfmt->samples_per_sec); + /* + * When looking for exact match of configuration ignore the vbps + * from NHLT table when ignore_vbps is true + */ if (wfmt->channels == num_ch && wfmt->samples_per_sec == rate && - vbps == _vbps && bps == _bps) + (ignore_vbps || vbps == _vbps) && bps == _bps) return &cfg->config; cfg = (struct nhlt_fmt_cfg *)(cfg->config.caps + cfg->config.size); @@ -289,6 +293,7 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, { struct nhlt_specific_cfg *cfg; struct nhlt_endpoint *epnt; + bool ignore_vbps = false; struct nhlt_fmt *fmt; int i; @@ -298,7 +303,26 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, dev_dbg(dev, "Looking for configuration:\n"); dev_dbg(dev, " vbus_id=%d link_type=%d dir=%d, dev_type=%d\n", bus_id, link_type, dir, dev_type); - dev_dbg(dev, " ch=%d fmt=%d/%d rate=%d\n", num_ch, vbps, bps, rate); + if (link_type == NHLT_LINK_DMIC && bps == 32 && (vbps == 24 || vbps == 32)) { + /* + * The DMIC hardware supports only one type of 32 bits sample + * size, which is 24 bit sampling on the MSB side and bits[1:0] + * are used for indicating the channel number. + * It has been observed that some NHLT tables have the vbps + * specified as 32 while some uses 24. + * The format these variations describe are identical, the + * hardware is configured and behaves the same way. + * Note: when the samples assumed to be vbps=32 then the 'noise' + * introduced by the lower two bits (channel number) have no + * real life implication on audio quality. + */ + dev_dbg(dev, + " ch=%d fmt=%d rate=%d (vbps is ignored for DMIC 32bit format)\n", + num_ch, bps, rate); + ignore_vbps = true; + } else { + dev_dbg(dev, " ch=%d fmt=%d/%d rate=%d\n", num_ch, vbps, bps, rate); + } dev_dbg(dev, "Endpoint count=%d\n", nhlt->endpoint_count); epnt = (struct nhlt_endpoint *)nhlt->desc; @@ -307,7 +331,8 @@ intel_nhlt_get_endpoint_blob(struct device *dev, struct nhlt_acpi_table *nhlt, if (nhlt_check_ep_match(dev, epnt, bus_id, link_type, dir, dev_type)) { fmt = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); - cfg = nhlt_get_specific_cfg(dev, fmt, num_ch, rate, vbps, bps); + cfg = nhlt_get_specific_cfg(dev, fmt, num_ch, rate, + vbps, bps, ignore_vbps); if (cfg) return cfg; } From baaacbff64d9f34b64f294431966d035aeadb81c Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 25 Oct 2023 15:24:06 +0800 Subject: [PATCH 22/82] ALSA: hda/realtek: Headset Mic VREF to 100% This platform need to set Mic VREF to 100%. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/0916af40f08a4348a3298a9a59e6967e@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 079876b7b3e7..0021f0f145a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1987,6 +1987,7 @@ enum { ALC887_FIXUP_ASUS_AUDIO, ALC887_FIXUP_ASUS_HMIC, ALCS1200A_FIXUP_MIC_VREF, + ALC888VD_FIXUP_MIC_100VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2540,6 +2541,13 @@ static const struct hda_fixup alc882_fixups[] = { {} } }, + [ALC888VD_FIXUP_MIC_100VREF] = { + .type = HDA_FIXUP_PINCTLS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, PIN_VREF100 }, /* headset mic */ + {} + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2609,6 +2617,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x10ec, 0x12d8, "iBase Elo Touch", ALC888VD_FIXUP_MIC_100VREF), SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), From cae2bdb579ecc9d4219c58a7d3fde1958118dc1d Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 29 Nov 2023 15:38:40 +0800 Subject: [PATCH 23/82] ALSA: hda/realtek: Add supported ALC257 for ChromeOS ChromeOS want to support ALC257. Add codec ID to some relation function. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/99a88a7dbdb045fd9d934abeb6cec15f@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0021f0f145a9..f9ddacfd920e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3265,6 +3265,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); @@ -3294,6 +3295,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); @@ -6505,6 +6507,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); From a337c355719c42a6c5b67e985ad753590ed844fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Nov 2023 16:13:21 +0100 Subject: [PATCH 24/82] ALSA: hda: Disable power-save on KONTRON SinglePC It's been reported that the runtime PM on KONTRON SinglePC (PCI SSID 1734:1232) caused a stall of playback after a bunch of invocations. (FWIW, this looks like an timing issue, and the stall happens rather on the controller side.) As a workaround, disable the default power-save on this platform. Cc: Link: https://lore.kernel.org/r/20231130151321.9813-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index db90feb49c16..2d1df3654424 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2242,6 +2242,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* KONTRON SinglePC may cause a stall at runtime resume */ + SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0), {} }; #endif /* CONFIG_PM */ From 19650c0f402f53abe48a55a1c49c8ed9576a088c Mon Sep 17 00:00:00 2001 From: Jeremy Soller Date: Mon, 27 Nov 2023 11:42:38 -0700 Subject: [PATCH 25/82] ASoC: amd: yc: Add DMI entry to support System76 Pangolin 13 Add pang13 quirk to enable the internal microphone. Signed-off-by: Jeremy Soller Signed-off-by: Tim Crawford Link: https://lore.kernel.org/r/20231127184237.32077-2-tcrawford@system76.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e2a510443bf1..c425652b0fad 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -388,6 +388,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_VERSION, "pang12"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "System76"), + DMI_MATCH(DMI_PRODUCT_VERSION, "pang13"), + } + }, {} }; From b24e3590c94ab0aba6e455996b502a83baa5c31c Mon Sep 17 00:00:00 2001 From: Malcolm Hart Date: Mon, 27 Nov 2023 20:36:00 +0000 Subject: [PATCH 26/82] ASoC: amd: yc: Fix non-functional mic on ASUS E1504FA This patch adds ASUSTeK COMPUTER INC "E1504FA" to the quirks file acp6x-mach.c to enable microphone array on ASUS Vivobook GO 15. I have this laptop and can confirm that the patch succeeds in enabling the microphone array. Signed-off-by: Malcolm Hart Cc: stable@vger.kernel.org Rule: add Link: https://lore.kernel.org/stable/875y1nt1bx.fsf%405harts.com Link: https://lore.kernel.org/r/871qcbszh0.fsf@5harts.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index c425652b0fad..d83cb6e4c62a 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -283,6 +283,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "E1504FA"), + } + }, { .driver_data = &acp6x_card, .matches = { From a2f35ed1d237c459100adb0c39bb811d7f170977 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Thu, 16 Nov 2023 17:44:21 +0100 Subject: [PATCH 27/82] ASoC: codecs: lpass-tx-macro: set active_decimator correct default value The -1 value for active_decimator[dai_id] is considered as "not set", but at probe the table is initialized a 0, this prevents enabling the DEC0 Mixer since it will be considered as already set. Initialize the table entries as -1 to fix tx_macro_tx_mixer_put(). Fixes: 1c6a7f5250ce ("ASoC: codecs: tx-macro: fix active_decimator array") Fixes: c1057a08af43 ("ASoC: codecs: tx-macro: fix kcontrol put") Signed-off-by: Neil Armstrong Link: https://lore.kernel.org/r/20231116-topic-sm8x50-upstream-tx-macro-fix-active-decimator-set-v1-1-6edf402f4b6f@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 82f9873ffada..124c2e144f33 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -2021,6 +2021,11 @@ static int tx_macro_probe(struct platform_device *pdev) tx->dev = dev; + /* Set active_decimator default value */ + tx->active_decimator[TX_MACRO_AIF1_CAP] = -1; + tx->active_decimator[TX_MACRO_AIF2_CAP] = -1; + tx->active_decimator[TX_MACRO_AIF3_CAP] = -1; + /* set MCLK and NPL rates */ clk_set_rate(tx->mclk, MCLK_FREQ); clk_set_rate(tx->npl, MCLK_FREQ); From a0575b4add21a243cc3257e75ad913cd5377d5f2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 28 Nov 2023 14:39:14 +0200 Subject: [PATCH 28/82] ASoC: hdac_hda: Conditionally register dais for HDMI and Analog The current driver is registering the same dais for each hdev found in the system which results duplicated widgets to be registered and the kernel log contains similar prints: snd_hda_codec_realtek ehdaudio0D0: ASoC: sink widget AIF1TX overwritten snd_hda_codec_realtek ehdaudio0D0: ASoC: source widget AIF1RX overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi2 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Codec Output Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Codec Input Pin1 overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Digital Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget Alt Analog Codec Playback overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Analog Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Digital Codec Capture overwritten skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: source widget Alt Analog Codec Capture overwritten To avoid such issue, split the dai array into HDMI and non HDMI array and register them conditionally: for HDMI hdev only register the dais needed for HDMI for non HDMI hdev do not register the HDMI dais. Depends-on: 3d1dc8b1030d ("ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available") Link: https://github.com/thesofproject/linux/issues/4509 Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231128123914.3986-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hda.c | 23 ++++++++++++++++++++--- 1 file changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 355f30779a34..b075689db2dc 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -132,6 +132,9 @@ static struct snd_soc_dai_driver hdac_hda_dais[] = { .sig_bits = 24, }, }, +}; + +static struct snd_soc_dai_driver hdac_hda_hdmi_dais[] = { { .id = HDAC_HDMI_0_DAI_ID, .name = "intel-hdmi-hifi1", @@ -607,8 +610,16 @@ static const struct snd_soc_component_driver hdac_hda_codec = { .endianness = 1, }; +static const struct snd_soc_component_driver hdac_hda_hdmi_codec = { + .probe = hdac_hda_codec_probe, + .remove = hdac_hda_codec_remove, + .idle_bias_on = false, + .endianness = 1, +}; + static int hdac_hda_dev_probe(struct hdac_device *hdev) { + struct hdac_hda_priv *hda_pvt = dev_get_drvdata(&hdev->dev); struct hdac_ext_link *hlink; int ret; @@ -621,9 +632,15 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) snd_hdac_ext_bus_link_get(hdev->bus, hlink); /* ASoC specific initialization */ - ret = devm_snd_soc_register_component(&hdev->dev, - &hdac_hda_codec, hdac_hda_dais, - ARRAY_SIZE(hdac_hda_dais)); + if (hda_pvt->need_display_power) + ret = devm_snd_soc_register_component(&hdev->dev, + &hdac_hda_hdmi_codec, hdac_hda_hdmi_dais, + ARRAY_SIZE(hdac_hda_hdmi_dais)); + else + ret = devm_snd_soc_register_component(&hdev->dev, + &hdac_hda_codec, hdac_hda_dais, + ARRAY_SIZE(hdac_hda_dais)); + if (ret < 0) { dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); return ret; From c447636970e3409ac39f0bb8c2dcff6b726f36b0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 15:14:10 +0200 Subject: [PATCH 29/82] ASoC: SOF: ipc4-topology: Correct data structures for the SRC module Separate the IPC message part as struct sof_ipc4_src_data. This struct describes the message payload passed to the firmware via the mailbox. It is not wise to be 'clever' and try to use the first part of a struct as IPC message without marking the message section as packed and aligned. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20231129131411.27516-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 21 +++++++++++---------- sound/soc/sof/ipc4-topology.h | 16 ++++++++++++---- 2 files changed, 23 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 19f36db30979..fae415b9235d 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -141,7 +141,7 @@ static const struct sof_topology_token gain_tokens[] = { /* SRC */ static const struct sof_topology_token src_tokens[] = { {SOF_TKN_SRC_RATE_OUT, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, - offsetof(struct sof_ipc4_src, sink_rate)}, + offsetof(struct sof_ipc4_src_data, sink_rate)}, }; static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = { @@ -811,11 +811,12 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget) swidget->private = src; - ret = sof_ipc4_get_audio_fmt(scomp, swidget, &src->available_fmt, &src->base_config); + ret = sof_ipc4_get_audio_fmt(scomp, swidget, &src->available_fmt, + &src->data.base_config); if (ret) goto err; - ret = sof_update_ipc_object(scomp, src, SOF_SRC_TOKENS, swidget->tuples, + ret = sof_update_ipc_object(scomp, &src->data, SOF_SRC_TOKENS, swidget->tuples, swidget->num_tuples, sizeof(*src), 1); if (ret) { dev_err(scomp->dev, "Parsing SRC tokens failed\n"); @@ -824,7 +825,7 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget) spipe->core_mask |= BIT(swidget->core); - dev_dbg(scomp->dev, "SRC sink rate %d\n", src->sink_rate); + dev_dbg(scomp->dev, "SRC sink rate %d\n", src->data.sink_rate); ret = sof_ipc4_widget_setup_msg(swidget, &src->msg); if (ret) @@ -1900,7 +1901,7 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget, u32 out_ref_rate, out_ref_channels, out_ref_valid_bits; int output_format_index, input_format_index; - input_format_index = sof_ipc4_init_input_audio_fmt(sdev, swidget, &src->base_config, + input_format_index = sof_ipc4_init_input_audio_fmt(sdev, swidget, &src->data.base_config, pipeline_params, available_fmt); if (input_format_index < 0) return input_format_index; @@ -1930,7 +1931,7 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget, */ out_ref_rate = params_rate(fe_params); - output_format_index = sof_ipc4_init_output_audio_fmt(sdev, &src->base_config, + output_format_index = sof_ipc4_init_output_audio_fmt(sdev, &src->data.base_config, available_fmt, out_ref_rate, out_ref_channels, out_ref_valid_bits); if (output_format_index < 0) { @@ -1940,10 +1941,10 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget, } /* update pipeline memory usage */ - sof_ipc4_update_resource_usage(sdev, swidget, &src->base_config); + sof_ipc4_update_resource_usage(sdev, swidget, &src->data.base_config); out_audio_fmt = &available_fmt->output_pin_fmts[output_format_index].audio_fmt; - src->sink_rate = out_audio_fmt->sampling_frequency; + src->data.sink_rate = out_audio_fmt->sampling_frequency; /* update pipeline_params for sink widgets */ return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt); @@ -2344,8 +2345,8 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget { struct sof_ipc4_src *src = swidget->private; - ipc_size = sizeof(struct sof_ipc4_base_module_cfg) + sizeof(src->sink_rate); - ipc_data = src; + ipc_size = sizeof(src->data); + ipc_data = &src->data; msg = &src->msg; break; diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 0a57b8ab3e08..127caca5262a 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -404,16 +404,24 @@ struct sof_ipc4_mixer { struct sof_ipc4_msg msg; }; -/** - * struct sof_ipc4_src SRC config data +/* + * struct sof_ipc4_src_data - IPC data for SRC * @base_config: IPC base config data * @sink_rate: Output rate for sink module + */ +struct sof_ipc4_src_data { + struct sof_ipc4_base_module_cfg base_config; + uint32_t sink_rate; +} __packed __aligned(4); + +/** + * struct sof_ipc4_src - SRC config data + * @data: IPC base config data * @available_fmt: Available audio format * @msg: IPC4 message struct containing header and data info */ struct sof_ipc4_src { - struct sof_ipc4_base_module_cfg base_config; - uint32_t sink_rate; + struct sof_ipc4_src_data data; struct sof_ipc4_available_audio_format available_fmt; struct sof_ipc4_msg msg; }; From e238b68e6dc89ddab52bd98216fe5623e94792b1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 15:14:11 +0200 Subject: [PATCH 30/82] ASoC: SOF: ipc4-topology: Correct data structures for the GAIN module Move the base_cfg to struct sof_ipc4_gain_data. This struct describes the message payload passed to the firmware via the mailbox. It is not wise to be 'clever' and try to use the first part of a struct as IPC message without marking the message section as packed and aligned. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20231129131411.27516-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-control.c | 20 ++++++++++---------- sound/soc/sof/ipc4-topology.c | 31 +++++++++++++++---------------- sound/soc/sof/ipc4-topology.h | 18 +++++++++++++----- 3 files changed, 38 insertions(+), 31 deletions(-) diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 938efaceb81c..b4cdcec33e12 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -89,7 +89,7 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; struct sof_ipc4_gain *gain = swidget->private; struct sof_ipc4_msg *msg = &cdata->msg; - struct sof_ipc4_gain_data data; + struct sof_ipc4_gain_params params; bool all_channels_equal = true; u32 value; int ret, i; @@ -109,20 +109,20 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge */ for (i = 0; i < scontrol->num_channels; i++) { if (all_channels_equal) { - data.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK; - data.init_val = cdata->chanv[0].value; + params.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK; + params.init_val = cdata->chanv[0].value; } else { - data.channels = cdata->chanv[i].channel; - data.init_val = cdata->chanv[i].value; + params.channels = cdata->chanv[i].channel; + params.init_val = cdata->chanv[i].value; } /* set curve type and duration from topology */ - data.curve_duration_l = gain->data.curve_duration_l; - data.curve_duration_h = gain->data.curve_duration_h; - data.curve_type = gain->data.curve_type; + params.curve_duration_l = gain->data.params.curve_duration_l; + params.curve_duration_h = gain->data.params.curve_duration_h; + params.curve_type = gain->data.params.curve_type; - msg->data_ptr = &data; - msg->data_size = sizeof(data); + msg->data_ptr = ¶ms; + msg->data_size = sizeof(params); ret = sof_ipc4_set_get_kcontrol_data(scontrol, true, lock); msg->data_ptr = NULL; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index fae415b9235d..e012b6e166ac 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -130,12 +130,12 @@ static const struct sof_topology_token comp_ext_tokens[] = { static const struct sof_topology_token gain_tokens[] = { {SOF_TKN_GAIN_RAMP_TYPE, SND_SOC_TPLG_TUPLE_TYPE_WORD, - get_token_u32, offsetof(struct sof_ipc4_gain_data, curve_type)}, + get_token_u32, offsetof(struct sof_ipc4_gain_params, curve_type)}, {SOF_TKN_GAIN_RAMP_DURATION, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, - offsetof(struct sof_ipc4_gain_data, curve_duration_l)}, + offsetof(struct sof_ipc4_gain_params, curve_duration_l)}, {SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD, - get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)}, + get_token_u32, offsetof(struct sof_ipc4_gain_params, init_val)}, }; /* SRC */ @@ -720,15 +720,15 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) swidget->private = gain; - gain->data.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK; - gain->data.init_val = SOF_IPC4_VOL_ZERO_DB; + gain->data.params.channels = SOF_IPC4_GAIN_ALL_CHANNELS_MASK; + gain->data.params.init_val = SOF_IPC4_VOL_ZERO_DB; - ret = sof_ipc4_get_audio_fmt(scomp, swidget, &gain->available_fmt, &gain->base_config); + ret = sof_ipc4_get_audio_fmt(scomp, swidget, &gain->available_fmt, &gain->data.base_config); if (ret) goto err; - ret = sof_update_ipc_object(scomp, &gain->data, SOF_GAIN_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(gain->data), 1); + ret = sof_update_ipc_object(scomp, &gain->data.params, SOF_GAIN_TOKENS, + swidget->tuples, swidget->num_tuples, sizeof(gain->data), 1); if (ret) { dev_err(scomp->dev, "Parsing gain tokens failed\n"); goto err; @@ -736,8 +736,8 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) dev_dbg(scomp->dev, "pga widget %s: ramp type: %d, ramp duration %d, initial gain value: %#x\n", - swidget->widget->name, gain->data.curve_type, gain->data.curve_duration_l, - gain->data.init_val); + swidget->widget->name, gain->data.params.curve_type, + gain->data.params.curve_duration_l, gain->data.params.init_val); ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg); if (ret) @@ -1826,7 +1826,7 @@ static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget, u32 out_ref_rate, out_ref_channels, out_ref_valid_bits; int ret; - ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &gain->base_config, + ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &gain->data.base_config, pipeline_params, available_fmt); if (ret < 0) return ret; @@ -1836,7 +1836,7 @@ static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget, out_ref_channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(in_fmt->fmt_cfg); out_ref_valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(in_fmt->fmt_cfg); - ret = sof_ipc4_init_output_audio_fmt(sdev, &gain->base_config, available_fmt, + ret = sof_ipc4_init_output_audio_fmt(sdev, &gain->data.base_config, available_fmt, out_ref_rate, out_ref_channels, out_ref_valid_bits); if (ret < 0) { dev_err(sdev->dev, "Failed to initialize output format for %s", @@ -1845,7 +1845,7 @@ static int sof_ipc4_prepare_gain_module(struct snd_sof_widget *swidget, } /* update pipeline memory usage */ - sof_ipc4_update_resource_usage(sdev, swidget, &gain->base_config); + sof_ipc4_update_resource_usage(sdev, swidget, &gain->data.base_config); return 0; } @@ -2324,9 +2324,8 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget { struct sof_ipc4_gain *gain = swidget->private; - ipc_size = sizeof(struct sof_ipc4_base_module_cfg) + - sizeof(struct sof_ipc4_gain_data); - ipc_data = gain; + ipc_size = sizeof(gain->data); + ipc_data = &gain->data; msg = &gain->msg; break; diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 127caca5262a..dce174a190dd 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -361,7 +361,7 @@ struct sof_ipc4_control_msg_payload { } __packed; /** - * struct sof_ipc4_gain_data - IPC gain blob + * struct sof_ipc4_gain_params - IPC gain parameters * @channels: Channels * @init_val: Initial value * @curve_type: Curve type @@ -369,24 +369,32 @@ struct sof_ipc4_control_msg_payload { * @curve_duration_l: Curve duration low part * @curve_duration_h: Curve duration high part */ -struct sof_ipc4_gain_data { +struct sof_ipc4_gain_params { uint32_t channels; uint32_t init_val; uint32_t curve_type; uint32_t reserved; uint32_t curve_duration_l; uint32_t curve_duration_h; -} __aligned(8); +} __packed __aligned(4); + +/** + * struct sof_ipc4_gain_data - IPC gain init blob + * @base_config: IPC base config data + * @params: Initial parameters for the gain module + */ +struct sof_ipc4_gain_data { + struct sof_ipc4_base_module_cfg base_config; + struct sof_ipc4_gain_params params; +} __packed __aligned(4); /** * struct sof_ipc4_gain - gain config data - * @base_config: IPC base config data * @data: IPC gain blob * @available_fmt: Available audio format * @msg: message structure for gain */ struct sof_ipc4_gain { - struct sof_ipc4_base_module_cfg base_config; struct sof_ipc4_gain_data data; struct sof_ipc4_available_audio_format available_fmt; struct sof_ipc4_msg msg; From c5c325bb5849868d76969d3fe014515f5e99eabc Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Pascal=20No=C3=ABl?= Date: Fri, 1 Dec 2023 17:37:44 -0800 Subject: [PATCH 31/82] ALSA: hda/realtek: Apply quirk for ASUS UM3504DA MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The ASUS UM3504DA uses a Realtek HDA codec and two CS35L41 amplifiers via I2C. Apply existing quirk to model. Signed-off-by: Pascal Noël Cc: Link: https://lore.kernel.org/r/20231202013744.12369-1-pascal@pascalcompiles.com Signed-off-by: Takashi Iwai --- 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 f9ddacfd920e..ddd74f5d3b30 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9963,6 +9963,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally RC71L_RC71L", ALC294_FIXUP_ASUS_ALLY), 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), + SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS UM3504DA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1970, "ASUS UX550VE", ALC289_FIXUP_ASUS_GA401), From b5338b1b901e41bd7cead66a0b3a796e9fa95684 Mon Sep 17 00:00:00 2001 From: Marian Postevca Date: Sun, 3 Dec 2023 00:29:51 +0200 Subject: [PATCH 32/82] ASoC: amd: acp: Add support for a new Huawei Matebook laptop This commit adds support for Huawei MateBook D16 2021 with Ryzen 4600H in driver acp3x-es83xx. Signed-off-by: Marian Postevca Link: https://lore.kernel.org/r/20231202223001.8025-1-posteuca@mutex.one Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 20cee7104c2b..3bc4b2e41650 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -103,6 +103,20 @@ static const struct config_entry config_table[] = { {} }, }, + { + .flags = FLAG_AMD_LEGACY, + .device = ACP_PCI_DEV_ID, + .dmi_table = (const struct dmi_system_id []) { + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"), + }, + }, + {} + }, + }, { .flags = FLAG_AMD_LEGACY, .device = ACP_PCI_DEV_ID, From 5f44de697383fcc9a9a1a78f99e09d1838704b90 Mon Sep 17 00:00:00 2001 From: David Rau Date: Fri, 1 Dec 2023 12:29:33 +0800 Subject: [PATCH 33/82] ASoC: da7219: Support low DC impedance headset Change the default MIC detection impedance threshold to 200ohm to support low mic DC impedance headset. Signed-off-by: David Rau Link: https://lore.kernel.org/r/20231201042933.26392-1-David.Rau.opensource@dm.renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/da7219-aad.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 4c4405942779..6bc068cdcbe2 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -696,7 +696,7 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) aad_pdata->mic_det_thr = da7219_aad_fw_mic_det_thr(dev, fw_val32); else - aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS; + aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_200_OHMS; if (fwnode_property_read_u32(aad_np, "dlg,jack-ins-deb", &fw_val32) >= 0) aad_pdata->jack_ins_deb = From 29046a78a3c0a1f8fa0427f164caa222f003cf5b Mon Sep 17 00:00:00 2001 From: Dinghao Liu Date: Mon, 4 Dec 2023 15:41:56 +0800 Subject: [PATCH 34/82] ASoC: wm_adsp: fix memleak in wm_adsp_buffer_populate When wm_adsp_buffer_read() fails, we should free buf->regions. Otherwise, the callers of wm_adsp_buffer_populate() will directly free buf on failure, which makes buf->regions a leaked memory. Fixes: a792af69b08f ("ASoC: wm_adsp: Refactor compress stream initialisation") Signed-off-by: Dinghao Liu Reviewed-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20231204074158.12026-1-dinghao.liu@zju.edu.cn Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 236b12b69ae5..c01e31175015 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1451,12 +1451,12 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) ret = wm_adsp_buffer_read(buf, caps->region_defs[i].base_offset, ®ion->base_addr); if (ret < 0) - return ret; + goto err; ret = wm_adsp_buffer_read(buf, caps->region_defs[i].size_offset, &offset); if (ret < 0) - return ret; + goto err; region->cumulative_size = offset; @@ -1467,6 +1467,10 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) } return 0; + +err: + kfree(buf->regions); + return ret; } static void wm_adsp_buffer_clear(struct wm_adsp_compr_buf *buf) From bbb8e71965c3737bdc691afd803a34bfd61cfbeb Mon Sep 17 00:00:00 2001 From: Sarah Grant Date: Fri, 1 Dec 2023 18:16:54 +0000 Subject: [PATCH 35/82] ALSA: usb-audio: Add Pioneer DJM-450 mixer controls These values mirror those of the Pioneer DJM-250MK2 as the channel layout appears identical based on my observations. This duplication could be removed in later contributions if desired. Signed-off-by: Sarah Grant Cc: Link: https://lore.kernel.org/r/20231201181654.5058-1-s@srd.tw Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 898bc3baca7b..c8d48566e175 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2978,6 +2978,7 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_850_IDX 0x2 #define SND_DJM_900NXS2_IDX 0x3 #define SND_DJM_750MK2_IDX 0x4 +#define SND_DJM_450_IDX 0x5 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -3108,6 +3109,31 @@ static const struct snd_djm_ctl snd_djm_ctls_250mk2[] = { }; +// DJM-450 +static const u16 snd_djm_opts_450_cap1[] = { + 0x0103, 0x0100, 0x0106, 0x0107, 0x0108, 0x0109, 0x010d, 0x010a }; + +static const u16 snd_djm_opts_450_cap2[] = { + 0x0203, 0x0200, 0x0206, 0x0207, 0x0208, 0x0209, 0x020d, 0x020a }; + +static const u16 snd_djm_opts_450_cap3[] = { + 0x030a, 0x0311, 0x0312, 0x0307, 0x0308, 0x0309, 0x030d }; + +static const u16 snd_djm_opts_450_pb1[] = { 0x0100, 0x0101, 0x0104 }; +static const u16 snd_djm_opts_450_pb2[] = { 0x0200, 0x0201, 0x0204 }; +static const u16 snd_djm_opts_450_pb3[] = { 0x0300, 0x0301, 0x0304 }; + +static const struct snd_djm_ctl snd_djm_ctls_450[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Ch1 Input", 450_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", 450_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", 450_cap3, 0, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Output", 450_pb1, 0, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch2 Output", 450_pb2, 1, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch3 Output", 450_pb3, 2, SND_DJM_WINDEX_PB) +}; + + // DJM-750 static const u16 snd_djm_opts_750_cap1[] = { 0x0101, 0x0103, 0x0106, 0x0107, 0x0108, 0x0109, 0x010a, 0x010f }; @@ -3203,6 +3229,7 @@ static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_850_IDX] = SND_DJM_DEVICE(850), [SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2), [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), + [SND_DJM_450_IDX] = SND_DJM_DEVICE(450), }; @@ -3454,6 +3481,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ err = snd_djm_controls_create(mixer, SND_DJM_250MK2_IDX); break; + case USB_ID(0x2b73, 0x0013): /* Pioneer DJ DJM-450 */ + err = snd_djm_controls_create(mixer, SND_DJM_450_IDX); + break; case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ err = snd_djm_controls_create(mixer, SND_DJM_750_IDX); break; From cd14dedf15be432066e63783c63d650f2800cd48 Mon Sep 17 00:00:00 2001 From: Aleksandrs Vinarskis Date: Mon, 4 Dec 2023 00:30:06 +0100 Subject: [PATCH 36/82] ALSA: hda/realtek: fix speakers on XPS 9530 (2023) XPS 9530 has 2 tweeters and 2 subwoofers powered by CS35L41 amplifier, SPI connected. For subwoofers to work, it requires both to enable amplifier support, and to enable output to subwoofers via 0x17 quirk (similalry to XPS 9510/9520). Signed-off-by: Aleksandrs Vinarskis Cc: Link: https://lore.kernel.org/r/20231203233006.100558-1-alex.vinarskis@gmail.com Signed-off-by: Takashi Iwai --- 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 ddd74f5d3b30..1c85e6dcef6c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9705,6 +9705,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), From 6f7e4664e597440dfbdb8b2931c561b717030d07 Mon Sep 17 00:00:00 2001 From: Bin Li Date: Mon, 4 Dec 2023 18:04:50 +0800 Subject: [PATCH 37/82] ALSA: hda/realtek: Enable headset on Lenovo M90 Gen5 Lenovo M90 Gen5 is equipped with ALC897, and it needs ALC897_FIXUP_HEADSET_MIC_PIN quirk to make its headset mic work. Signed-off-by: Bin Li Cc: Link: https://lore.kernel.org/r/20231204100450.642783-1-bin.li@canonical.com Signed-off-by: Takashi Iwai --- 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 1c85e6dcef6c..d799d0ad7623 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12198,6 +12198,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x3321, "Lenovo ThinkCentre M70 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x331b, "Lenovo ThinkCentre M90 Gen4", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x3364, "Lenovo ThinkCentre M90 Gen5", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x3742, "Lenovo TianYi510Pro-14IOB", ALC897_FIXUP_HEADSET_MIC_PIN2), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), From fb9ad24485087e0f00d84bee7a5914640b2b9024 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 4 Dec 2023 12:47:35 +0000 Subject: [PATCH 38/82] ASoC: ops: add correct range check for limiting volume Volume can have ranges that start with negative values, ex: -84dB to +40dB. Apply correct range check in snd_soc_limit_volume before setting the platform_max. Without this patch, for example setting a 0dB limit on a volume range of -84dB to +40dB would fail. Signed-off-by: Srinivas Kandagatla Tested-by: Johan Hovold Reviewed-by: Johan Hovold Link: https://lore.kernel.org/r/20231204124736.132185-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 55b009d3c681..2d25748ca706 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -661,7 +661,7 @@ int snd_soc_limit_volume(struct snd_soc_card *card, kctl = snd_soc_card_get_kcontrol(card, name); if (kctl) { struct soc_mixer_control *mc = (struct soc_mixer_control *)kctl->private_value; - if (max <= mc->max) { + if (max <= mc->max - mc->min) { mc->platform_max = max; ret = 0; } From 716d4e5373e9d1ae993485ab2e3b893bf7104fb1 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 4 Dec 2023 12:47:36 +0000 Subject: [PATCH 39/82] ASoC: qcom: sc8280xp: Limit speaker digital volumes Limit the speaker digital gains to 0dB so that the users will not damage them. Currently there is a limit in UCM, but this does not stop the user form changing the digital gains from command line. So limit this in driver which makes the speakers more safer without active speaker protection in place. Signed-off-by: Srinivas Kandagatla Reviewed-by: Johan Hovold Tested-by: Johan Hovold Link: https://lore.kernel.org/r/20231204124736.132185-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index d93b18f07be5..39cb0b889aff 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -27,6 +27,23 @@ struct sc8280xp_snd_data { static int sc8280xp_snd_init(struct snd_soc_pcm_runtime *rtd) { struct sc8280xp_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct snd_soc_card *card = rtd->card; + + switch (cpu_dai->id) { + case WSA_CODEC_DMA_RX_0: + case WSA_CODEC_DMA_RX_1: + /* + * set limit of 0dB on Digital Volume for Speakers, + * this can prevent damage of speakers to some extent without + * active speaker protection + */ + snd_soc_limit_volume(card, "WSA_RX0 Digital Volume", 84); + snd_soc_limit_volume(card, "WSA_RX1 Digital Volume", 84); + break; + default: + break; + } return qcom_snd_wcd_jack_setup(rtd, &data->jack, &data->jack_setup); } From 2b3a7a302c9804e463f2ea5b54dc3a6ad106a344 Mon Sep 17 00:00:00 2001 From: Jason Zhang Date: Wed, 6 Dec 2023 09:31:39 +0800 Subject: [PATCH 40/82] ALSA: pcm: fix out-of-bounds in snd_pcm_state_names The pcm state can be SNDRV_PCM_STATE_DISCONNECTED at disconnect callback, and there is not an entry of SNDRV_PCM_STATE_DISCONNECTED in snd_pcm_state_names. This patch adds the missing entry to resolve this issue. cat /proc/asound/card2/pcm0p/sub0/status That results in stack traces like the following: [ 99.702732][ T5171] Unexpected kernel BRK exception at EL1 [ 99.702774][ T5171] Internal error: BRK handler: f2005512 [#1] PREEMPT SMP [ 99.703858][ T5171] Modules linked in: bcmdhd(E) (...) [ 99.747425][ T5171] CPU: 3 PID: 5171 Comm: cat Tainted: G C OE 5.10.189-android13-4-00003-g4a17384380d8-ab11086999 #1 [ 99.748447][ T5171] Hardware name: Rockchip RK3588 CVTE V10 Board (DT) [ 99.749024][ T5171] pstate: 60400005 (nZCv daif +PAN -UAO -TCO BTYPE=--) [ 99.749616][ T5171] pc : snd_pcm_substream_proc_status_read+0x264/0x2bc [ 99.750204][ T5171] lr : snd_pcm_substream_proc_status_read+0xa4/0x2bc [ 99.750778][ T5171] sp : ffffffc0175abae0 [ 99.751132][ T5171] x29: ffffffc0175abb80 x28: ffffffc009a2c498 [ 99.751665][ T5171] x27: 0000000000000001 x26: ffffff810cbae6e8 [ 99.752199][ T5171] x25: 0000000000400cc0 x24: ffffffc0175abc60 [ 99.752729][ T5171] x23: 0000000000000000 x22: ffffff802f558400 [ 99.753263][ T5171] x21: ffffff81d8d8ff00 x20: ffffff81020cdc00 [ 99.753795][ T5171] x19: ffffff802d110000 x18: ffffffc014fbd058 [ 99.754326][ T5171] x17: 0000000000000000 x16: 0000000000000000 [ 99.754861][ T5171] x15: 000000000000c276 x14: ffffffff9a976fda [ 99.755392][ T5171] x13: 0000000065689089 x12: 000000000000d72e [ 99.755923][ T5171] x11: ffffff802d110000 x10: 00000000000000e0 [ 99.756457][ T5171] x9 : 9c431600c8385d00 x8 : 0000000000000008 [ 99.756990][ T5171] x7 : 0000000000000000 x6 : 000000000000003f [ 99.757522][ T5171] x5 : 0000000000000040 x4 : ffffffc0175abb70 [ 99.758056][ T5171] x3 : 0000000000000001 x2 : 0000000000000001 [ 99.758588][ T5171] x1 : 0000000000000000 x0 : 0000000000000000 [ 99.759123][ T5171] Call trace: [ 99.759404][ T5171] snd_pcm_substream_proc_status_read+0x264/0x2bc [ 99.759958][ T5171] snd_info_seq_show+0x54/0xa4 [ 99.760370][ T5171] seq_read_iter+0x19c/0x7d4 [ 99.760770][ T5171] seq_read+0xf0/0x128 [ 99.761117][ T5171] proc_reg_read+0x100/0x1f8 [ 99.761515][ T5171] vfs_read+0xf4/0x354 [ 99.761869][ T5171] ksys_read+0x7c/0x148 [ 99.762226][ T5171] __arm64_sys_read+0x20/0x30 [ 99.762625][ T5171] el0_svc_common+0xd0/0x1e4 [ 99.763023][ T5171] el0_svc+0x28/0x98 [ 99.763358][ T5171] el0_sync_handler+0x8c/0xf0 [ 99.763759][ T5171] el0_sync+0x1b8/0x1c0 [ 99.764118][ T5171] Code: d65f03c0 b9406102 17ffffae 94191565 (d42aa240) [ 99.764715][ T5171] ---[ end trace 1eeffa3e17c58e10 ]--- [ 99.780720][ T5171] Kernel panic - not syncing: BRK handler: Fatal exception Signed-off-by: Jason Zhang Cc: Link: https://lore.kernel.org/r/20231206013139.20506-1-jason.zhang@rock-chips.com Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 20bb2d7c8d4b..6d0c9c37796c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -253,6 +253,7 @@ static const char * const snd_pcm_state_names[] = { STATE(DRAINING), STATE(PAUSED), STATE(SUSPENDED), + STATE(DISCONNECTED), }; static const char * const snd_pcm_access_names[] = { From 33038efb64f7576bac635164021f5c984d4c755f Mon Sep 17 00:00:00 2001 From: Tim Bosse Date: Wed, 6 Dec 2023 09:26:29 -0500 Subject: [PATCH 41/82] ALSA: hda/realtek: add new Framework laptop to quirks The Framework Laptop 13 (AMD Ryzen 7040Series) has an ALC295 with a disconnected or faulty headset mic presence detect similar to the previous models. It works with the same quirk chain as 309d7363ca3d9fcdb92ff2d958be14d7e8707f68. This model has a VID:PID of f111:0006. Signed-off-by: Tim Bosse Cc: Link: https://lore.kernel.org/r/20231206142629.388615-1-flinn@timbos.se Signed-off-by: Takashi Iwai --- 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 d799d0ad7623..51e1bfd7bdde 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10273,6 +10273,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. From d20d36755a605a21e737b6b16c566658589b1811 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Tue, 5 Dec 2023 14:01:18 -0800 Subject: [PATCH 42/82] ASoC: SOF: mediatek: mt8186: Revert Add Google Steelix topology compatible This reverts commit 505c83212da5bfca95109421b8f5d9f8c6cdfef2. This is not an official topology from the SOF project. Topologies are named based on the card configuration and are NOT board specific. Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20231205220131.2585913-1-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8186/mt8186.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index e0d88e7aa8ca..b69fa788b16f 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -597,9 +597,6 @@ static struct snd_sof_dsp_ops sof_mt8186_ops = { static struct snd_sof_of_mach sof_mt8186_machs[] = { { - .compatible = "google,steelix", - .sof_tplg_filename = "sof-mt8186-google-steelix.tplg" - }, { .compatible = "mediatek,mt8186", .sof_tplg_filename = "sof-mt8186.tplg", }, From 12e102b1bd22ee00361559d57a5876445bcb2407 Mon Sep 17 00:00:00 2001 From: Ricardo Rivera-Matos Date: Wed, 6 Dec 2023 10:03:16 -0600 Subject: [PATCH 43/82] ASoC: cs35l45: Use modern pm_ops Make use of the recently introduced EXPORT_GPL_DEV_PM_OPS() macro, to conditionally export the runtime/system PM functions. Replace the old SET_{RUNTIME,SYSTEM_SLEEP,NOIRQ_SYSTEM_SLEEP}_PM_OPS() helpers with their modern alternatives and get rid of the now unnecessary '__maybe_unused' annotations on all PM functions. Additionally, use the pm_ptr() macro to fix the following errors when building with CONFIG_PM disabled: Signed-off-by: Ricardo Rivera-Matos Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231206160318.1255034-2-rriveram@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l45-i2c.c | 2 +- sound/soc/codecs/cs35l45-spi.c | 2 +- sound/soc/codecs/cs35l45.c | 9 ++++----- 3 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 77e0f8750f37..bc2af1ed0fe9 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -62,7 +62,7 @@ static struct i2c_driver cs35l45_i2c_driver = { .driver = { .name = "cs35l45", .of_match_table = cs35l45_of_match, - .pm = &cs35l45_pm_ops, + .pm = pm_ptr(&cs35l45_pm_ops), }, .id_table = cs35l45_id_i2c, .probe = cs35l45_i2c_probe, diff --git a/sound/soc/codecs/cs35l45-spi.c b/sound/soc/codecs/cs35l45-spi.c index 5efb77530cc3..39e203a5f060 100644 --- a/sound/soc/codecs/cs35l45-spi.c +++ b/sound/soc/codecs/cs35l45-spi.c @@ -64,7 +64,7 @@ static struct spi_driver cs35l45_spi_driver = { .driver = { .name = "cs35l45", .of_match_table = cs35l45_of_match, - .pm = &cs35l45_pm_ops, + .pm = pm_ptr(&cs35l45_pm_ops), }, .id_table = cs35l45_id_spi, .probe = cs35l45_spi_probe, diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index b68853e42fd1..4f4df166f5f0 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -982,7 +982,7 @@ static int cs35l45_exit_hibernate(struct cs35l45_private *cs35l45) return -ETIMEDOUT; } -static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) +static int cs35l45_runtime_suspend(struct device *dev) { struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); @@ -999,7 +999,7 @@ static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) return 0; } -static int __maybe_unused cs35l45_runtime_resume(struct device *dev) +static int cs35l45_runtime_resume(struct device *dev) { struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); int ret; @@ -1466,10 +1466,9 @@ void cs35l45_remove(struct cs35l45_private *cs35l45) } EXPORT_SYMBOL_NS_GPL(cs35l45_remove, SND_SOC_CS35L45); -const struct dev_pm_ops cs35l45_pm_ops = { - SET_RUNTIME_PM_OPS(cs35l45_runtime_suspend, cs35l45_runtime_resume, NULL) +EXPORT_GPL_DEV_PM_OPS(cs35l45_pm_ops) = { + RUNTIME_PM_OPS(cs35l45_runtime_suspend, cs35l45_runtime_resume, NULL) }; -EXPORT_SYMBOL_NS_GPL(cs35l45_pm_ops, SND_SOC_CS35L45); MODULE_DESCRIPTION("ASoC CS35L45 driver"); MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, "); From c3c8b088949b9ccb88da2f84d3c3cc06580a6a43 Mon Sep 17 00:00:00 2001 From: Ricardo Rivera-Matos Date: Wed, 6 Dec 2023 10:03:17 -0600 Subject: [PATCH 44/82] ASoC: cs35l45: Prevent IRQ handling when suspending/resuming Use the SYSTEM_SLEEP_PM_OPS handlers to prevent handling an IRQ when the system is in the middle of suspending or resuming. Signed-off-by: Ricardo Rivera-Matos Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231206160318.1255034-3-rriveram@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l45.c | 43 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 4f4df166f5f0..28f76fccf277 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -1026,6 +1026,46 @@ static int cs35l45_runtime_resume(struct device *dev) return ret; } +static int cs35l45_sys_suspend(struct device *dev) +{ + struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); + + dev_dbg(cs35l45->dev, "System suspend, disabling IRQ\n"); + disable_irq(cs35l45->irq); + + return 0; +} + +static int cs35l45_sys_suspend_noirq(struct device *dev) +{ + struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); + + dev_dbg(cs35l45->dev, "Late system suspend, reenabling IRQ\n"); + enable_irq(cs35l45->irq); + + return 0; +} + +static int cs35l45_sys_resume_noirq(struct device *dev) +{ + struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); + + dev_dbg(cs35l45->dev, "Early system resume, disabling IRQ\n"); + disable_irq(cs35l45->irq); + + return 0; +} + +static int cs35l45_sys_resume(struct device *dev) +{ + struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); + + dev_dbg(cs35l45->dev, "System resume, reenabling IRQ\n"); + enable_irq(cs35l45->irq); + + return 0; +} + static int cs35l45_apply_property_config(struct cs35l45_private *cs35l45) { struct device_node *node = cs35l45->dev->of_node; @@ -1468,6 +1508,9 @@ EXPORT_SYMBOL_NS_GPL(cs35l45_remove, SND_SOC_CS35L45); EXPORT_GPL_DEV_PM_OPS(cs35l45_pm_ops) = { RUNTIME_PM_OPS(cs35l45_runtime_suspend, cs35l45_runtime_resume, NULL) + + SYSTEM_SLEEP_PM_OPS(cs35l45_sys_suspend, cs35l45_sys_resume) + NOIRQ_SYSTEM_SLEEP_PM_OPS(cs35l45_sys_suspend_noirq, cs35l45_sys_resume_noirq) }; MODULE_DESCRIPTION("ASoC CS35L45 driver"); From a0ffa8115e1ea9786b03edc3f431d2f4ef3e7a2e Mon Sep 17 00:00:00 2001 From: Ricardo Rivera-Matos Date: Wed, 6 Dec 2023 10:03:18 -0600 Subject: [PATCH 45/82] ASoC: cs35l45: Prevents spinning during runtime suspend Masks the "DSP Virtual Mailbox 2 write" interrupt when before issuing the hibernate command to the DSP. The interrupt is unmasked when exiting runtime suspend as it is required for DSP operation. Without this change the DSP fires an interrupt when hibernating causing the system spin between runtime suspend and runtime resume. Signed-off-by: Ricardo Rivera-Matos Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231206160318.1255034-4-rriveram@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l45.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 28f76fccf277..44c221745c3b 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -947,6 +947,8 @@ static int cs35l45_enter_hibernate(struct cs35l45_private *cs35l45) cs35l45_setup_hibernate(cs35l45); + regmap_set_bits(cs35l45->regmap, CS35L45_IRQ1_MASK_2, CS35L45_DSP_VIRT2_MBOX_MASK); + // Don't wait for ACK since bus activity would wake the device regmap_write(cs35l45->regmap, CS35L45_DSP_VIRT1_MBOX_1, CSPL_MBOX_CMD_HIBERNATE); @@ -967,6 +969,8 @@ static int cs35l45_exit_hibernate(struct cs35l45_private *cs35l45) CSPL_MBOX_CMD_OUT_OF_HIBERNATE); if (!ret) { dev_dbg(cs35l45->dev, "Wake success at cycle: %d\n", j); + regmap_clear_bits(cs35l45->regmap, CS35L45_IRQ1_MASK_2, + CS35L45_DSP_VIRT2_MBOX_MASK); return 0; } usleep_range(100, 200); From 8804fa04a492f4176ea407390052292912227820 Mon Sep 17 00:00:00 2001 From: Mario Limonciello Date: Wed, 6 Dec 2023 13:39:27 -0600 Subject: [PATCH 46/82] ALSA: hda/realtek: Add Framework laptop 16 to quirks The Framework 16" laptop has the same controller as other Framework models. Apply the presence detection quirk. Signed-off-by: Mario Limonciello Cc: Link: https://lore.kernel.org/r/20231206193927.2996-1-mario.limonciello@amd.com Signed-off-by: Takashi Iwai --- 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 51e1bfd7bdde..5b5f298870be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10273,6 +10273,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0005, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 From eb99b1b72a424a79f56c972e0fd7ad01fe93a008 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Wed, 6 Dec 2023 22:32:11 +0000 Subject: [PATCH 47/82] ALSA: pcmtest: stop timer before buffer is released Stop timer in the 'trigger' and 'sync_stop' callbacks since we want the timer to be stopped before the DMA buffer is released. Otherwise, it could trigger a kernel panic in some circumstances, for instance when the DMA buffer is already released but the timer callback is still running. Signed-off-by: Ivan Orlov Link: https://lore.kernel.org/r/20231206223211.12761-1-ivan.orlov0322@gmail.com Signed-off-by: Takashi Iwai --- sound/drivers/pcmtest.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index b59b78a09224..b8bff5522bce 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -397,7 +397,6 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream) struct pcmtst_buf_iter *v_iter = substream->runtime->private_data; timer_shutdown_sync(&v_iter->timer_instance); - v_iter->substream = NULL; playback_capture_test = !v_iter->is_buf_corrupted; kfree(v_iter); return 0; @@ -435,6 +434,7 @@ static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_PUSH: // We can't call timer_shutdown_sync here, as it is forbidden to sleep here v_iter->suspend = true; + timer_delete(&v_iter->timer_instance); break; } @@ -512,12 +512,22 @@ static int snd_pcmtst_ioctl(struct snd_pcm_substream *substream, unsigned int cm return snd_pcm_lib_ioctl(substream, cmd, arg); } +static int snd_pcmtst_sync_stop(struct snd_pcm_substream *substream) +{ + struct pcmtst_buf_iter *v_iter = substream->runtime->private_data; + + timer_delete_sync(&v_iter->timer_instance); + + return 0; +} + static const struct snd_pcm_ops snd_pcmtst_playback_ops = { .open = snd_pcmtst_pcm_open, .close = snd_pcmtst_pcm_close, .trigger = snd_pcmtst_pcm_trigger, .hw_params = snd_pcmtst_pcm_hw_params, .ioctl = snd_pcmtst_ioctl, + .sync_stop = snd_pcmtst_sync_stop, .hw_free = snd_pcmtst_pcm_hw_free, .prepare = snd_pcmtst_pcm_prepare, .pointer = snd_pcmtst_pcm_pointer, @@ -530,6 +540,7 @@ static const struct snd_pcm_ops snd_pcmtst_capture_ops = { .hw_params = snd_pcmtst_pcm_hw_params, .hw_free = snd_pcmtst_pcm_hw_free, .ioctl = snd_pcmtst_ioctl, + .sync_stop = snd_pcmtst_sync_stop, .prepare = snd_pcmtst_pcm_prepare, .pointer = snd_pcmtst_pcm_pointer, }; From 634e5e1e06f5cdd614a1bc429ecb243a51cc009d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Dec 2023 19:20:35 +0100 Subject: [PATCH 48/82] ALSA: hda/realtek: Add quirk for Lenovo Yoga Pro 7 Lenovo Yoga Pro 7 14APH8 (PCI SSID 17aa:3882) seems requiring the similar workaround like Yoga 9 model for the bass speaker. Cc: Link: https://lore.kernel.org/r/CAGGk=CRRQ1L9p771HsXTN_ebZP41Qj+3gw35Gezurn+nokRewg@mail.gmail.com Link: https://lore.kernel.org/r/20231207182035.30248-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- 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 5b5f298870be..0377912e9264 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10206,6 +10206,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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 power mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3882, "Lenovo Yoga Pro 7 14APH8", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 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), From 3b1ff57e24a7bcd2e2a8426dd2013a80d1fa96eb Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 8 Dec 2023 15:21:26 +0200 Subject: [PATCH 49/82] ALSA: hda/hdmi: add force-connect quirk for NUC5CPYB MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add one more older NUC model that requires quirk to force all pins to be connected. The display codec pins are not registered properly without the force-connect quirk. The codec will report only one pin as having external connectivity, but i915 finds all three connectors on the system, so the two drivers are not in sync. Issue found with DRM igt-gpu-tools test kms_hdmi_inject@inject-audio. Link: https://gitlab.freedesktop.org/drm/igt-gpu-tools/-/issues/3 Cc: Ville Syrjälä Cc: Jani Saarinen Signed-off-by: Kai Vehmanen Cc: Link: https://lore.kernel.org/r/20231208132127.2438067-2-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1cde2a69bdb4..b152c941414f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1994,6 +1994,7 @@ static const struct snd_pci_quirk force_connect_list[] = { SND_PCI_QUIRK(0x103c, 0x8711, "HP", 1), SND_PCI_QUIRK(0x103c, 0x8715, "HP", 1), SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1), + SND_PCI_QUIRK(0x8086, 0x2060, "Intel NUC5CPYB", 1), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1), {} }; From 924f5ca2975b2993ee81a7ecc3c809943a70f334 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 8 Dec 2023 15:21:27 +0200 Subject: [PATCH 50/82] ALSA: hda/hdmi: add force-connect quirks for ASUSTeK Z170 variants MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On ASUSTeK Z170M PLUS and Z170 PRO GAMING systems, the display codec pins are not registered properly without the force-connect quirk. The codec will report only one pin as having external connectivity, but i915 finds all three connectors on the system, so the two drivers are not in sync. Issue found with DRM igt-gpu-tools test kms_hdmi_inject@inject-audio. Link: https://gitlab.freedesktop.org/drm/intel/-/issues/9801 Cc: Ville Syrjälä Cc: Jani Saarinen Signed-off-by: Kai Vehmanen Cc: Link: https://lore.kernel.org/r/20231208132127.2438067-3-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b152c941414f..78cee53fee02 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1993,6 +1993,8 @@ static const struct snd_pci_quirk force_connect_list[] = { SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), SND_PCI_QUIRK(0x103c, 0x8711, "HP", 1), SND_PCI_QUIRK(0x103c, 0x8715, "HP", 1), + SND_PCI_QUIRK(0x1043, 0x86ae, "ASUS", 1), /* Z170 PRO */ + SND_PCI_QUIRK(0x1043, 0x86c7, "ASUS", 1), /* Z170M PLUS */ SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1), SND_PCI_QUIRK(0x8086, 0x2060, "Intel NUC5CPYB", 1), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1), From 9b726bf6ae11add6a7a52883a21f90ff9cbca916 Mon Sep 17 00:00:00 2001 From: Hartmut Knaack Date: Sat, 9 Dec 2023 15:47:07 +0100 Subject: [PATCH 51/82] ALSA: hda/realtek: Apply mute LED quirk for HP15-db The HP laptop 15-db0403ng uses the ALC236 codec and controls the mute LED using COEF 0x07 index 1. Sound card subsystem: Hewlett-Packard Company Device [103c:84ae] Use the existing quirk for this model. Signed-off-by: Hartmut Knaack Cc: Link: https://lore.kernel.org/r/e61815d0-f1c7-b164-e49d-6ca84771476a@gmx.de Signed-off-by: Takashi Iwai --- 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 0377912e9264..e45d4c405f8f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9795,6 +9795,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84ae, "HP 15-db0403ng", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), From 75a25d31b80770485641ad2789a854955f5c1e40 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Sat, 9 Dec 2023 22:18:29 +0100 Subject: [PATCH 52/82] ALSA: hda/tas2781: leave hda_component in usable state Unloading then loading the module causes a NULL ponter dereference. The hda_unbind zeroes the hda_component, later the hda_bind tries to dereference the codec field. The hda_component is only initialized once by tas2781_generic_fixup. Set only previously modified fields to NULL. BUG: kernel NULL pointer dereference, address: 0000000000000322 Call Trace: ? __die+0x23/0x70 ? page_fault_oops+0x171/0x4e0 ? exc_page_fault+0x7f/0x180 ? asm_exc_page_fault+0x26/0x30 ? tas2781_hda_bind+0x59/0x140 [snd_hda_scodec_tas2781_i2c] component_bind_all+0xf3/0x240 try_to_bring_up_aggregate_device+0x1c3/0x270 __component_add+0xbc/0x1a0 tas2781_hda_i2c_probe+0x289/0x3a0 [snd_hda_scodec_tas2781_i2c] i2c_device_probe+0x136/0x2e0 Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Cc: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/8b8ed2bd5f75fbb32e354a3226c2f966fa85b46b.1702156522.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index fb802802939e..b42837105c22 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -612,9 +612,13 @@ static void tas2781_hda_unbind(struct device *dev, { struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); struct hda_component *comps = master_data; + comps = &comps[tas_priv->index]; - if (comps[tas_priv->index].dev == dev) - memset(&comps[tas_priv->index], 0, sizeof(*comps)); + if (comps->dev == dev) { + comps->dev = NULL; + memset(comps->name, 0, sizeof(comps->name)); + comps->playback_hook = NULL; + } tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); From 33071422714a4c9587753b0ccc130ca59323bf42 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Mon, 11 Dec 2023 00:37:33 +0100 Subject: [PATCH 53/82] ALSA: hda/tas2781: handle missing EFI calibration data The code does not properly check whether the calibration variable is available in the EFI. If it is not available, it causes a NULL pointer dereference. Check the return value of the first get_variable call also. BUG: kernel NULL pointer dereference, address: 0000000000000000 Call Trace: ? __die+0x23/0x70 ? page_fault_oops+0x171/0x4e0 ? srso_alias_return_thunk+0x5/0x7f ? schedule+0x5e/0xd0 ? exc_page_fault+0x7f/0x180 ? asm_exc_page_fault+0x26/0x30 ? crc32_body+0x2c/0x120 ? tas2781_save_calibration+0xe4/0x220 [snd_hda_scodec_tas2781_i2c] tasdev_fw_ready+0x1af/0x280 [snd_hda_scodec_tas2781_i2c] request_firmware_work_func+0x59/0xa0 Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/f1f6583bda918f78556f67d522ca7b3b91cebbd5.1702251102.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index b42837105c22..d3dafc9d150b 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -455,9 +455,9 @@ static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) status = efi.get_variable(efi_name, &efi_guid, &attr, &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); - if (status != EFI_SUCCESS) - return -EINVAL; } + if (status != EFI_SUCCESS) + return -EINVAL; tmp_val = (unsigned int *)tas_priv->cali_data.data; From 02a914ed475dd928c7b2b6c9d1da9b0b27fa724d Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 5 Dec 2023 11:57:15 +0000 Subject: [PATCH 54/82] ASoC: Intel: soc-acpi-intel-mtl-match: Change CS35L56 prefixes to AMPn Change the ALSA prefix for the CS35L56 to "AMPn". This keeps them consistent with the CS35L56 HDA driver. It also avoids coding the chip ID into the control name, so that other Cirrus amps with the same controls can have the same control names. Signed-off-by: Richard Fitzgerald Fixes: 05fe62842804 ("ASoC: Intel: soc-acpi-intel-mtl-match: add acpi match table for cdb35l56-eight-c") Link: https://msgid.link/r/20231205115715.2460386-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 301b8142d554..9008b6768205 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -306,13 +306,13 @@ static const struct snd_soc_acpi_adr_device cs35l56_1_adr[] = { .adr = 0x00013701FA355601ull, .num_endpoints = 1, .endpoints = &spk_r_endpoint, - .name_prefix = "cs35l56-8" + .name_prefix = "AMP8" }, { .adr = 0x00013601FA355601ull, .num_endpoints = 1, .endpoints = &spk_3_endpoint, - .name_prefix = "cs35l56-7" + .name_prefix = "AMP7" } }; @@ -321,13 +321,13 @@ static const struct snd_soc_acpi_adr_device cs35l56_2_adr[] = { .adr = 0x00023301FA355601ull, .num_endpoints = 1, .endpoints = &spk_l_endpoint, - .name_prefix = "cs35l56-1" + .name_prefix = "AMP1" }, { .adr = 0x00023201FA355601ull, .num_endpoints = 1, .endpoints = &spk_2_endpoint, - .name_prefix = "cs35l56-2" + .name_prefix = "AMP2" } }; From dc96528b176fa6e55a3dc01060fe9d97be450ce9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 11 Dec 2023 16:00:18 +0000 Subject: [PATCH 55/82] ASoC: cs42l43: Don't enable bias sense during type detect Alas on some headsets the bias sense can cause problems with the type detection. It can occasionally be falsely triggered by the type detect itself and as the clamp is applied when this happens, it will cause a headset to be incorrectly identified as headphones. As such it should be disabled whilst running type detect. This does mean a jack removal during type detect will cause a larger click but that is unfortunately unavoidable. Fixes: 1e4ce0d5c023 ("ASoC: cs42l43: Move headset bias sense enable earlier in process") Signed-off-by: Charles Keepax Link: https://msgid.link/r/20231211160019.2034442-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 73454de068cf..54a3ea606443 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -237,7 +237,7 @@ int cs42l43_set_jack(struct snd_soc_component *component, return ret; } -static void cs42l43_start_hs_bias(struct cs42l43_codec *priv, bool force_high) +static void cs42l43_start_hs_bias(struct cs42l43_codec *priv, bool type_detect) { struct cs42l43 *cs42l43 = priv->core; unsigned int val = 0x3 << CS42L43_HSBIAS_MODE_SHIFT; @@ -247,16 +247,17 @@ static void cs42l43_start_hs_bias(struct cs42l43_codec *priv, bool force_high) regmap_update_bits(cs42l43->regmap, CS42L43_HS2, CS42L43_HS_CLAMP_DISABLE_MASK, CS42L43_HS_CLAMP_DISABLE_MASK); - if (!force_high && priv->bias_low) - val = 0x2 << CS42L43_HSBIAS_MODE_SHIFT; + if (!type_detect) { + if (priv->bias_low) + val = 0x2 << CS42L43_HSBIAS_MODE_SHIFT; - if (priv->bias_sense_ua) { - regmap_update_bits(cs42l43->regmap, - CS42L43_HS_BIAS_SENSE_AND_CLAMP_AUTOCONTROL, - CS42L43_HSBIAS_SENSE_EN_MASK | - CS42L43_AUTO_HSBIAS_CLAMP_EN_MASK, - CS42L43_HSBIAS_SENSE_EN_MASK | - CS42L43_AUTO_HSBIAS_CLAMP_EN_MASK); + if (priv->bias_sense_ua) + regmap_update_bits(cs42l43->regmap, + CS42L43_HS_BIAS_SENSE_AND_CLAMP_AUTOCONTROL, + CS42L43_HSBIAS_SENSE_EN_MASK | + CS42L43_AUTO_HSBIAS_CLAMP_EN_MASK, + CS42L43_HSBIAS_SENSE_EN_MASK | + CS42L43_AUTO_HSBIAS_CLAMP_EN_MASK); } regmap_update_bits(cs42l43->regmap, CS42L43_MIC_DETECT_CONTROL_1, From 6c6fa2641402e8e753262fb61ed9a15a7cb225ad Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Thu, 14 Dec 2023 00:28:16 +0100 Subject: [PATCH 56/82] ALSA: hda/tas2781: call cleanup functions only once If the module can load the RCA but not the firmware binary, it will call the cleanup functions. Then unloading the module causes general protection fault due to double free. Do not call the cleanup functions in tasdev_fw_ready. general protection fault, probably for non-canonical address 0x6f2b8a2bff4c8fec: 0000 [#1] PREEMPT SMP NOPTI Call Trace: ? die_addr+0x36/0x90 ? exc_general_protection+0x1c5/0x430 ? asm_exc_general_protection+0x26/0x30 ? tasdevice_config_info_remove+0x6d/0xd0 [snd_soc_tas2781_fmwlib] tas2781_hda_unbind+0xaa/0x100 [snd_hda_scodec_tas2781_i2c] component_unbind+0x2e/0x50 component_unbind_all+0x92/0xa0 component_del+0xa8/0x140 tas2781_hda_remove.isra.0+0x32/0x60 [snd_hda_scodec_tas2781_i2c] i2c_device_remove+0x26/0xb0 Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/1a0885c424bb21172702d254655882b59ef6477a.1702510018.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index d3dafc9d150b..c8ee5f809c38 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -550,11 +550,6 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tas2781_save_calibration(tas_priv); out: - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - /*If DSP FW fail, kcontrol won't be created */ - tasdevice_config_info_remove(tas_priv); - tasdevice_dsp_remove(tas_priv); - } mutex_unlock(&tas_priv->codec_lock); if (fmw) release_firmware(fmw); From 315deab289924c83ab1ded50022e8db95d6e428b Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Thu, 14 Dec 2023 00:49:20 +0100 Subject: [PATCH 57/82] ALSA: hda/tas2781: reset the amp before component_add Calling component_add starts loading the firmware, the callback function writes the program to the amplifiers. If the module resets the amplifiers after component_add, it happens that one of the amplifiers does not work because the reset and program writing are interleaving. Call tas2781_reset before component_add to ensure reliable initialization. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/4d23bf58558e23ee8097de01f70f1eb8d9de2d15.1702511246.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index c8ee5f809c38..63a90c7e8976 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -674,14 +674,14 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) pm_runtime_put_autosuspend(tas_priv->dev); + tas2781_reset(tas_priv); + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); if (ret) { dev_err(tas_priv->dev, "Register component failed: %d\n", ret); pm_runtime_disable(tas_priv->dev); - goto err; } - tas2781_reset(tas_priv); err: if (ret) tas2781_hda_remove(&clt->dev); From 02a460adfc4920d4da775fb59ab3e54036daef22 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Villeret?= Date: Thu, 14 Dec 2023 21:36:32 +0100 Subject: [PATCH 58/82] ALSA: hda/realtek: Add quirk for ASUS ROG GV302XA MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Asus ROG Flowx13 (GV302XA) seems require same patch as others asus products Signed-off-by: Clément Villeret Cc: Link: https://lore.kernel.org/r/0a27bf4b-3056-49ac-9651-ebd7f3e36328@gmail.com Signed-off-by: Takashi Iwai --- 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 e45d4c405f8f..bbfa64c64d05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9954,6 +9954,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), + SND_PCI_QUIRK(0x1043, 0x1533, "ASUS GV302XA", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), From ec1de5c214eb5a892fdb7c450748249d5e2840f5 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 15 Dec 2023 00:33:27 +0100 Subject: [PATCH 59/82] ALSA: hda/tas2781: select program 0, conf 0 by default Currently, cur_prog/cur_conf remains at the default value (-1), while program 0 has been loaded into the amplifiers. In the playback hook, tasdevice_tuning_switch tries to restore the cur_prog/cur_conf. In the runtime_resume/system_resume, tasdevice_prmg_load tries to load the cur_prog as well. Set cur_prog and cur_conf to 0 if available in the firmware. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/038add0bdca1f979cc7abcce8f24cbcd3544084b.1702596646.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 63a90c7e8976..2fb1a7037c82 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -543,6 +543,10 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; tasdevice_prmg_load(tas_priv, 0); + if (tas_priv->fmw->nr_programs > 0) + tas_priv->cur_prog = 0; + if (tas_priv->fmw->nr_configurations > 0) + tas_priv->cur_conf = 0; /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. From f32c80d34249e1cfb2e647ab3c8ef38a460c787f Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Thu, 14 Dec 2023 23:04:44 +0100 Subject: [PATCH 60/82] ASoC: tas2781: check the validity of prm_no/cfg_no Add additional checks for program/config numbers to avoid loading from invalid addresses. If prm_no/cfg_no is negative, skip uploading program/config. The tas2781-hda driver caused a NULL pointer dereference after loading module, and before first runtime_suspend. the state was: tas_priv->cur_conf = -1; tas_priv->tasdevice[i].cur_conf = 0; program = &(tas_fmw->programs[-1]); BUG: kernel NULL pointer dereference, address: 0000000000000010 Call Trace: ? __die+0x23/0x70 ? page_fault_oops+0x171/0x4e0 ? vprintk_emit+0x175/0x2b0 ? exc_page_fault+0x7f/0x180 ? asm_exc_page_fault+0x26/0x30 ? tasdevice_load_block_kernel+0x21/0x310 [snd_soc_tas2781_fmwlib] tasdevice_select_tuningprm_cfg+0x268/0x3a0 [snd_soc_tas2781_fmwlib] tasdevice_tuning_switch+0x69/0x710 [snd_soc_tas2781_fmwlib] tas2781_hda_playback_hook+0xd4/0x110 [snd_hda_scodec_tas2781_i2c] Fixes: 915f5eadebd2 ("ASoC: tas2781: firmware lib") CC: Signed-off-by: Gergo Koteles Link: https://msgid.link/r/523780155bfdca9bc0acd39efc79ed039454818d.1702591356.git.soyer@irl.hu Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-fmwlib.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 4efe95b60aaa..5c09e441a936 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2189,11 +2189,11 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, goto out; } - conf = &(tas_fmw->configs[cfg_no]); for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { if (cfg_info[rca_conf_no]->active_dev & (1 << i)) { - if (tas_priv->tasdevice[i].cur_prog != prm_no - || tas_priv->force_fwload_status) { + if (prm_no >= 0 + && (tas_priv->tasdevice[i].cur_prog != prm_no + || tas_priv->force_fwload_status)) { tas_priv->tasdevice[i].cur_conf = -1; tas_priv->tasdevice[i].is_loading = true; prog_status++; @@ -2228,7 +2228,8 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, } for (i = 0, status = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].cur_conf != cfg_no + if (cfg_no >= 0 + && tas_priv->tasdevice[i].cur_conf != cfg_no && (cfg_info[rca_conf_no]->active_dev & (1 << i)) && (tas_priv->tasdevice[i].is_loaderr == false)) { status++; @@ -2238,6 +2239,7 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, } if (status) { + conf = &(tas_fmw->configs[cfg_no]); status = 0; tasdevice_load_data(tas_priv, &(conf->dev_data)); for (i = 0; i < tas_priv->ndev; i++) { @@ -2281,7 +2283,7 @@ int tasdevice_prmg_load(void *context, int prm_no) } for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].cur_prog != prm_no) { + 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++; @@ -2326,7 +2328,7 @@ int tasdevice_prmg_calibdata_load(void *context, int prm_no) } for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) { - if (tas_priv->tasdevice[i].cur_prog != prm_no) { + 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++; From 48d6b91798a6694fdd6edb62799754b9d3fe0792 Mon Sep 17 00:00:00 2001 From: Jeremie Knuesel Date: Sun, 17 Dec 2023 12:22:43 +0100 Subject: [PATCH 61/82] ALSA: usb-audio: Increase delay in MOTU M quirk Increase the quirk delay from 2 seconds to 4 seconds. This reflects a change in the Windows driver in which the delay was increased to about 3.7 seconds. The larger delay fixes an issue where the device fails to work unless it was powered up early during boot. Also clarify in the quirk comment that the quirk is only applied to older devices (USB ID 07fd:0008). Signed-off-by: Jeremie Knuesel Suggested-by: Alexander Tsoy Cc: Link: https://bugzilla.kernel.org/show_bug.cgi?id=211975 Link: https://lore.kernel.org/r/20231217112243.33409-1-knuesel@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab2b938502eb..07cc6a201579 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1387,7 +1387,7 @@ static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) { - msleep(2000); + msleep(4000); return 0; } @@ -1630,7 +1630,7 @@ int snd_usb_apply_boot_quirk_once(struct usb_device *dev, unsigned int id) { switch (id) { - case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x07fd, 0x0008): /* MOTU M Series, 1st hardware version */ return snd_usb_motu_m_series_boot_quirk(dev); } From 13d605e32e4cfdedcecdf3d98d21710ffe887708 Mon Sep 17 00:00:00 2001 From: Ghanshyam Agrawal Date: Sun, 17 Dec 2023 13:30:19 +0530 Subject: [PATCH 62/82] kselftest: alsa: fixed a print formatting warning A statement used %d print formatter where %s should have been used. The same has been fixed in this commit. Signed-off-by: Ghanshyam Agrawal Link: 5aaf9efffc57 ("kselftest: alsa: Add simplistic test for ALSA mixer controls kselftest") Link: https://lore.kernel.org/r/20231217080019.1063476-1-ghanshyam1898@gmail.com Signed-off-by: Takashi Iwai --- tools/testing/selftests/alsa/mixer-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 21e482b23f50..23df154fcdd7 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -138,7 +138,7 @@ static void find_controls(void) err = snd_ctl_elem_info(card_data->handle, ctl_data->info); if (err < 0) { - ksft_print_msg("%s getting info for %d\n", + ksft_print_msg("%s getting info for %s\n", snd_strerror(err), ctl_data->name); } From 99c7bb44f5749373bc01b73af02b50b69bcbf43d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 17 Dec 2023 22:32:20 +0100 Subject: [PATCH 63/82] ASoC: Intel: bytcr_rt5640: Add quirk for the Medion Lifetab S10346 Add a quirk for the Medion Lifetab S10346, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://msgid.link/r/20231217213221.49424-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ed14d9e4aa53..ea9e562358b7 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -894,6 +894,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Medion Lifetab S10346 */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are much too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "10/22/2015"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Mele PCG03 Mini PC */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Mini PC"), From b1b6131bca35a55a69fadc39d51577968fa2ee97 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 17 Dec 2023 22:32:21 +0100 Subject: [PATCH 64/82] ASoC: Intel: bytcr_rt5640: Add new swapped-speakers quirk Some BYTCR x86 tablets with a rt5640 codec have the left and right channels of their speakers swapped. Add a new BYT_RT5640_SWAPPED_SPEAKERS quirk for this which sets cfg-spk:swapped in the components string to let userspace know about the swapping so that the UCM profile can configure the mixer to correct this. Enable this new quirk on the Medion Lifetab S10346 which has its speakers swapped. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://msgid.link/r/20231217213221.49424-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ea9e562358b7..42466b4b1ca4 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -83,6 +83,7 @@ enum { #define BYT_RT5640_HSMIC2_ON_IN1 BIT(27) #define BYT_RT5640_JD_HP_ELITEP_1000G2 BIT(28) #define BYT_RT5640_USE_AMCR0F28 BIT(29) +#define BYT_RT5640_SWAPPED_SPEAKERS BIT(30) #define BYTCR_INPUT_DEFAULTS \ (BYT_RT5640_IN3_MAP | \ @@ -157,6 +158,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk MONO_SPEAKER enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS) dev_info(dev, "quirk NO_SPEAKERS enabled\n"); + if (byt_rt5640_quirk & BYT_RT5640_SWAPPED_SPEAKERS) + dev_info(dev, "quirk SWAPPED_SPEAKERS enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_LINEOUT) dev_info(dev, "quirk LINEOUT enabled\n"); if (byt_rt5640_quirk & BYT_RT5640_LINEOUT_AS_HP2) @@ -903,6 +906,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_MATCH(DMI_BIOS_DATE, "10/22/2015"), }, .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_SWAPPED_SPEAKERS | BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, @@ -1631,11 +1635,11 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) const char *platform_name; struct acpi_device *adev; struct device *codec_dev; + const char *cfg_spk; bool sof_parent; int ret_val = 0; int dai_index = 0; - int i, cfg_spk; - int aif; + int i, aif; is_bytcr = false; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -1795,13 +1799,16 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } if (byt_rt5640_quirk & BYT_RT5640_NO_SPEAKERS) { - cfg_spk = 0; + cfg_spk = "0"; spk_type = "none"; } else if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) { - cfg_spk = 1; + cfg_spk = "1"; spk_type = "mono"; + } else if (byt_rt5640_quirk & BYT_RT5640_SWAPPED_SPEAKERS) { + cfg_spk = "swapped"; + spk_type = "swapped"; } else { - cfg_spk = 2; + cfg_spk = "2"; spk_type = "stereo"; } @@ -1816,7 +1823,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) headset2_string = " cfg-hs2:in1"; snprintf(byt_rt5640_components, sizeof(byt_rt5640_components), - "cfg-spk:%d cfg-mic:%s aif:%d%s%s", cfg_spk, + "cfg-spk:%s cfg-mic:%s aif:%d%s%s", cfg_spk, map_name[BYT_RT5640_MAP(byt_rt5640_quirk)], aif, lineout_string, headset2_string); byt_rt5640_card.components = byt_rt5640_components; From 8c4c216db8fb84be9c4ca60d72b88882066cf28f Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 18 Dec 2023 15:12:15 +0000 Subject: [PATCH 65/82] ALSA: hda: cs35l41: Add config table to support many laptops without _DSD This make use of the CS35L41 HDA Property framework, which supports laptops which do not have the _DSD properties in their ACPI. Add configuration table to be able to use a generic function which allows laptops to be supported just by adding an entry into the table. Use configuration table function for existing system 103C89C6. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231218151221.388745-2-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 2 + sound/pci/hda/cs35l41_hda.h | 5 +- sound/pci/hda/cs35l41_hda_property.c | 304 +++++++++++++++++++++++---- 3 files changed, 269 insertions(+), 42 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index cbd7d8badf91..92ca2b3b6c92 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1826,6 +1826,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i if (cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type)) gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); gpiod_put(cs35l41->reset_gpio); + gpiod_put(cs35l41->cs_gpio); acpi_dev_put(cs35l41->dacpi); kfree(cs35l41->acpi_subsystem_id); @@ -1853,6 +1854,7 @@ void cs35l41_hda_remove(struct device *dev) if (cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type)) gpiod_set_value_cansleep(cs35l41->reset_gpio, 0); gpiod_put(cs35l41->reset_gpio); + gpiod_put(cs35l41->cs_gpio); kfree(cs35l41->acpi_subsystem_id); } EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index ce3f2bb6ffd0..3d925d677213 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -35,8 +35,8 @@ struct cs35l41_amp_efi_data { } __packed; enum cs35l41_hda_spk_pos { - CS35l41_LEFT, - CS35l41_RIGHT, + CS35L41_LEFT, + CS35L41_RIGHT, }; enum cs35l41_hda_gpio_function { @@ -50,6 +50,7 @@ struct cs35l41_hda { struct device *dev; struct regmap *regmap; struct gpio_desc *reset_gpio; + struct gpio_desc *cs_gpio; struct cs35l41_hw_cfg hw_cfg; struct hda_codec *codec; diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index c83328971728..f90423ded85d 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -6,9 +6,271 @@ // // Author: Stefan Binding +#include #include #include #include "cs35l41_hda_property.h" +#include + +#define MAX_AMPS 4 + +struct cs35l41_config { + const char *ssid; + enum { + SPI, + I2C + } bus; + int num_amps; + enum { + INTERNAL, + EXTERNAL + } boost_type; + u8 channel[MAX_AMPS]; + int reset_gpio_index; /* -1 if no reset gpio */ + int spkid_gpio_index; /* -1 if no spkid gpio */ + int cs_gpio_index; /* -1 if no cs gpio, or cs-gpios already exists, max num amps == 2 */ + int boost_ind_nanohenry; /* Required if boost_type == Internal */ + int boost_peak_milliamp; /* Required if boost_type == Internal */ + int boost_cap_microfarad; /* Required if boost_type == Internal */ +}; + +static const struct cs35l41_config cs35l41_config_table[] = { +/* + * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. + * We can override the _DSD to correct the boost type here. + * Since this laptop has valid ACPI, we do not need to handle cs-gpios, since that already exists + * in the ACPI. The Reset GPIO is also valid, so we can use the Reset defined in _DSD. + */ + { "103C89C6", SPI, 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, + {} +}; + +static int cs35l41_add_gpios(struct cs35l41_hda *cs35l41, struct device *physdev, int reset_gpio, + int spkid_gpio, int cs_gpio_index, int num_amps) +{ + struct acpi_gpio_mapping *gpio_mapping; + struct acpi_gpio_params *reset_gpio_params; + struct acpi_gpio_params *spkid_gpio_params; + struct acpi_gpio_params *cs_gpio_params; + unsigned int num_entries = 0; + unsigned int reset_index, spkid_index, csgpio_index; + int i; + + /* + * GPIO Mapping only needs to be done once, since it would be available for subsequent amps + */ + if (cs35l41->dacpi->driver_gpios) + return 0; + + if (reset_gpio >= 0) { + reset_index = num_entries; + num_entries++; + } + + if (spkid_gpio >= 0) { + spkid_index = num_entries; + num_entries++; + } + + if ((cs_gpio_index >= 0) && (num_amps == 2)) { + csgpio_index = num_entries; + num_entries++; + } + + if (!num_entries) + return 0; + + /* must include termination entry */ + num_entries++; + + gpio_mapping = devm_kcalloc(physdev, num_entries, sizeof(struct acpi_gpio_mapping), + GFP_KERNEL); + + if (!gpio_mapping) + goto err; + + if (reset_gpio >= 0) { + gpio_mapping[reset_index].name = "reset-gpios"; + reset_gpio_params = devm_kcalloc(physdev, num_amps, sizeof(struct acpi_gpio_params), + GFP_KERNEL); + if (!reset_gpio_params) + goto err; + + for (i = 0; i < num_amps; i++) + reset_gpio_params[i].crs_entry_index = reset_gpio; + + gpio_mapping[reset_index].data = reset_gpio_params; + gpio_mapping[reset_index].size = num_amps; + } + + if (spkid_gpio >= 0) { + gpio_mapping[spkid_index].name = "spk-id-gpios"; + spkid_gpio_params = devm_kcalloc(physdev, num_amps, sizeof(struct acpi_gpio_params), + GFP_KERNEL); + if (!spkid_gpio_params) + goto err; + + for (i = 0; i < num_amps; i++) + spkid_gpio_params[i].crs_entry_index = spkid_gpio; + + gpio_mapping[spkid_index].data = spkid_gpio_params; + gpio_mapping[spkid_index].size = num_amps; + } + + if ((cs_gpio_index >= 0) && (num_amps == 2)) { + gpio_mapping[csgpio_index].name = "cs-gpios"; + /* only one GPIO CS is supported without using _DSD, obtained using index 0 */ + cs_gpio_params = devm_kzalloc(physdev, sizeof(struct acpi_gpio_params), GFP_KERNEL); + if (!cs_gpio_params) + goto err; + + cs_gpio_params->crs_entry_index = cs_gpio_index; + + gpio_mapping[csgpio_index].data = cs_gpio_params; + gpio_mapping[csgpio_index].size = 1; + } + + return devm_acpi_dev_add_driver_gpios(physdev, gpio_mapping); +err: + devm_kfree(physdev, gpio_mapping); + devm_kfree(physdev, reset_gpio_params); + devm_kfree(physdev, spkid_gpio_params); + devm_kfree(physdev, cs_gpio_params); + return -ENOMEM; +} + +static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + const struct cs35l41_config *cfg; + struct gpio_desc *cs_gpiod; + struct spi_device *spi; + bool dsd_found; + int ret; + + for (cfg = cs35l41_config_table; cfg->ssid; cfg++) { + if (!strcasecmp(cfg->ssid, cs35l41->acpi_subsystem_id)) + break; + } + + if (!cfg->ssid) + return -ENOENT; + + if (!cs35l41->dacpi || cs35l41->dacpi != ACPI_COMPANION(physdev)) { + dev_err(cs35l41->dev, "ACPI Device does not match, cannot override _DSD.\n"); + return -ENODEV; + } + + dev_info(cs35l41->dev, "Adding DSD properties for %s\n", cs35l41->acpi_subsystem_id); + + dsd_found = acpi_dev_has_props(cs35l41->dacpi); + + if (!dsd_found) { + ret = cs35l41_add_gpios(cs35l41, physdev, cfg->reset_gpio_index, + cfg->spkid_gpio_index, cfg->cs_gpio_index, + cfg->num_amps); + if (ret) { + dev_err(cs35l41->dev, "Error adding GPIO mapping: %d\n", ret); + return ret; + } + } else if (cfg->reset_gpio_index >= 0 || cfg->spkid_gpio_index >= 0) { + dev_warn(cs35l41->dev, "Cannot add Reset/Speaker ID/SPI CS GPIO Mapping, " + "_DSD already exists.\n"); + } + + if (cfg->bus == SPI) { + cs35l41->index = id; + /* + * Manually set the Chip Select for the second amp in the node. + * This is only supported for systems with 2 amps, since we cannot expand the + * default number of chip selects without using cs-gpios + * The CS GPIO must be set high prior to communicating with the first amp (which + * uses a native chip select), to ensure the second amp does not clash with the + * first. + */ + if (cfg->cs_gpio_index >= 0) { + spi = to_spi_device(cs35l41->dev); + + if (cfg->num_amps != 2) { + dev_warn(cs35l41->dev, + "Cannot update SPI CS, Number of Amps (%d) != 2\n", + cfg->num_amps); + } else if (dsd_found) { + dev_warn(cs35l41->dev, + "Cannot update SPI CS, _DSD already exists.\n"); + } else { + /* + * This is obtained using driver_gpios, since only one GPIO for CS + * exists, this can be obtained using index 0. + */ + cs_gpiod = gpiod_get_index(physdev, "cs", 0, GPIOD_OUT_LOW); + if (IS_ERR(cs_gpiod)) { + dev_err(cs35l41->dev, + "Unable to get Chip Select GPIO descriptor\n"); + return PTR_ERR(cs_gpiod); + } + if (id == 1) { + spi_set_csgpiod(spi, 0, cs_gpiod); + cs35l41->cs_gpio = cs_gpiod; + } else { + gpiod_set_value_cansleep(cs_gpiod, true); + gpiod_put(cs_gpiod); + } + spi_setup(spi); + } + } + } else { + if (cfg->num_amps > 2) + /* + * i2c addresses for 3/4 amps are used in order: 0x40, 0x41, 0x42, 0x43, + * subtracting 0x40 would give zero-based index + */ + cs35l41->index = id - 0x40; + else + /* i2c addr 0x40 for first amp (always), 0x41/0x42 for 2nd amp */ + cs35l41->index = id == 0x40 ? 0 : 1; + } + + if (cfg->num_amps == 3) + /* 3 amps means a center channel, so no duplicate channels */ + cs35l41->channel_index = 0; + else + /* + * if 4 amps, there are duplicate channels, so they need different indexes + * if 2 amps, no duplicate channels, channel_index would be 0 + */ + cs35l41->channel_index = cs35l41->index / 2; + + cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(cs35l41->dacpi), "reset", + cs35l41->index, GPIOD_OUT_LOW, + "cs35l41-reset"); + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, cs35l41->index, cfg->num_amps, -1); + + hw_cfg->spk_pos = cfg->channel[cs35l41->index]; + + if (cfg->boost_type == INTERNAL) { + hw_cfg->bst_type = CS35L41_INT_BOOST; + hw_cfg->bst_ind = cfg->boost_ind_nanohenry; + hw_cfg->bst_ipk = cfg->boost_peak_milliamp; + hw_cfg->bst_cap = cfg->boost_cap_microfarad; + hw_cfg->gpio1.func = CS35L41_NOT_USED; + hw_cfg->gpio1.valid = true; + } else { + hw_cfg->bst_type = CS35L41_EXT_BOOST; + hw_cfg->bst_ind = -1; + hw_cfg->bst_ipk = -1; + hw_cfg->bst_cap = -1; + hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; + hw_cfg->gpio1.valid = true; + } + + hw_cfg->gpio2.func = CS35L41_INTERRUPT; + hw_cfg->gpio2.valid = true; + hw_cfg->valid = true; + + return 0; +} /* * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. @@ -43,44 +305,6 @@ static int lenovo_legion_no_acpi(struct cs35l41_hda *cs35l41, struct device *phy return 0; } -/* - * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. - * We can override the _DSD to correct the boost type here. - * Since this laptop has valid ACPI, we do not need to handle cs-gpios, since that already exists - * in the ACPI. - */ -static int hp_vision_acpi_fix(struct cs35l41_hda *cs35l41, struct device *physdev, int id, - const char *hid) -{ - struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; - - dev_info(cs35l41->dev, "Adding DSD properties for %s\n", cs35l41->acpi_subsystem_id); - - cs35l41->index = id; - cs35l41->channel_index = 0; - - /* - * This system has _DSD, it just contains an error, so we can still get the reset using - * the "reset" label. - */ - cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(cs35l41->dacpi), "reset", - cs35l41->index, GPIOD_OUT_LOW, - "cs35l41-reset"); - cs35l41->speaker_id = -ENOENT; - hw_cfg->spk_pos = cs35l41->index ? 0 : 1; // right:left - hw_cfg->gpio1.func = CS35L41_NOT_USED; - hw_cfg->gpio1.valid = true; - hw_cfg->gpio2.func = CS35L41_INTERRUPT; - hw_cfg->gpio2.valid = true; - hw_cfg->bst_type = CS35L41_INT_BOOST; - hw_cfg->bst_ind = 1000; - hw_cfg->bst_ipk = 4500; - hw_cfg->bst_cap = 24; - hw_cfg->valid = true; - - return 0; -} - struct cs35l41_prop_model { const char *hid; const char *ssid; @@ -91,7 +315,7 @@ struct cs35l41_prop_model { static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, - { "CSC3551", "103C89C6", hp_vision_acpi_fix }, + { "CSC3551", "103C89C6", generic_dsd_config }, {} }; @@ -104,7 +328,7 @@ int cs35l41_add_dsd_properties(struct cs35l41_hda *cs35l41, struct device *physd if (!strcmp(model->hid, hid) && (!model->ssid || (cs35l41->acpi_subsystem_id && - !strcmp(model->ssid, cs35l41->acpi_subsystem_id)))) + !strcasecmp(model->ssid, cs35l41->acpi_subsystem_id)))) return model->add_prop(cs35l41, physdev, id, hid); } From b592ed2e1d78a475f781802e441c499ab446975b Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 18 Dec 2023 15:12:16 +0000 Subject: [PATCH 66/82] ALSA: hda: cs35l41: Support additional ASUS ROG 2023 models Add new model entries into configuration table. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231218151221.388745-3-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 36 ++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index f90423ded85d..a0d808ed640a 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -42,6 +42,24 @@ static const struct cs35l41_config cs35l41_config_table[] = { * in the ACPI. The Reset GPIO is also valid, so we can use the Reset defined in _DSD. */ { "103C89C6", SPI, 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, + { "10431433", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431463", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431473", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431483", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431493", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104314D3", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104314E3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431503", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431533", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431573", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431663", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "104317F3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431C9F", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CAF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CCF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CDF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CEF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431D1F", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, {} }; @@ -316,6 +334,24 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, { "CSC3551", "103C89C6", generic_dsd_config }, + { "CSC3551", "10431433", generic_dsd_config }, + { "CSC3551", "10431463", generic_dsd_config }, + { "CSC3551", "10431473", generic_dsd_config }, + { "CSC3551", "10431483", generic_dsd_config }, + { "CSC3551", "10431493", generic_dsd_config }, + { "CSC3551", "104314D3", generic_dsd_config }, + { "CSC3551", "104314E3", generic_dsd_config }, + { "CSC3551", "10431503", generic_dsd_config }, + { "CSC3551", "10431533", generic_dsd_config }, + { "CSC3551", "10431573", generic_dsd_config }, + { "CSC3551", "10431663", generic_dsd_config }, + { "CSC3551", "104317F3", generic_dsd_config }, + { "CSC3551", "10431C9F", generic_dsd_config }, + { "CSC3551", "10431CAF", generic_dsd_config }, + { "CSC3551", "10431CCF", generic_dsd_config }, + { "CSC3551", "10431CDF", generic_dsd_config }, + { "CSC3551", "10431CEF", generic_dsd_config }, + { "CSC3551", "10431D1F", generic_dsd_config }, {} }; From a40ce9f4bdbebfbf55fdd83a5284fbaaf222f0b9 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 18 Dec 2023 15:12:17 +0000 Subject: [PATCH 67/82] ALSA: hda/realtek: Add quirks for ASUS ROG 2023 models These models use 2xCS35L41amps with HDA using SPI and I2C. All models use Internal Boost. Some models also use Realtek Speakers in conjunction with CS35L41. All models require DSD support to be added inside cs35l41_hda_property.c Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231218151221.388745-4-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bbfa64c64d05..9c3de6a80e73 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9948,22 +9948,25 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), - SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650P", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), - SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), - SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), - SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), - SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X/GA402N", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604VI/VC/VE/VG/VJ/VQ/VU/VV/VY/VZ", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603VQ/VU/VV/VJ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601VV/VU/VJ/VQ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G614JY/JZ/JG", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS G513PI/PU/PV", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1503, "ASUS G733PY/PZ/PZV/PYV", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), - SND_PCI_QUIRK(0x1043, 0x1533, "ASUS GV302XA", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301V", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1533, "ASUS GV302XA/XJ/XQ/XU/XV/XI", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301VV/VQ/VU/VJ/VA/VC/VE/VVC/VQC/VUC/VJC/VEC/VCC", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), - SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZI/ZJ/ZQ/ZU/ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), 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 RC71L_RC71L", ALC294_FIXUP_ASUS_ALLY), + SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), 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), SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS UM3504DA", ALC294_FIXUP_CS35L41_I2C_2), @@ -9988,10 +9991,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), - SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JI", ALC285_FIXUP_ASUS_HEADSET_MIC), - SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), + SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), - SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS ROG Strix G17 2023 (G713PV)", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1ccf, "ASUS G814JU/JV/JI", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1cdf, "ASUS G814JY/JZ/JG", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1cef, "ASUS G834JY/JZ/JI/JG", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS G713PI/PU/PV/PVN", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), From b257187bcff4bccc9e7a8f1b8a1a5526ff815af1 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 18 Dec 2023 15:12:18 +0000 Subject: [PATCH 68/82] ALSA: hda: cs35l41: Support additional ASUS Zenbook 2022 Models Add new model entries into configuration table. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231218151221.388745-5-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index a0d808ed640a..07fe72bb128a 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -42,6 +42,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { * in the ACPI. The Reset GPIO is also valid, so we can use the Reset defined in _DSD. */ { "103C89C6", SPI, 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, + { "104312AF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431433", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431463", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431473", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, @@ -60,6 +61,11 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431CDF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431CEF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431D1F", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431DA2", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10431E02", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10431EE2", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "10431F12", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431F62", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, {} }; @@ -334,6 +340,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, { "CSC3551", "103C89C6", generic_dsd_config }, + { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, { "CSC3551", "10431463", generic_dsd_config }, { "CSC3551", "10431473", generic_dsd_config }, @@ -352,6 +359,11 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431CDF", generic_dsd_config }, { "CSC3551", "10431CEF", generic_dsd_config }, { "CSC3551", "10431D1F", generic_dsd_config }, + { "CSC3551", "10431DA2", generic_dsd_config }, + { "CSC3551", "10431E02", generic_dsd_config }, + { "CSC3551", "10431EE2", generic_dsd_config }, + { "CSC3551", "10431F12", generic_dsd_config }, + { "CSC3551", "10431F62", generic_dsd_config }, {} }; From 51d976079976c800ef19ed1b542602fcf63f0edb Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 18 Dec 2023 15:12:19 +0000 Subject: [PATCH 69/82] ALSA: hda/realtek: Add quirks for ASUS Zenbook 2022 Models These models use 2xCS35L41amps with HDA using SPI and I2C. Models use internal and external boost. All models require DSD support to be added inside cs35l41_hda_property.c Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231218151221.388745-6-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c3de6a80e73..66652320822f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10000,17 +10000,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS G713PI/PU/PV/PVN", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS UX3402VA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1f62, "ASUS UX7602ZM", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), - SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1f62, "ASUS UX7602ZM", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC245_FIXUP_CS35L41_SPI_2), From 2b35b66d82dc4641ba60f7f3c36c0040eedb74e2 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 18 Dec 2023 15:12:20 +0000 Subject: [PATCH 70/82] ALSA: hda: cs35l41: Support additional ASUS Zenbook 2023 Models Add new model entries into configuration table. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231218151221.388745-7-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 07fe72bb128a..c9eb70290973 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -54,7 +54,11 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431533", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431573", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431663", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "104316D3", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "104316F3", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "104317F3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431863", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104318D3", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "10431C9F", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431CAF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431CCF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, @@ -65,6 +69,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431E02", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "10431EE2", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "10431F12", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431F1F", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, {} }; @@ -352,7 +357,11 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431533", generic_dsd_config }, { "CSC3551", "10431573", generic_dsd_config }, { "CSC3551", "10431663", generic_dsd_config }, + { "CSC3551", "104316D3", generic_dsd_config }, + { "CSC3551", "104316F3", generic_dsd_config }, { "CSC3551", "104317F3", generic_dsd_config }, + { "CSC3551", "10431863", generic_dsd_config }, + { "CSC3551", "104318D3", generic_dsd_config }, { "CSC3551", "10431C9F", generic_dsd_config }, { "CSC3551", "10431CAF", generic_dsd_config }, { "CSC3551", "10431CCF", generic_dsd_config }, @@ -363,6 +372,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431E02", generic_dsd_config }, { "CSC3551", "10431EE2", generic_dsd_config }, { "CSC3551", "10431F12", generic_dsd_config }, + { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, {} }; From ae53e2198cb811f7ee7c5cd4580bf42e88086fa5 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 18 Dec 2023 15:12:21 +0000 Subject: [PATCH 71/82] ALSA: hda/realtek: Add quirks for ASUS Zenbook 2023 Models These models use 2xCS35L41amps with HDA using SPI and I2C. Models use internal and external boost. All models require DSD support to be added inside cs35l41_hda_property.c Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231218151221.388745-8-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 66652320822f..c3a756528886 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9963,10 +9963,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZI/ZJ/ZQ/ZU/ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x16d3, "ASUS UX5304VA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS UX7602VI/BZ", ALC245_FIXUP_CS35L41_SPI_2), 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, 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), SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS UM3504DA", ALC294_FIXUP_CS35L41_I2C_2), @@ -10013,6 +10016,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1f1f, "ASUS H7604JI/JV/J3D", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1f62, "ASUS UX7602ZM", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), From 025222a9d6d25eee2ad9a1bb5a8b29b34b5ba576 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 18 Dec 2023 15:56:52 +0100 Subject: [PATCH 72/82] ASoC: hdmi-codec: fix missing report for jack initial status This fixes a problem introduced while fixing ELD reporting with no jack set. Most driver using the hdmi-codec will call the 'plugged_cb' callback directly when registered to report the initial state of the HDMI connector. With the commit mentionned, this occurs before jack is ready and the initial report is lost for platforms actually providing a jack for HDMI. Fix this by storing the hdmi connector status regardless of jack being set or not and report the last status when jack gets set. With this, the initial state is reported correctly even if it is disconnected. This was not done initially and is also a fix. Fixes: 15be353d55f9 ("ASoC: hdmi-codec: register hpd callback on component probe") Reported-by: Zhengqiao Xia Closes: https://lore.kernel.org/alsa-devel/CADYyEwTNyY+fR9SgfDa-g6iiDwkU3MUdPVCYexs2_3wbcM8_vg@mail.gmail.com/ Cc: Hsin-Yi Wang Tested-by: Zhengqiao Xia Signed-off-by: Jerome Brunet Link: https://msgid.link/r/20231218145655.134929-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 20da1eaa4f1c..0938671700c6 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -850,8 +850,9 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai) static void hdmi_codec_jack_report(struct hdmi_codec_priv *hcp, unsigned int jack_status) { - if (hcp->jack && jack_status != hcp->jack_status) { - snd_soc_jack_report(hcp->jack, jack_status, SND_JACK_LINEOUT); + if (jack_status != hcp->jack_status) { + if (hcp->jack) + snd_soc_jack_report(hcp->jack, jack_status, SND_JACK_LINEOUT); hcp->jack_status = jack_status; } } @@ -880,6 +881,13 @@ static int hdmi_codec_set_jack(struct snd_soc_component *component, if (hcp->hcd.ops->hook_plugged_cb) { hcp->jack = jack; + + /* + * Report the initial jack status which may have been provided + * by the parent hdmi driver while the hpd hook was registered. + */ + snd_soc_jack_report(jack, hcp->jack_status, SND_JACK_LINEOUT); + return 0; } From 8f0f01647550daf9cd8752c1656dcb0136d79ce1 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 19 Dec 2023 10:30:57 +0800 Subject: [PATCH 73/82] ASoC: fsl_sai: Fix channel swap issue on i.MX8MP When flag mclk_with_tere and mclk_direction_output enabled, The SAI transmitter or receiver will be enabled in very early stage, that if FSL_SAI_xMR is set by previous case, for example previous case is one channel, current case is two channels, then current case started with wrong xMR in the beginning, then channel swap happen. The patch is to clear xMR in hw_free() to avoid such channel swap issue. Fixes: 3e4a82612998 ("ASoC: fsl_sai: MCLK bind with TX/RX enable bit") Signed-off-by: Shengjiu Wang Reviewed-by: Daniel Baluta Link: https://msgid.link/r/1702953057-4499-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 32bbe5056a63..546bd4e333b5 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -714,6 +714,9 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; unsigned int ofs = sai->soc_data->reg_offset; + /* Clear xMR to avoid channel swap with mclk_with_tere enabled case */ + regmap_write(sai->regmap, FSL_SAI_xMR(tx), 0); + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, 0); From ed7326a24a1a9af65fafefd86b505e7c3b968f6d Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 19 Dec 2023 16:22:31 +0000 Subject: [PATCH 74/82] ALSA: hda: cs35l41: Do not allow uninitialised variables to be freed Initialise the variables to NULL so that they cannot be uninitialised when devm_kfree is called. Found by static analysis. Fixes: 8c4c216db8fb ("ALSA: hda: cs35l41: Add config table to support many laptops without _DSD") Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231219162232.790358-2-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index c9eb70290973..73b304e6c83c 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -77,10 +77,10 @@ static const struct cs35l41_config cs35l41_config_table[] = { static int cs35l41_add_gpios(struct cs35l41_hda *cs35l41, struct device *physdev, int reset_gpio, int spkid_gpio, int cs_gpio_index, int num_amps) { - struct acpi_gpio_mapping *gpio_mapping; - struct acpi_gpio_params *reset_gpio_params; - struct acpi_gpio_params *spkid_gpio_params; - struct acpi_gpio_params *cs_gpio_params; + struct acpi_gpio_mapping *gpio_mapping = NULL; + struct acpi_gpio_params *reset_gpio_params = NULL; + struct acpi_gpio_params *spkid_gpio_params = NULL; + struct acpi_gpio_params *cs_gpio_params = NULL; unsigned int num_entries = 0; unsigned int reset_index, spkid_index, csgpio_index; int i; From 916d051730ae48aef8b588fd096fefca4bc0590a Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 19 Dec 2023 16:22:32 +0000 Subject: [PATCH 75/82] ALSA: hda: cs35l41: Only add SPI CS GPIO if SPI is enabled in kernel If CONFIG_SPI is not set in the kernel, there is no point in trying to set the chip selects. We can selectively compile it. Fixes: 8c4c216db8fb ("ALSA: hda: cs35l41: Add config table to support many laptops without _DSD") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202312192256.lJelQEoZ-lkp@intel.com/ Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20231219162232.790358-3-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 73b304e6c83c..194e1179a253 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -210,6 +210,8 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde if (cfg->bus == SPI) { cs35l41->index = id; + +#if IS_ENABLED(CONFIG_SPI) /* * Manually set the Chip Select for the second amp in the node. * This is only supported for systems with 2 amps, since we cannot expand the @@ -249,6 +251,7 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde spi_setup(spi); } } +#endif } else { if (cfg->num_amps > 2) /* From 6dad45f4d28977bd1948973107cf325d431e5b7e Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 22 Dec 2023 00:48:56 +0100 Subject: [PATCH 76/82] ALSA: hda/tas2781: do not use regcache There are two problems with using regcache in this module. The amplifier has 3 addressing levels (BOOK, PAGE, REG). The firmware contains blocks that must be written to BOOK 0x8C. The regcache doesn't know anything about BOOK, so regcache_sync writes invalid values to the actual BOOK. The module handles 2 or more separate amplifiers. The amplifiers have different register values, and the module uses only one regmap/regcache for all the amplifiers. The regcache_sync only writes the last amplifier used. The module successfully restores all the written register values (RC profile, program, configuration, calibration) without regcache. Remove regcache functions and set regmap cache_type to REGCACHE_NONE. Link: https://lore.kernel.org/r/21a183b5a08cb23b193af78d4b1114cc59419272.1701906455.git.soyer@irl.hu/ Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Acked-by: Mark Brown CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/491aeed0e2eecc3b704ec856f815db21bad3ba0e.1703202126.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 17 +---------------- sound/soc/codecs/tas2781-comlib.c | 2 +- 2 files changed, 2 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 2fb1a7037c82..e4c54b2a012c 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -717,8 +717,6 @@ static int tas2781_runtime_suspend(struct device *dev) tas_priv->tasdevice[i].cur_conf = -1; } - regcache_cache_only(tas_priv->regmap, true); - regcache_mark_dirty(tas_priv->regmap); mutex_unlock(&tas_priv->codec_lock); @@ -730,20 +728,11 @@ static int tas2781_runtime_resume(struct device *dev) struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); unsigned long calib_data_sz = tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; - int ret; dev_dbg(tas_priv->dev, "Runtime Resume\n"); mutex_lock(&tas_priv->codec_lock); - regcache_cache_only(tas_priv->regmap, false); - ret = regcache_sync(tas_priv->regmap); - if (ret) { - dev_err(tas_priv->dev, - "Failed to restore register cache: %d\n", ret); - goto out; - } - tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); /* If calibrated data occurs error, dsp will still works with default @@ -752,10 +741,9 @@ static int tas2781_runtime_resume(struct device *dev) if (tas_priv->cali_data.total_sz > calib_data_sz) tas2781_apply_calib(tas_priv); -out: mutex_unlock(&tas_priv->codec_lock); - return ret; + return 0; } static int tas2781_system_suspend(struct device *dev) @@ -770,10 +758,7 @@ static int tas2781_system_suspend(struct device *dev) return ret; /* Shutdown chip before system suspend */ - regcache_cache_only(tas_priv->regmap, false); tasdevice_tuning_switch(tas_priv, 1); - regcache_cache_only(tas_priv->regmap, true); - regcache_mark_dirty(tas_priv->regmap); /* * Reset GPIO may be shared, so cannot reset here. diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index ffb26e4a7e2f..933cd008e9f5 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -39,7 +39,7 @@ static const struct regmap_range_cfg tasdevice_ranges[] = { static const struct regmap_config tasdevice_regmap = { .reg_bits = 8, .val_bits = 8, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_NONE, .ranges = tasdevice_ranges, .num_ranges = ARRAY_SIZE(tasdevice_ranges), .max_register = 256 * 128, From a0c9f7f2e0a46554737509459552789300a4e854 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 22 Dec 2023 01:11:54 +0100 Subject: [PATCH 77/82] ALSA: hda/tas2781: fix typos in comment Correct typos. Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/ead5609d63e71e8e87c13e1767decca5b272d696.1703203812.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index e4c54b2a012c..54d135405788 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -421,9 +421,9 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) } } -/* Update the calibrate data, including speaker impedance, f0, etc, into algo. +/* Update the calibration data, including speaker impedance, f0, etc, into algo. * Calibrate data is done by manufacturer in the factory. These data are used - * by Algo for calucating the speaker temperature, speaker membrance excursion + * by Algo for calculating the speaker temperature, speaker membrane excursion * and f0 in real time during playback. */ static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) From e7aa105657f7f62f54a493480588895cc8a9a1a7 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 22 Dec 2023 01:34:47 +0100 Subject: [PATCH 78/82] ALSA: hda/tas2781: move set_drv_data outside tasdevice_init allow driver specific driver data in tas2781-hda-i2c and tas2781-i2c Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/1398bd8bf3e935b1595a99128320e4a1913e210a.1703204848.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 2 ++ sound/soc/codecs/tas2781-comlib.c | 2 -- sound/soc/codecs/tas2781-i2c.c | 2 ++ 3 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 54d135405788..6d11bced78f0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -659,6 +659,8 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) if (!tas_priv) return -ENOMEM; + dev_set_drvdata(&clt->dev, tas_priv); + tas_priv->irq_info.irq = clt->irq; ret = tas2781_read_acpi(tas_priv, device_name); if (ret) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 933cd008e9f5..00e35169ae49 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -316,8 +316,6 @@ int tasdevice_init(struct tasdevice_priv *tas_priv) tas_priv->tasdevice[i].cur_conf = -1; } - dev_set_drvdata(tas_priv->dev, tas_priv); - mutex_init(&tas_priv->codec_lock); out: diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 55cd5e3c23a5..917b1c15f71d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -689,6 +689,8 @@ static int tasdevice_i2c_probe(struct i2c_client *i2c) if (!tas_priv) return -ENOMEM; + dev_set_drvdata(&i2c->dev, tas_priv); + if (ACPI_HANDLE(&i2c->dev)) { acpi_id = acpi_match_device(i2c->dev.driver->acpi_match_table, &i2c->dev); From 4e7914eb1dae377b8e6de59c96b0653aacb47646 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Fri, 22 Dec 2023 01:34:48 +0100 Subject: [PATCH 79/82] ALSA: hda/tas2781: remove sound controls in unbind Remove sound controls in hda_unbind to make module loadable after module unload. Add a driver specific struct (tas2781_hda) to store the controls. This patch depends on patch: ALSA: hda/tas2781: do not use regcache Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") CC: stable@vger.kernel.org Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/362aa3e2f81b9259a3e5222f576bec5debfc5e88.1703204848.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 223 +++++++++++++++++++------------- 1 file changed, 130 insertions(+), 93 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 6d11bced78f0..dfe281b57aa6 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -65,6 +65,15 @@ enum calib_data { CALIB_MAX }; +struct tas2781_hda { + struct device *dev; + struct tasdevice_priv *priv; + struct snd_kcontrol *dsp_prog_ctl; + struct snd_kcontrol *dsp_conf_ctl; + struct snd_kcontrol *prof_ctl; + struct snd_kcontrol *snd_ctls[3]; +}; + static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) { struct tasdevice_priv *tas_priv = data; @@ -125,26 +134,26 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) static void tas2781_hda_playback_hook(struct device *dev, int action) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - dev_dbg(tas_priv->dev, "%s: action = %d\n", __func__, action); + dev_dbg(tas_hda->dev, "%s: action = %d\n", __func__, action); switch (action) { case HDA_GEN_PCM_ACT_OPEN: pm_runtime_get_sync(dev); - mutex_lock(&tas_priv->codec_lock); - tasdevice_tuning_switch(tas_priv, 0); - mutex_unlock(&tas_priv->codec_lock); + mutex_lock(&tas_hda->priv->codec_lock); + tasdevice_tuning_switch(tas_hda->priv, 0); + mutex_unlock(&tas_hda->priv->codec_lock); break; case HDA_GEN_PCM_ACT_CLOSE: - mutex_lock(&tas_priv->codec_lock); - tasdevice_tuning_switch(tas_priv, 1); - mutex_unlock(&tas_priv->codec_lock); + mutex_lock(&tas_hda->priv->codec_lock); + tasdevice_tuning_switch(tas_hda->priv, 1); + mutex_unlock(&tas_hda->priv->codec_lock); pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); break; default: - dev_dbg(tas_priv->dev, "Playback action not supported: %d\n", + dev_dbg(tas_hda->dev, "Playback action not supported: %d\n", action); break; } @@ -477,9 +486,28 @@ static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) return 0; } +static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) +{ + struct hda_codec *codec = tas_hda->priv->codec; + + if (tas_hda->dsp_prog_ctl) + snd_ctl_remove(codec->card, tas_hda->dsp_prog_ctl); + + if (tas_hda->dsp_conf_ctl) + snd_ctl_remove(codec->card, tas_hda->dsp_conf_ctl); + + for (int i = ARRAY_SIZE(tas_hda->snd_ctls) - 1; i >= 0; i--) + if (tas_hda->snd_ctls[i]) + snd_ctl_remove(codec->card, tas_hda->snd_ctls[i]); + + if (tas_hda->prof_ctl) + snd_ctl_remove(codec->card, tas_hda->prof_ctl); +} + static void tasdev_fw_ready(const struct firmware *fmw, void *context) { struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); struct hda_codec *codec = tas_priv->codec; int i, ret; @@ -490,8 +518,8 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) if (ret) goto out; - ret = snd_ctl_add(codec->card, - snd_ctl_new1(&tas2781_prof_ctrl, tas_priv)); + tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl, tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", @@ -500,8 +528,9 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) } for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { - ret = snd_ctl_add(codec->card, - snd_ctl_new1(&tas2781_snd_controls[i], tas_priv)); + tas_hda->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->snd_ctls[i]); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", @@ -523,8 +552,9 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) goto out; } - ret = snd_ctl_add(codec->card, - snd_ctl_new1(&tas2781_dsp_prog_ctrl, tas_priv)); + tas_hda->dsp_prog_ctl = snd_ctl_new1(&tas2781_dsp_prog_ctrl, + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", @@ -532,8 +562,9 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) goto out; } - ret = snd_ctl_add(codec->card, - snd_ctl_new1(&tas2781_dsp_conf_ctrl, tas_priv)); + tas_hda->dsp_conf_ctl = snd_ctl_new1(&tas2781_dsp_conf_ctrl, + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); if (ret) { dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", @@ -554,27 +585,27 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tas2781_save_calibration(tas_priv); out: - mutex_unlock(&tas_priv->codec_lock); + mutex_unlock(&tas_hda->priv->codec_lock); if (fmw) release_firmware(fmw); - pm_runtime_mark_last_busy(tas_priv->dev); - pm_runtime_put_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_hda->dev); + pm_runtime_put_autosuspend(tas_hda->dev); } static int tas2781_hda_bind(struct device *dev, struct device *master, void *master_data) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); struct hda_component *comps = master_data; struct hda_codec *codec; unsigned int subid; int ret; - if (!comps || tas_priv->index < 0 || - tas_priv->index >= HDA_MAX_COMPONENTS) + if (!comps || tas_hda->priv->index < 0 || + tas_hda->priv->index >= HDA_MAX_COMPONENTS) return -EINVAL; - comps = &comps[tas_priv->index]; + comps = &comps[tas_hda->priv->index]; if (comps->dev) return -EBUSY; @@ -583,10 +614,10 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, switch (subid) { case 0x17aa: - tas_priv->catlog_id = LENOVO; + tas_hda->priv->catlog_id = LENOVO; break; default: - tas_priv->catlog_id = OTHERS; + tas_hda->priv->catlog_id = OTHERS; break; } @@ -596,7 +627,7 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, strscpy(comps->name, dev_name(dev), sizeof(comps->name)); - ret = tascodec_init(tas_priv, codec, tasdev_fw_ready); + ret = tascodec_init(tas_hda->priv, codec, tasdev_fw_ready); if (!ret) comps->playback_hook = tas2781_hda_playback_hook; @@ -609,9 +640,9 @@ static int tas2781_hda_bind(struct device *dev, struct device *master, static void tas2781_hda_unbind(struct device *dev, struct device *master, void *master_data) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); struct hda_component *comps = master_data; - comps = &comps[tas_priv->index]; + comps = &comps[tas_hda->priv->index]; if (comps->dev == dev) { comps->dev = NULL; @@ -619,10 +650,12 @@ static void tas2781_hda_unbind(struct device *dev, comps->playback_hook = NULL; } - tasdevice_config_info_remove(tas_priv); - tasdevice_dsp_remove(tas_priv); + tas2781_hda_remove_controls(tas_hda); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + tasdevice_config_info_remove(tas_hda->priv); + tasdevice_dsp_remove(tas_hda->priv); + + tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; } static const struct component_ops tas2781_hda_comp_ops = { @@ -632,21 +665,21 @@ static const struct component_ops tas2781_hda_comp_ops = { static void tas2781_hda_remove(struct device *dev) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - pm_runtime_get_sync(tas_priv->dev); - pm_runtime_disable(tas_priv->dev); + pm_runtime_get_sync(tas_hda->dev); + pm_runtime_disable(tas_hda->dev); - component_del(tas_priv->dev, &tas2781_hda_comp_ops); + component_del(tas_hda->dev, &tas2781_hda_comp_ops); - pm_runtime_put_noidle(tas_priv->dev); + pm_runtime_put_noidle(tas_hda->dev); - tasdevice_remove(tas_priv); + tasdevice_remove(tas_hda->priv); } static int tas2781_hda_i2c_probe(struct i2c_client *clt) { - struct tasdevice_priv *tas_priv; + struct tas2781_hda *tas_hda; const char *device_name; int ret; @@ -655,37 +688,42 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) else return -ENODEV; - tas_priv = tasdevice_kzalloc(clt); - if (!tas_priv) + tas_hda = devm_kzalloc(&clt->dev, sizeof(*tas_hda), GFP_KERNEL); + if (!tas_hda) return -ENOMEM; - dev_set_drvdata(&clt->dev, tas_priv); + dev_set_drvdata(&clt->dev, tas_hda); + tas_hda->dev = &clt->dev; - tas_priv->irq_info.irq = clt->irq; - ret = tas2781_read_acpi(tas_priv, device_name); + tas_hda->priv = tasdevice_kzalloc(clt); + if (!tas_hda->priv) + return -ENOMEM; + + tas_hda->priv->irq_info.irq = clt->irq; + ret = tas2781_read_acpi(tas_hda->priv, device_name); if (ret) - return dev_err_probe(tas_priv->dev, ret, + return dev_err_probe(tas_hda->dev, ret, "Platform not supported\n"); - ret = tasdevice_init(tas_priv); + ret = tasdevice_init(tas_hda->priv); if (ret) goto err; - pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); - pm_runtime_use_autosuspend(tas_priv->dev); - pm_runtime_mark_last_busy(tas_priv->dev); - pm_runtime_set_active(tas_priv->dev); - pm_runtime_get_noresume(tas_priv->dev); - pm_runtime_enable(tas_priv->dev); + pm_runtime_set_autosuspend_delay(tas_hda->dev, 3000); + pm_runtime_use_autosuspend(tas_hda->dev); + pm_runtime_mark_last_busy(tas_hda->dev); + pm_runtime_set_active(tas_hda->dev); + pm_runtime_get_noresume(tas_hda->dev); + pm_runtime_enable(tas_hda->dev); - pm_runtime_put_autosuspend(tas_priv->dev); + pm_runtime_put_autosuspend(tas_hda->dev); - tas2781_reset(tas_priv); + tas2781_reset(tas_hda->priv); - ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + ret = component_add(tas_hda->dev, &tas2781_hda_comp_ops); if (ret) { - dev_err(tas_priv->dev, "Register component failed: %d\n", ret); - pm_runtime_disable(tas_priv->dev); + dev_err(tas_hda->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(tas_hda->dev); } err: @@ -701,66 +739,65 @@ static void tas2781_hda_i2c_remove(struct i2c_client *clt) static int tas2781_runtime_suspend(struct device *dev) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); int i; - dev_dbg(tas_priv->dev, "Runtime Suspend\n"); + dev_dbg(tas_hda->dev, "Runtime Suspend\n"); - mutex_lock(&tas_priv->codec_lock); + mutex_lock(&tas_hda->priv->codec_lock); - if (tas_priv->playback_started) { - tasdevice_tuning_switch(tas_priv, 1); - tas_priv->playback_started = false; + if (tas_hda->priv->playback_started) { + tasdevice_tuning_switch(tas_hda->priv, 1); + tas_hda->priv->playback_started = false; } - for (i = 0; i < tas_priv->ndev; i++) { - tas_priv->tasdevice[i].cur_book = -1; - tas_priv->tasdevice[i].cur_prog = -1; - tas_priv->tasdevice[i].cur_conf = -1; + for (i = 0; i < tas_hda->priv->ndev; i++) { + tas_hda->priv->tasdevice[i].cur_book = -1; + tas_hda->priv->tasdevice[i].cur_prog = -1; + tas_hda->priv->tasdevice[i].cur_conf = -1; } - - mutex_unlock(&tas_priv->codec_lock); + mutex_unlock(&tas_hda->priv->codec_lock); return 0; } static int tas2781_runtime_resume(struct device *dev) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); unsigned long calib_data_sz = - tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + tas_hda->priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; - dev_dbg(tas_priv->dev, "Runtime Resume\n"); + dev_dbg(tas_hda->dev, "Runtime Resume\n"); - mutex_lock(&tas_priv->codec_lock); + mutex_lock(&tas_hda->priv->codec_lock); - tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + tasdevice_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog); /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ - if (tas_priv->cali_data.total_sz > calib_data_sz) - tas2781_apply_calib(tas_priv); + if (tas_hda->priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_hda->priv); - mutex_unlock(&tas_priv->codec_lock); + mutex_unlock(&tas_hda->priv->codec_lock); return 0; } static int tas2781_system_suspend(struct device *dev) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); int ret; - dev_dbg(tas_priv->dev, "System Suspend\n"); + dev_dbg(tas_hda->priv->dev, "System Suspend\n"); ret = pm_runtime_force_suspend(dev); if (ret) return ret; /* Shutdown chip before system suspend */ - tasdevice_tuning_switch(tas_priv, 1); + tasdevice_tuning_switch(tas_hda->priv, 1); /* * Reset GPIO may be shared, so cannot reset here. @@ -771,33 +808,33 @@ static int tas2781_system_suspend(struct device *dev) static int tas2781_system_resume(struct device *dev) { - struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); unsigned long calib_data_sz = - tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + tas_hda->priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; int i, ret; - dev_dbg(tas_priv->dev, "System Resume\n"); + dev_info(tas_hda->priv->dev, "System Resume\n"); ret = pm_runtime_force_resume(dev); if (ret) return ret; - mutex_lock(&tas_priv->codec_lock); + mutex_lock(&tas_hda->priv->codec_lock); - for (i = 0; i < tas_priv->ndev; i++) { - tas_priv->tasdevice[i].cur_book = -1; - tas_priv->tasdevice[i].cur_prog = -1; - tas_priv->tasdevice[i].cur_conf = -1; + for (i = 0; i < tas_hda->priv->ndev; i++) { + tas_hda->priv->tasdevice[i].cur_book = -1; + tas_hda->priv->tasdevice[i].cur_prog = -1; + tas_hda->priv->tasdevice[i].cur_conf = -1; } - tas2781_reset(tas_priv); - tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + tas2781_reset(tas_hda->priv); + tasdevice_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog); /* If calibrated data occurs error, dsp will still work with default * calibrated data inside algo. */ - if (tas_priv->cali_data.total_sz > calib_data_sz) - tas2781_apply_calib(tas_priv); - mutex_unlock(&tas_priv->codec_lock); + if (tas_hda->priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_hda->priv); + mutex_unlock(&tas_hda->priv->codec_lock); return 0; } From ee694e7db47e1af00ffb29f569904a9ed576868f Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 21 Dec 2023 13:25:16 +0000 Subject: [PATCH 80/82] ALSA: hda: cs35l41: Support additional Dell models without _DSD Add new model entries into configuration table. Signed-off-by: Stefan Binding Cc: # v6.7+ Link: https://lore.kernel.org/r/20231221132518.3213-2-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 194e1179a253..5eab2de0d4bb 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -35,6 +35,10 @@ struct cs35l41_config { }; static const struct cs35l41_config cs35l41_config_table[] = { + { "10280B27", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280B28", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280BEB", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, + { "10280C4D", I2C, 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT }, 0, 1, -1, 1000, 4500, 24 }, /* * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. * We can override the _DSD to correct the boost type here. @@ -347,6 +351,10 @@ struct cs35l41_prop_model { static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, + { "CSC3551", "10280B27", generic_dsd_config }, + { "CSC3551", "10280B28", generic_dsd_config }, + { "CSC3551", "10280BEB", generic_dsd_config }, + { "CSC3551", "10280C4D", generic_dsd_config }, { "CSC3551", "103C89C6", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, From d110858a6925827609d11db8513d76750483ec06 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 21 Dec 2023 13:25:17 +0000 Subject: [PATCH 81/82] ALSA: hda: cs35l41: Prevent firmware load if SPI speed too low Some laptops without _DSD have the SPI speed set very low in the BIOS. Since the SPI controller uses this speed as its max speed, the SPI transactions are very slow. Firmware download writes to many registers, and if the SPI speed is too slow, it can take a long time to download. For this reason, disable firmware loading if the maximum SPI speed is too low. Without Firmware, audio playback will work, but the volume will be low to ensure safe operation of the CS35L41. Signed-off-by: Stefan Binding Cc: # v6.7+ Link: https://lore.kernel.org/r/20231221132518.3213-3-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 25 ++++++++-- sound/pci/hda/cs35l41_hda.h | 12 ++++- sound/pci/hda/cs35l41_hda_i2c.c | 2 +- sound/pci/hda/cs35l41_hda_property.c | 74 +++++++++++++--------------- sound/pci/hda/cs35l41_hda_spi.c | 2 +- 5 files changed, 70 insertions(+), 45 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 92ca2b3b6c92..d3fa6e136744 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -12,6 +12,7 @@ #include #include #include +#include #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -996,6 +997,11 @@ static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) __be32 halo_sts; int ret; + if (cs35l41->bypass_fw) { + dev_warn(cs35l41->dev, "Bypassing Firmware.\n"); + return 0; + } + ret = cs35l41_init_dsp(cs35l41); if (ret) { dev_warn(cs35l41->dev, "Cannot Initialize Firmware. Error: %d\n", ret); @@ -1588,6 +1594,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i u32 values[HDA_MAX_COMPONENTS]; struct acpi_device *adev; struct device *physdev; + struct spi_device *spi; const char *sub; char *property; size_t nval; @@ -1610,7 +1617,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); if (!ret) { dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); - goto put_physdev; + goto out; } property = "cirrus,dev-index"; @@ -1701,8 +1708,20 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i hw_cfg->bst_type = CS35L41_EXT_BOOST; hw_cfg->valid = true; +out: put_device(physdev); + cs35l41->bypass_fw = false; + if (cs35l41->control_bus == SPI) { + spi = to_spi_device(cs35l41->dev); + if (spi->max_speed_hz < CS35L41_MAX_ACCEPTABLE_SPI_SPEED_HZ) { + dev_warn(cs35l41->dev, + "SPI speed is too slow to support firmware download: %d Hz.\n", + spi->max_speed_hz); + cs35l41->bypass_fw = true; + } + } + return 0; err: @@ -1711,14 +1730,13 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; acpi_dev_put(cs35l41->dacpi); -put_physdev: put_device(physdev); return ret; } int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, - struct regmap *regmap) + struct regmap *regmap, enum control_bus control_bus) { unsigned int regid, reg_revid; struct cs35l41_hda *cs35l41; @@ -1737,6 +1755,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i cs35l41->dev = dev; cs35l41->irq = irq; cs35l41->regmap = regmap; + cs35l41->control_bus = control_bus; dev_set_drvdata(dev, cs35l41); ret = cs35l41_hda_read_acpi(cs35l41, device_name, id); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index 3d925d677213..43d55292b327 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -20,6 +20,8 @@ #include #include +#define CS35L41_MAX_ACCEPTABLE_SPI_SPEED_HZ 1000000 + struct cs35l41_amp_cal_data { u32 calTarget[2]; u32 calTime[2]; @@ -46,6 +48,11 @@ enum cs35l41_hda_gpio_function { CS35l41_SYNC, }; +enum control_bus { + I2C, + SPI +}; + struct cs35l41_hda { struct device *dev; struct regmap *regmap; @@ -74,6 +81,9 @@ struct cs35l41_hda { struct cs_dsp cs_dsp; struct acpi_device *dacpi; bool mute_override; + enum control_bus control_bus; + bool bypass_fw; + }; enum halo_state { @@ -85,7 +95,7 @@ enum halo_state { extern const struct dev_pm_ops cs35l41_hda_pm_ops; int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, - struct regmap *regmap); + struct regmap *regmap, enum control_bus control_bus); void cs35l41_hda_remove(struct device *dev); int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id); diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index b44536fbba17..603e9bff3a71 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -30,7 +30,7 @@ static int cs35l41_hda_i2c_probe(struct i2c_client *clt) return -ENODEV; return cs35l41_hda_probe(&clt->dev, device_name, clt->addr, clt->irq, - devm_regmap_init_i2c(clt, &cs35l41_regmap_i2c)); + devm_regmap_init_i2c(clt, &cs35l41_regmap_i2c), I2C); } static void cs35l41_hda_i2c_remove(struct i2c_client *clt) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 5eab2de0d4bb..be2b01b596c2 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -16,10 +16,6 @@ struct cs35l41_config { const char *ssid; - enum { - SPI, - I2C - } bus; int num_amps; enum { INTERNAL, @@ -35,46 +31,46 @@ struct cs35l41_config { }; static const struct cs35l41_config cs35l41_config_table[] = { - { "10280B27", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10280B28", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10280BEB", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, - { "10280C4D", I2C, 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT }, 0, 1, -1, 1000, 4500, 24 }, + { "10280B27", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280B28", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280BEB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, + { "10280C4D", 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT }, 0, 1, -1, 1000, 4500, 24 }, /* * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. * We can override the _DSD to correct the boost type here. * Since this laptop has valid ACPI, we do not need to handle cs-gpios, since that already exists * in the ACPI. The Reset GPIO is also valid, so we can use the Reset defined in _DSD. */ - { "103C89C6", SPI, 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, - { "104312AF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431433", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431463", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431473", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, - { "10431483", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, - { "10431493", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "104314D3", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "104314E3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431503", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431533", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431573", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431663", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, - { "104316D3", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "104316F3", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "104317F3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431863", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "104318D3", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, - { "10431C9F", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CAF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CCF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CDF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CEF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431D1F", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431DA2", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "10431E02", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "10431EE2", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, - { "10431F12", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431F1F", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, - { "10431F62", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "103C89C6", 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, + { "104312AF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431433", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431463", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431473", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431483", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431493", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104314D3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104314E3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431503", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431533", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431573", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431663", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "104316D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "104316F3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "104317F3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431863", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104318D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "10431C9F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CAF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CCF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CDF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CEF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431D1F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431DA2", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10431E02", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10431EE2", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, + { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, {} }; @@ -212,7 +208,7 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde "_DSD already exists.\n"); } - if (cfg->bus == SPI) { + if (cs35l41->control_bus == SPI) { cs35l41->index = id; #if IS_ENABLED(CONFIG_SPI) diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index eb287aa5f782..b76c0dfd5fef 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -26,7 +26,7 @@ static int cs35l41_hda_spi_probe(struct spi_device *spi) return -ENODEV; return cs35l41_hda_probe(&spi->dev, device_name, spi_get_chipselect(spi, 0), spi->irq, - devm_regmap_init_spi(spi, &cs35l41_regmap_spi)); + devm_regmap_init_spi(spi, &cs35l41_regmap_spi), SPI); } static void cs35l41_hda_spi_remove(struct spi_device *spi) From 423206604b28174698d77bf5ea81365cdd6c0f77 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 21 Dec 2023 13:25:18 +0000 Subject: [PATCH 82/82] ALSA: hda/realtek: Add quirks for Dell models These models use 2 or 4 CS35L41 amps with HDA using SPI and I2C. Models use internal and external boost. All models require DSD support to be added inside cs35l41_hda_property.c Signed-off-by: Stefan Binding Cc: # v6.7+ Link: https://lore.kernel.org/r/20231221132518.3213-4-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c3a756528886..19040887ff67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6956,6 +6956,11 @@ static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); } +static void cs35l41_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 4); +} + static void cs35l41_fixup_spi_two(struct hda_codec *codec, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(codec, action, "spi", "CSC3551", 2); @@ -7441,6 +7446,7 @@ enum { ALC287_FIXUP_LEGION_16ACHG6, ALC287_FIXUP_CS35L41_I2C_2, ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED, + ALC287_FIXUP_CS35L41_I2C_4, ALC245_FIXUP_CS35L41_SPI_2, ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_4, @@ -9427,6 +9433,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_MUTE_LED, }, + [ALC287_FIXUP_CS35L41_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_four, + }, [ALC245_FIXUP_CS35L41_SPI_2] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_two, @@ -9703,6 +9713,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b27, "Dell", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1028, 0x0b28, "Dell", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2), @@ -9713,6 +9725,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0c1c, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1d, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), + SND_PCI_QUIRK(0x1028, 0x0c4d, "Dell", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC289_FIXUP_DELL_CS35L41_SPI_2),