diff options
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/amd/yc/acp6x-mach.c | 21 | ||||
-rw-r--r-- | sound/soc/codecs/aw88395/aw88395_lib.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/cs35l56-i2c.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/cs35l56.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/cs42l42-sdw.c | 20 | ||||
-rw-r--r-- | sound/soc/codecs/cs42l42.c | 21 | ||||
-rw-r--r-- | sound/soc/codecs/cs42l42.h | 1 | ||||
-rw-r--r-- | sound/soc/codecs/cs42l43.c | 14 | ||||
-rw-r--r-- | sound/soc/codecs/rt5640.c | 29 | ||||
-rw-r--r-- | sound/soc/codecs/wm8960.c | 19 | ||||
-rw-r--r-- | sound/soc/codecs/wm_adsp.c | 13 | ||||
-rw-r--r-- | sound/soc/fsl/imx-audmix.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/imx-pcm-rpmsg.c | 1 | ||||
-rw-r--r-- | sound/soc/fsl/imx-rpmsg.c | 8 | ||||
-rw-r--r-- | sound/soc/intel/avs/boards/hdaudio.c | 3 | ||||
-rw-r--r-- | sound/soc/meson/axg-spdifin.c | 49 | ||||
-rw-r--r-- | sound/soc/sh/rcar/core.c | 1 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 23 | ||||
-rw-r--r-- | sound/soc/soc-utils.c | 1 | ||||
-rw-r--r-- | sound/soc/sof/core.c | 3 | ||||
-rw-r--r-- | sound/soc/sof/intel/mtl.c | 2 | ||||
-rw-r--r-- | sound/soc/sof/intel/mtl.h | 1 | ||||
-rw-r--r-- | sound/soc/sof/ipc4-topology.c | 2 | ||||
-rw-r--r-- | sound/soc/sof/sof-audio.c | 3 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_audio_graph_card.c | 30 |
25 files changed, 175 insertions, 96 deletions
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 3ec15b46fa35..94e9eb8e73f2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -217,6 +217,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82QF"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "82TL"), } }, @@ -224,6 +231,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "82UG"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "82V2"), } }, @@ -265,6 +279,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { { .driver_data = &acp6x_card, .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Micro-Star International Co., Ltd."), + DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 B7ED"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"), DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"), } 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 9f4f2f4f23f5..d10e0e2380e8 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/cs35l56.c b/sound/soc/codecs/cs35l56.c index 600b79c62ec4..f2e7c6d0be46 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1207,6 +1207,7 @@ void cs35l56_remove(struct cs35l56_private *cs35l56) flush_workqueue(cs35l56->dsp_wq); destroy_workqueue(cs35l56->dsp_wq); + pm_runtime_dont_use_autosuspend(cs35l56->base.dev); pm_runtime_suspend(cs35l56->base.dev); pm_runtime_disable(cs35l56->base.dev); 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 1a95c370fc4c..5643c666d7d0 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/wm8960.c b/sound/soc/codecs/wm8960.c index 0a50180750e8..7689fe3cc86d 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1468,8 +1468,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) } wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); - if (IS_ERR(wm8960->regmap)) - return PTR_ERR(wm8960->regmap); + if (IS_ERR(wm8960->regmap)) { + ret = PTR_ERR(wm8960->regmap); + goto bulk_disable; + } if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); @@ -1479,13 +1481,14 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) ret = i2c_master_recv(i2c, &val, sizeof(val)); if (ret >= 0) { dev_err(&i2c->dev, "Not wm8960, wm8960 reg can not read by i2c\n"); - return -EINVAL; + ret = -EINVAL; + goto bulk_disable; } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); - return ret; + goto bulk_disable; } if (wm8960->pdata.shared_lrclk) { @@ -1494,7 +1497,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) if (ret != 0) { dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", ret); - return ret; + goto bulk_disable; } } @@ -1528,7 +1531,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c) ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8960, &wm8960_dai, 1); + if (ret) + goto bulk_disable; + return 0; + +bulk_disable: + regulator_bulk_disable(ARRAY_SIZE(wm8960->supplies), wm8960->supplies); return ret; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6fc34f41b175..d1b9238d391e 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/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 0b58df56f4da..aeb81aa61184 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-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index d63782b8bdef..bb736d45c9e0 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -19,6 +19,7 @@ static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 3c7b95db2eac..b578f9a32d7f 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/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/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d86880169075..bc2f2849ecfb 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -112,34 +112,6 @@ static int axg_spdifin_prepare(struct snd_pcm_substream *substream, return 0; } -static int axg_spdifin_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - int ret; - - ret = clk_prepare_enable(priv->refclk); - if (ret) { - dev_err(dai->dev, - "failed to enable spdifin reference clock\n"); - return ret; - } - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, - SPDIFIN_CTRL0_EN); - - return 0; -} - -static void axg_spdifin_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); - - regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); - clk_disable_unprepare(priv->refclk); -} - static void axg_spdifin_write_mode_param(struct regmap *map, int mode, unsigned int val, unsigned int num_per_reg, @@ -251,17 +223,32 @@ static int axg_spdifin_dai_probe(struct snd_soc_dai *dai) ret = axg_spdifin_sample_mode_config(dai, priv); if (ret) { dev_err(dai->dev, "mode configuration failed\n"); - clk_disable_unprepare(priv->pclk); - return ret; + goto pclk_err; } + ret = clk_prepare_enable(priv->refclk); + if (ret) { + dev_err(dai->dev, + "failed to enable spdifin reference clock\n"); + goto pclk_err; + } + + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, + SPDIFIN_CTRL0_EN); + return 0; + +pclk_err: + clk_disable_unprepare(priv->pclk); + return ret; } static int axg_spdifin_dai_remove(struct snd_soc_dai *dai) { struct axg_spdifin *priv = snd_soc_dai_get_drvdata(dai); + regmap_update_bits(priv->map, SPDIFIN_CTRL0, SPDIFIN_CTRL0_EN, 0); + clk_disable_unprepare(priv->refclk); clk_disable_unprepare(priv->pclk); return 0; } @@ -270,8 +257,6 @@ static const struct snd_soc_dai_ops axg_spdifin_ops = { .probe = axg_spdifin_dai_probe, .remove = axg_spdifin_dai_remove, .prepare = axg_spdifin_prepare, - .startup = axg_spdifin_startup, - .shutdown = axg_spdifin_shutdown, }; static int axg_spdifin_iec958_info(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e29c2fee9521..1bd7114c472a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1303,6 +1303,7 @@ audio_graph: if (i >= RSND_MAX_COMPONENT) { dev_info(dev, "reach to max component\n"); of_node_put(node); + of_node_put(ports); break; } } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb0723876851..54704250c0a2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -985,6 +985,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; + struct snd_pcm_hw_params tmp_params; int i, ret = 0; snd_soc_dpcm_mutex_assert_held(rtd); @@ -998,7 +999,6 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { - struct snd_pcm_hw_params codec_params; unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream); /* @@ -1019,23 +1019,22 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, continue; /* copy params for each codec */ - codec_params = *params; + tmp_params = *params; /* fixup params based on TDM slot masks */ if (tdm_mask) - soc_pcm_codec_params_fixup(&codec_params, tdm_mask); + soc_pcm_codec_params_fixup(&tmp_params, tdm_mask); ret = snd_soc_dai_hw_params(codec_dai, substream, - &codec_params); + &tmp_params); if(ret < 0) goto out; - soc_pcm_set_dai_params(codec_dai, &codec_params); - snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); + soc_pcm_set_dai_params(codec_dai, &tmp_params); + snd_soc_dapm_update_dai(substream, &tmp_params, codec_dai); } for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - struct snd_pcm_hw_params cpu_params; unsigned int ch_mask = 0; int j; @@ -1047,7 +1046,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, continue; /* copy params for each cpu */ - cpu_params = *params; + tmp_params = *params; if (!rtd->dai_link->codec_ch_maps) goto hw_params; @@ -1062,16 +1061,16 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, /* fixup cpu channel number */ if (ch_mask) - soc_pcm_codec_params_fixup(&cpu_params, ch_mask); + soc_pcm_codec_params_fixup(&tmp_params, ch_mask); hw_params: - ret = snd_soc_dai_hw_params(cpu_dai, substream, &cpu_params); + ret = snd_soc_dai_hw_params(cpu_dai, substream, &tmp_params); if (ret < 0) goto out; /* store the parameters for each DAI */ - soc_pcm_set_dai_params(cpu_dai, &cpu_params); - snd_soc_dapm_update_dai(substream, &cpu_params, cpu_dai); + soc_pcm_set_dai_params(cpu_dai, &tmp_params); + snd_soc_dapm_update_dai(substream, &tmp_params, cpu_dai); } ret = snd_soc_pcm_component_hw_params(substream, params); diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 11607c5f5d5a..9c746e4edef7 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -217,6 +217,7 @@ int snd_soc_dai_is_dummy(struct snd_soc_dai *dai) return 1; return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dai_is_dummy); int snd_soc_component_is_dummy(struct snd_soc_component *component) { 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 b84ca58da9d5..f9412517eaf2 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -460,7 +460,7 @@ 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 f2a30cd31378..7cb63e6b24dc 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 e7ef77012c35..e5405f854a91 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); diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index 1f2c5018bf5a..4737e776d383 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -10,6 +10,7 @@ #include <linux/platform_device.h> #include <sound/graph_card.h> #include <sound/pcm_params.h> +#include <sound/soc-dai.h> #define MAX_PLLA_OUT0_DIV 128 @@ -44,6 +45,21 @@ struct tegra_audio_cdata { unsigned int plla_out0_rates[NUM_RATE_TYPE]; }; +static bool need_clk_update(struct snd_soc_dai *dai) +{ + if (snd_soc_dai_is_dummy(dai) || + !dai->driver->ops || + !dai->driver->name) + return false; + + if (strstr(dai->driver->name, "I2S") || + strstr(dai->driver->name, "DMIC") || + strstr(dai->driver->name, "DSPK")) + return true; + + return false; +} + /* Setup PLL clock as per the given sample rate */ static int tegra_audio_graph_update_pll(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -140,19 +156,7 @@ static int tegra_audio_graph_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int err; - /* - * This gets called for each DAI link (FE or BE) when DPCM is used. - * We may not want to update PLLA rate for each call. So PLLA update - * must be restricted to external I/O links (I2S, DMIC or DSPK) since - * they actually depend on it. I/O modules update their clocks in - * hw_param() of their respective component driver and PLLA rate - * update here helps them to derive appropriate rates. - * - * TODO: When more HW accelerators get added (like sample rate - * converter, volume gain controller etc., which don't really - * depend on PLLA) we need a better way to filter here. - */ - if (cpu_dai->driver->ops && rtd->dai_link->no_pcm) { + if (need_clk_update(cpu_dai)) { err = tegra_audio_graph_update_pll(substream, params); if (err) return err; |