From 0c6498a59fbbcbf3d0a58c282dd6f0bca0eed92a Mon Sep 17 00:00:00 2001 From: Matus Malych Date: Sun, 12 Nov 2023 17:54:04 +0100 Subject: [PATCH 01/30] 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 02/30] 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 03/30] 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 04/30] 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 05/30] 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 06/30] 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 07/30] 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 08/30] 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 09/30] 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 10/30] 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 cdba4301adda7c60a2064bf808e48fccd352aaa9 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Wed, 22 Nov 2023 18:01:23 +0800 Subject: [PATCH 11/30] 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 12/30] 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 13/30] 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 14/30] 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 15/30] 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 16/30] 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 17/30] 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 18/30] 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 19/30] 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 19650c0f402f53abe48a55a1c49c8ed9576a088c Mon Sep 17 00:00:00 2001 From: Jeremy Soller Date: Mon, 27 Nov 2023 11:42:38 -0700 Subject: [PATCH 20/30] 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 21/30] 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 22/30] 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 23/30] 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 24/30] 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 25/30] 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 b5338b1b901e41bd7cead66a0b3a796e9fa95684 Mon Sep 17 00:00:00 2001 From: Marian Postevca Date: Sun, 3 Dec 2023 00:29:51 +0200 Subject: [PATCH 26/30] 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 27/30] 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 28/30] 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 fb9ad24485087e0f00d84bee7a5914640b2b9024 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Mon, 4 Dec 2023 12:47:35 +0000 Subject: [PATCH 29/30] 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 30/30] 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); }