diff options
author | Mark Brown <broonie@kernel.org> | 2023-09-26 16:14:44 +0200 |
---|---|---|
committer | Mark Brown <broonie@kernel.org> | 2023-09-26 16:14:44 +0200 |
commit | af08458988cb5dd4b4ff87cfb9da81c6d2c8ef7a (patch) | |
tree | 5a71f1c4e3099d7cfe4c6f880e707173b6ec9cda | |
parent | e952e89b0602aeb856396eac4306098249c43548 (diff) | |
parent | 2b21207afd06714986a3d22442ed4860ba4f9ced (diff) | |
download | linux-af08458988cb5dd4b4ff87cfb9da81c6d2c8ef7a.tar.gz linux-af08458988cb5dd4b4ff87cfb9da81c6d2c8ef7a.tar.bz2 linux-af08458988cb5dd4b4ff87cfb9da81c6d2c8ef7a.zip |
ASoC: Merge up fixes
For the benefit of CI.
27 files changed, 183 insertions, 68 deletions
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index 4f51b2fa82db..c3c989ef2a2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -26,6 +26,7 @@ properties: - const: rockchip,rk3568-spdif - items: - enum: + - rockchip,rk3128-spdif - rockchip,rk3188-spdif - rockchip,rk3288-spdif - rockchip,rk3308-spdif diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 49b70c70dc69..79d4254d1f9b 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1863,15 +1863,15 @@ static int cs_dsp_adsp2_setup_algs(struct cs_dsp *dsp) return PTR_ERR(adsp2_alg); for (i = 0; i < n_algs; i++) { - cs_dsp_info(dsp, - "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n", - i, be32_to_cpu(adsp2_alg[i].alg.id), - (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, - (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, - be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff, - be32_to_cpu(adsp2_alg[i].xm), - be32_to_cpu(adsp2_alg[i].ym), - be32_to_cpu(adsp2_alg[i].zm)); + cs_dsp_dbg(dsp, + "%d: ID %x v%d.%d.%d XM@%x YM@%x ZM@%x\n", + i, be32_to_cpu(adsp2_alg[i].alg.id), + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff0000) >> 16, + (be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff00) >> 8, + be32_to_cpu(adsp2_alg[i].alg.ver) & 0xff, + be32_to_cpu(adsp2_alg[i].xm), + be32_to_cpu(adsp2_alg[i].ym), + be32_to_cpu(adsp2_alg[i].zm)); alg_region = cs_dsp_create_region(dsp, WMFW_ADSP2_XM, adsp2_alg[i].alg.id, @@ -1996,14 +1996,14 @@ static int cs_dsp_halo_setup_algs(struct cs_dsp *dsp) return PTR_ERR(halo_alg); for (i = 0; i < n_algs; i++) { - cs_dsp_info(dsp, - "%d: ID %x v%d.%d.%d XM@%x YM@%x\n", - i, be32_to_cpu(halo_alg[i].alg.id), - (be32_to_cpu(halo_alg[i].alg.ver) & 0xff0000) >> 16, - (be32_to_cpu(halo_alg[i].alg.ver) & 0xff00) >> 8, - be32_to_cpu(halo_alg[i].alg.ver) & 0xff, - be32_to_cpu(halo_alg[i].xm_base), - be32_to_cpu(halo_alg[i].ym_base)); + cs_dsp_dbg(dsp, + "%d: ID %x v%d.%d.%d XM@%x YM@%x\n", + i, be32_to_cpu(halo_alg[i].alg.id), + (be32_to_cpu(halo_alg[i].alg.ver) & 0xff0000) >> 16, + (be32_to_cpu(halo_alg[i].alg.ver) & 0xff00) >> 8, + be32_to_cpu(halo_alg[i].alg.ver) & 0xff, + be32_to_cpu(halo_alg[i].xm_base), + be32_to_cpu(halo_alg[i].ym_base)); ret = cs_dsp_halo_create_regions(dsp, halo_alg[i].alg.id, halo_alg[i].alg.ver, diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index 8ee1baa03269..87dd0ccade4c 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -452,11 +452,13 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev, if ((aw_bin->all_bin_parse_num != 1) || (aw_bin->header_info[0].bin_data_type != DATA_TYPE_REGISTER)) { dev_err(aw_dev->dev, "bin num or type error"); + ret = -EINVAL; goto parse_bin_failed; } if (aw_bin->header_info[0].valid_data_len % 4) { dev_err(aw_dev->dev, "bin data len get error!"); + ret = -EINVAL; goto parse_bin_failed; } diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 7063c400e896..9e5670b09af6 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -27,7 +27,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return -ENOMEM; cs35l56->base.dev = dev; - cs35l56->base.can_hibernate = true; i2c_set_clientdata(client, cs35l56); cs35l56->base.regmap = devm_regmap_init_i2c(client, regmap_config); diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index eeab07c850f9..974bae4abfad 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -344,6 +344,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral, switch (status) { case SDW_SLAVE_ATTACHED: dev_dbg(cs42l42->dev, "ATTACHED\n"); + + /* + * The SoundWire core can report stale ATTACH notifications + * if we hard-reset CS42L42 in probe() but it had already been + * enumerated. Reject the ATTACH if we haven't yet seen an + * UNATTACH report for the device being in reset. + */ + if (cs42l42->sdw_waiting_first_unattach) + break; + /* * Initialise codec, this only needs to be done once. * When resuming from suspend, resume callback will handle re-init of codec, @@ -354,6 +364,16 @@ static int cs42l42_sdw_update_status(struct sdw_slave *peripheral, break; case SDW_SLAVE_UNATTACHED: dev_dbg(cs42l42->dev, "UNATTACHED\n"); + + if (cs42l42->sdw_waiting_first_unattach) { + /* + * SoundWire core has seen that CS42L42 is not on + * the bus so release RESET and wait for ATTACH. + */ + cs42l42->sdw_waiting_first_unattach = false; + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + } + break; default: break; diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a0de0329406a..2961340f15e2 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -2320,7 +2320,26 @@ int cs42l42_common_probe(struct cs42l42_private *cs42l42, if (cs42l42->reset_gpio) { dev_dbg(cs42l42->dev, "Found reset GPIO\n"); - gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + + /* + * ACPI can override the default GPIO state we requested + * so ensure that we start with RESET low. + */ + gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); + + /* Ensure minimum reset pulse width */ + usleep_range(10, 500); + + /* + * On SoundWire keep the chip in reset until we get an UNATTACH + * notification from the SoundWire core. This acts as a + * synchronization point to reject stale ATTACH notifications + * if the chip was already enumerated before we reset it. + */ + if (cs42l42->sdw_peripheral) + cs42l42->sdw_waiting_first_unattach = true; + else + gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 4bd7b85a5747..7785125b73ab 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -53,6 +53,7 @@ struct cs42l42_private { u8 stream_use; bool hp_adc_up_pending; bool suspended; + bool sdw_waiting_first_unattach; bool init_done; }; diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 4e3bc15f1b25..532183095818 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2077,7 +2077,8 @@ static const struct cs42l43_irq cs42l43_irqs[] = { static int cs42l43_request_irq(struct cs42l43_codec *priv, struct irq_domain *dom, const char * const name, - unsigned int irq, irq_handler_t handler) + unsigned int irq, irq_handler_t handler, + unsigned long flags) { int ret; @@ -2087,8 +2088,8 @@ static int cs42l43_request_irq(struct cs42l43_codec *priv, dev_dbg(priv->dev, "Request IRQ %d for %s\n", ret, name); - ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, IRQF_ONESHOT, - name, priv); + ret = devm_request_threaded_irq(priv->dev, ret, NULL, handler, + IRQF_ONESHOT | flags, name, priv); if (ret) return dev_err_probe(priv->dev, ret, "Failed to request IRQ %s\n", name); @@ -2124,11 +2125,11 @@ static int cs42l43_shutter_irq(struct cs42l43_codec *priv, return 0; } - ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler); + ret = cs42l43_request_irq(priv, dom, close_name, close_irq, handler, IRQF_SHARED); if (ret) return ret; - return cs42l43_request_irq(priv, dom, open_name, open_irq, handler); + return cs42l43_request_irq(priv, dom, open_name, open_irq, handler, IRQF_SHARED); } static int cs42l43_codec_probe(struct platform_device *pdev) @@ -2178,7 +2179,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev) for (i = 0; i < ARRAY_SIZE(cs42l43_irqs); i++) { ret = cs42l43_request_irq(priv, dom, cs42l43_irqs[i].name, - cs42l43_irqs[i].irq, cs42l43_irqs[i].handler); + cs42l43_irqs[i].irq, + cs42l43_irqs[i].handler, 0); if (ret) goto err_pm; } diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 15e1a62b9e57..e8cdc166bdaa 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2403,13 +2403,11 @@ static irqreturn_t rt5640_irq(int irq, void *data) struct rt5640_priv *rt5640 = data; int delay = 0; - if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { - cancel_delayed_work_sync(&rt5640->jack_work); + if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) delay = 100; - } if (rt5640->jack) - queue_delayed_work(system_long_wq, &rt5640->jack_work, delay); + mod_delayed_work(system_long_wq, &rt5640->jack_work, delay); return IRQ_HANDLED; } @@ -2565,10 +2563,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, if (jack_data && jack_data->use_platform_clock) rt5640->use_platform_clock = jack_data->use_platform_clock; - ret = devm_request_threaded_irq(component->dev, rt5640->irq, - NULL, rt5640_irq, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, - "rt5640", rt5640); + ret = request_irq(rt5640->irq, rt5640_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "rt5640", rt5640); if (ret) { dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640_disable_jack_detect(component); @@ -2621,14 +2618,14 @@ static void rt5640_enable_hda_jack_detect( rt5640->jack = jack; - ret = devm_request_threaded_irq(component->dev, rt5640->irq, - NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT, - "rt5640", rt5640); + ret = request_irq(rt5640->irq, rt5640_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); - rt5640->irq = -ENXIO; + rt5640->jack = NULL; return; } + rt5640->irq_requested = true; /* sync initial jack state */ queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); @@ -2801,12 +2798,12 @@ static int rt5640_suspend(struct snd_soc_component *component) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); - if (rt5640->irq) { + if (rt5640->jack) { /* disable jack interrupts during system suspend */ disable_irq(rt5640->irq); + rt5640_cancel_work(rt5640); } - rt5640_cancel_work(rt5640); snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); rt5640_reset(component); regcache_cache_only(rt5640->regmap, true); @@ -2829,9 +2826,6 @@ static int rt5640_resume(struct snd_soc_component *component) regcache_cache_only(rt5640->regmap, false); regcache_sync(rt5640->regmap); - if (rt5640->irq) - enable_irq(rt5640->irq); - if (rt5640->jack) { if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { snd_soc_component_update_bits(component, @@ -2859,6 +2853,7 @@ static int rt5640_resume(struct snd_soc_component *component) } } + enable_irq(rt5640->irq); queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index db847e80a9c6..236b12b69ae5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -687,7 +687,10 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, struct wm_coeff_ctl *ctl; int ret; + mutex_lock(&dsp->cs_dsp.pwr_lock); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); + mutex_unlock(&dsp->cs_dsp.pwr_lock); + if (ret < 0) return ret; @@ -703,8 +706,14 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), - 0, buf, len); + int ret; + + mutex_lock(&dsp->cs_dsp.pwr_lock); + ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), + 0, buf, len); + mutex_unlock(&dsp->cs_dsp.pwr_lock); + + return ret; } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5b31c12a56f9..0957ff7c55c2 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -52,8 +52,8 @@ struct codec_priv { unsigned long mclk_freq; unsigned long free_freq; u32 mclk_id; - u32 fll_id; - u32 pll_id; + int fll_id; + int pll_id; }; /** @@ -206,7 +206,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } /* Specific configuration for PLL */ - if (codec_priv->pll_id && codec_priv->fll_id) { + if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) pll_out = priv->sample_rate * 384; else @@ -248,7 +248,7 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) priv->streams &= ~BIT(substream->stream); - if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { + if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) { /* Force freq to be free_freq to avoid error message in codec */ ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0), codec_priv->mclk_id, @@ -621,6 +621,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dapm_routes = audio_map; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.driver_name = DRIVER_NAME; + + priv->codec_priv.fll_id = -1; + priv->codec_priv.pll_id = -1; + /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 1e4020fae05a..8a9a30dd31e2 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -710,10 +710,15 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir) { unsigned int ofs = sai->soc_data->reg_offset; bool tx = dir == TX; - u32 xcsr, count = 100; + u32 xcsr, count = 100, mask; + + if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output) + mask = FSL_SAI_CSR_TERE; + else + mask = FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE; regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), - FSL_SAI_CSR_TERE | FSL_SAI_CSR_BCE, 0); + mask, 0); /* TERE will remain set till the end of current frame */ do { diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index b2c12e4ed5bf..2aeb18397bcb 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -315,7 +315,7 @@ static int imx_audmix_probe(struct platform_device *pdev) if (IS_ERR(priv->cpu_mclk)) { ret = PTR_ERR(priv->cpu_mclk); dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); - return -EINVAL; + return ret; } priv->audmix_pdev = audmix_pdev; diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index e0c416a2eff8..a9324712e3fa 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -89,6 +89,14 @@ static int imx_rpmsg_probe(struct platform_device *pdev) SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + /* + * i.MX rpmsg sound cards work on codec slave mode. MCLK will be + * disabled by CPU DAI driver in hw_free(). Some codec requires MCLK + * present at power up/down sequence. So need to set ignore_pmdown_time + * to power down codec immediately before MCLK is turned off. + */ + data->dai.ignore_pmdown_time = 1; + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 36ce3a4343f9..11f186ea662a 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -310,7 +310,8 @@ int simple_util_startup(struct snd_pcm_substream *substream) if (fixed_sysclk % props->mclk_fs) { dev_err(rtd->dev, "fixed sysclk %u not divisible by mclk_fs %u\n", fixed_sysclk, props->mclk_fs); - return -EINVAL; + ret = -EINVAL; + goto codec_err; } ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, fixed_rate, fixed_rate); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index a88812ab3ed1..b95b86315502 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -758,10 +758,12 @@ static int simple_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link = priv->dai_link; struct simple_dai_props *dai_props = priv->dai_props; + ret = -EINVAL; + cinfo = dev->platform_data; if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); - return -EINVAL; + goto err; } if (!cinfo->name || @@ -770,7 +772,7 @@ static int simple_probe(struct platform_device *pdev) !cinfo->platform || !cinfo->cpu_dai.name) { dev_err(dev, "insufficient simple_util_info settings\n"); - return -EINVAL; + goto err; } cpus = dai_link->cpus; diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index cb00bc86ac94..8876558f19a1 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -55,6 +55,9 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int return -ENOMEM; dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL); + if (!dl[i].codecs->name) + return -ENOMEM; + dl[i].codecs->dai_name = pcm->name; dl[i].num_codecs = 1; dl[i].num_cpus = 1; diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index f8a3e8a91761..9904a9e33ccc 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -808,6 +808,16 @@ static const struct platform_device_id board_ids[] = { SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | SOF_ES8336_JD_INVERTED), }, + { + .name = "mtl_es83x6_c1_h02", + .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | + SOF_NO_OF_HDMI_CAPTURE_SSP(2) | + SOF_HDMI_CAPTURE_1_SSP(0) | + SOF_HDMI_CAPTURE_2_SSP(2) | + SOF_SSP_HDMI_CAPTURE_PRESENT | + SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | + SOF_ES8336_JD_INVERTED), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b36cdf374a82..9cb666588fe6 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -373,6 +373,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .callback = sof_sdw_quirk_cb, .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B14"), + }, + /* No Jack */ + .driver_data = (void *)SOF_SDW_TGL_HDMI, + }, + + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B29"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index b513eceb60c3..6e712ad954c8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -675,18 +675,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .sof_tplg_filename = "sof-adl-rt1316-l2-mono-rt714-l3.tplg", }, { - .link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */ - .links = adl_sdw_rt1316_link1_rt714_link0, - .drv_name = "sof_sdw", - .sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg", - }, - { .link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */ .links = adl_sdw_rt1316_link12_rt714_link0, .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt1316-l12-rt714-l0.tplg", }, { + .link_mask = 0x3, /* rt1316 on link1 & rt714 on link0 */ + .links = adl_sdw_rt1316_link1_rt714_link0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-adl-rt1316-l1-mono-rt714-l0.tplg", + }, + { .link_mask = 0x5, /* 2 active links required */ .links = adl_sdw_rt1316_link2_rt714_link0, .drv_name = "sof_sdw", 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 b6409d2bd1fb..fbacabc93d6d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -30,6 +30,16 @@ static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; +static const struct snd_soc_acpi_codecs mtl_essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + +static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = { + .num_codecs = 1, + .codecs = {"INTC10B0"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -52,6 +62,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_rt1019p_amp, .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "mtl_es83x6_c1_h02", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &mtl_lt6911_hdmi, + .sof_tplg_filename = "sof-mtl-es83x6-ssp1-hdmi-ssp02.tplg", + }, + { + .comp_ids = &mtl_essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-mtl-es8336", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 63ae0c2310d7..092ca09f3631 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -44,8 +44,8 @@ static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, * platforms which make use of the snd_dmaengine_dai_dma_data struct for their * DAI DMA data. Internally the function will first call * snd_hwparams_to_dma_slave_config to fill in the slave config based on the - * hw_params, followed by snd_dmaengine_set_config_from_dai_data to fill in the - * remaining fields based on the DAI DMA data. + * hw_params, followed by snd_dmaengine_pcm_set_config_from_dai_data to fill in + * the remaining fields based on the DAI DMA data. */ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 30db685cc5f4..2d1616b81485 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -486,10 +486,9 @@ int snd_sof_device_remove(struct device *dev) snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_remove(sdev); + sof_ops_free(sdev); } - sof_ops_free(sdev); - /* release firmware */ snd_sof_fw_unload(sdev); diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 0e91169c685e..254dbbeac1d0 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -463,7 +463,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) /* step 3: wait for IPC DONE bit from ROM */ ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->ipc_ack, status, ((status & chip->ipc_ack_mask) == chip->ipc_ack_mask), - HDA_DSP_REG_POLL_INTERVAL_US, MTL_DSP_PURGE_TIMEOUT_US); + HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_INIT_TIMEOUT_US); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, "timeout waiting for purge IPC done\n"); diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index 02181490f12a..95696b3d7c4c 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -62,7 +62,6 @@ #define MTL_DSP_IRQSTS_IPC BIT(0) #define MTL_DSP_IRQSTS_SDW BIT(6) -#define MTL_DSP_PURGE_TIMEOUT_US 20000000 /* 20s */ #define MTL_DSP_REG_POLL_INTERVAL_US 10 /* 10 us */ /* Memory windows */ diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index afd23d3f16cd..bf91c8786162 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -231,7 +231,7 @@ static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp, ret = sof_update_ipc_object(scomp, available_fmt, SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(available_fmt), 1); + swidget->num_tuples, sizeof(*available_fmt), 1); if (ret) { dev_err(scomp->dev, "Failed to parse audio format token count\n"); return ret; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 9c2359d10ecf..563fe6f7789f 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -212,7 +212,8 @@ widget_free: sof_widget_free_unlocked(sdev, swidget); use_count_decremented = true; core_put: - snd_sof_dsp_core_put(sdev, swidget->core); + if (!use_count_decremented) + snd_sof_dsp_core_put(sdev, swidget->core); pipe_widget_free: if (swidget->id != snd_soc_dapm_scheduler) sof_widget_free_unlocked(sdev, swidget->spipe->pipe_widget); |