diff options
Diffstat (limited to 'sound/soc')
29 files changed, 401 insertions, 122 deletions
diff --git a/sound/soc/codecs/aw88081.c b/sound/soc/codecs/aw88081.c index ad16ab6812cd..3dd8428f08cc 100644 --- a/sound/soc/codecs/aw88081.c +++ b/sound/soc/codecs/aw88081.c @@ -1295,9 +1295,19 @@ static int aw88081_i2c_probe(struct i2c_client *i2c) aw88081_dai, ARRAY_SIZE(aw88081_dai)); } +#if defined(CONFIG_OF) +static const struct of_device_id aw88081_of_match[] = { + { .compatible = "awinic,aw88081" }, + { .compatible = "awinic,aw88083" }, + { } +}; +MODULE_DEVICE_TABLE(of, aw88081_of_match); +#endif + static struct i2c_driver aw88081_i2c_driver = { .driver = { .name = AW88081_I2C_NAME, + .of_match_table = of_match_ptr(aw88081_of_match), }, .probe = aw88081_i2c_probe, .id_table = aw88081_i2c_id, diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index ac19a572fe70..20e6ab6f0d4a 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -702,6 +702,9 @@ static void cs42l43_clear_jack(struct cs42l43_codec *priv) CS42L43_PGA_WIDESWING_MODE_EN_MASK, 0); regmap_update_bits(cs42l43->regmap, CS42L43_STEREO_MIC_CTRL, CS42L43_JACK_STEREO_CONFIG_MASK, 0); + regmap_update_bits(cs42l43->regmap, CS42L43_STEREO_MIC_CLAMP_CTRL, + CS42L43_SMIC_HPAMP_CLAMP_DIS_FRC_MASK, + CS42L43_SMIC_HPAMP_CLAMP_DIS_FRC_MASK); regmap_update_bits(cs42l43->regmap, CS42L43_HS2, CS42L43_HSDET_MODE_MASK | CS42L43_HSDET_MANUAL_MODE_MASK, 0x2 << CS42L43_HSDET_MODE_SHIFT); diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 17019b1d680b..bc01ff65bd6f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -842,12 +842,28 @@ static void print_eld_info(struct snd_info_entry *entry, static int hdmi_dai_proc_new(struct hdmi_codec_priv *hcp, struct snd_soc_dai *dai) { + struct snd_soc_component *component = dai->component; + struct snd_soc_card *card = component->card; + struct snd_soc_dai *d; + struct snd_soc_pcm_runtime *rtd; struct snd_info_entry *entry; char name[32]; - int err; + int err, i, id = 0; - snprintf(name, sizeof(name), "eld#%d", dai->id); - err = snd_card_proc_new(dai->component->card->snd_card, name, &entry); + /* + * To avoid duplicate proc entry, find its rtd and use rtd->id + * instead of dai->id + */ + for_each_card_rtds(card, rtd) { + for_each_rtd_dais(rtd, i, d) + if (d == dai) { + id = rtd->id; + goto found; + } + } +found: + snprintf(name, sizeof(name), "eld#%d", id); + err = snd_card_proc_new(card->snd_card, name, &entry); if (err < 0) return err; diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index b261fa373e65..c1fb71cfb5d0 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -63,6 +63,10 @@ #define CDC_WSA_TX_SPKR_PROT_CLK_DISABLE 0 #define CDC_WSA_TX_SPKR_PROT_PCM_RATE_MASK GENMASK(3, 0) #define CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K 0 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_16K 1 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_24K 2 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_32K 3 +#define CDC_WSA_TX_SPKR_PROT_PCM_RATE_48K 4 #define CDC_WSA_TX0_SPKR_PROT_PATH_CFG0 (0x0248) #define CDC_WSA_TX1_SPKR_PROT_PATH_CTL (0x0264) #define CDC_WSA_TX1_SPKR_PROT_PATH_CFG0 (0x0268) @@ -407,6 +411,7 @@ struct wsa_macro { int ear_spkr_gain; int spkr_gain_offset; int spkr_mode; + u32 pcm_rate_vi; int is_softclip_on[WSA_MACRO_SOFTCLIP_MAX]; int softclip_clk_users[WSA_MACRO_SOFTCLIP_MAX]; struct regmap *regmap; @@ -1280,6 +1285,7 @@ static int wsa_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); int ret; switch (substream->stream) { @@ -1292,6 +1298,11 @@ static int wsa_macro_hw_params(struct snd_pcm_substream *substream, return ret; } break; + case SNDRV_PCM_STREAM_CAPTURE: + if (dai->id == WSA_MACRO_AIF_VI) + wsa->pcm_rate_vi = params_rate(params); + + break; default: break; } @@ -1448,35 +1459,11 @@ static void wsa_macro_mclk_enable(struct wsa_macro *wsa, bool mclk_enable) } } -static int wsa_macro_mclk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static void wsa_macro_enable_disable_vi_sense(struct snd_soc_component *component, bool enable, + u32 tx_reg0, u32 tx_reg1, u32 val) { - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); - - wsa_macro_mclk_enable(wsa, event == SND_SOC_DAPM_PRE_PMU); - return 0; -} - -static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) -{ - struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); - u32 tx_reg0, tx_reg1; - - if (test_bit(WSA_MACRO_TX0, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { - tx_reg0 = CDC_WSA_TX0_SPKR_PROT_PATH_CTL; - tx_reg1 = CDC_WSA_TX1_SPKR_PROT_PATH_CTL; - } else if (test_bit(WSA_MACRO_TX1, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { - tx_reg0 = CDC_WSA_TX2_SPKR_PROT_PATH_CTL; - tx_reg1 = CDC_WSA_TX3_SPKR_PROT_PATH_CTL; - } - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - /* Enable V&I sensing */ + if (enable) { + /* Enable V&I sensing */ snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_RESET_MASK, CDC_WSA_TX_SPKR_PROT_RESET); @@ -1485,10 +1472,10 @@ static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, CDC_WSA_TX_SPKR_PROT_RESET); snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_PCM_RATE_MASK, - CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K); + val); snd_soc_component_update_bits(component, tx_reg1, CDC_WSA_TX_SPKR_PROT_PCM_RATE_MASK, - CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K); + val); snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_CLK_EN_MASK, CDC_WSA_TX_SPKR_PROT_CLK_ENABLE); @@ -1501,9 +1488,7 @@ static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, tx_reg1, CDC_WSA_TX_SPKR_PROT_RESET_MASK, CDC_WSA_TX_SPKR_PROT_NO_RESET); - break; - case SND_SOC_DAPM_POST_PMD: - /* Disable V&I sensing */ + } else { snd_soc_component_update_bits(component, tx_reg0, CDC_WSA_TX_SPKR_PROT_RESET_MASK, CDC_WSA_TX_SPKR_PROT_RESET); @@ -1516,6 +1501,72 @@ static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, tx_reg1, CDC_WSA_TX_SPKR_PROT_CLK_EN_MASK, CDC_WSA_TX_SPKR_PROT_CLK_DISABLE); + } +} + +static void wsa_macro_enable_disable_vi_feedback(struct snd_soc_component *component, + bool enable, u32 rate) +{ + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); + + if (test_bit(WSA_MACRO_TX0, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) + wsa_macro_enable_disable_vi_sense(component, enable, + CDC_WSA_TX0_SPKR_PROT_PATH_CTL, + CDC_WSA_TX1_SPKR_PROT_PATH_CTL, rate); + + if (test_bit(WSA_MACRO_TX1, &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) + wsa_macro_enable_disable_vi_sense(component, enable, + CDC_WSA_TX2_SPKR_PROT_PATH_CTL, + CDC_WSA_TX3_SPKR_PROT_PATH_CTL, rate); +} + +static int wsa_macro_mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); + + wsa_macro_mclk_enable(wsa, event == SND_SOC_DAPM_PRE_PMU); + return 0; +} + +static int wsa_macro_enable_vi_feedback(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); + u32 rate_val; + + switch (wsa->pcm_rate_vi) { + case 8000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K; + break; + case 16000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_16K; + break; + case 24000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_24K; + break; + case 32000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_32K; + break; + case 48000: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_48K; + break; + default: + rate_val = CDC_WSA_TX_SPKR_PROT_PCM_RATE_8K; + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Enable V&I sensing */ + wsa_macro_enable_disable_vi_feedback(component, true, rate_val); + break; + case SND_SOC_DAPM_POST_PMD: + /* Disable V&I sensing */ + wsa_macro_enable_disable_vi_feedback(component, false, rate_val); break; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 343e3bcef0ca..dba78efadc85 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4286,7 +4286,7 @@ static void rt5645_i2c_remove(struct i2c_client *i2c) * Since the rt5645_btn_check_callback() can queue jack_detect_work, * the timer need to be delted first */ - del_timer_sync(&rt5645->btn_check_timer); + timer_delete_sync(&rt5645->btn_check_timer); cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); @@ -4318,7 +4318,7 @@ static int rt5645_sys_suspend(struct device *dev) { struct rt5645_priv *rt5645 = dev_get_drvdata(dev); - del_timer_sync(&rt5645->btn_check_timer); + timer_delete_sync(&rt5645->btn_check_timer); cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index e0d1991cffdb..bcb6d7c6f301 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -31,9 +31,7 @@ #include "rl6231.h" #include "rt5665.h" -#define RT5665_NUM_SUPPLIES 3 - -static const char *rt5665_supply_names[RT5665_NUM_SUPPLIES] = { +static const char * const rt5665_supply_names[] = { "AVDD", "MICVDD", "VBAT", @@ -46,7 +44,6 @@ struct rt5665_priv { struct gpio_desc *gpiod_ldo1_en; struct gpio_desc *gpiod_reset; struct snd_soc_jack *hs_jack; - struct regulator_bulk_data supplies[RT5665_NUM_SUPPLIES]; struct delayed_work jack_detect_work; struct delayed_work calibrate_work; struct delayed_work jd_check_work; @@ -4471,8 +4468,6 @@ static void rt5665_remove(struct snd_soc_component *component) struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); regmap_write(rt5665->regmap, RT5665_RESET, 0); - - regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); } #ifdef CONFIG_PM @@ -4758,7 +4753,7 @@ static int rt5665_i2c_probe(struct i2c_client *i2c) { struct rt5665_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5665_priv *rt5665; - int i, ret; + int ret; unsigned int val; rt5665 = devm_kzalloc(&i2c->dev, sizeof(struct rt5665_priv), @@ -4774,24 +4769,13 @@ static int rt5665_i2c_probe(struct i2c_client *i2c) else rt5665_parse_dt(rt5665, &i2c->dev); - for (i = 0; i < ARRAY_SIZE(rt5665->supplies); i++) - rt5665->supplies[i].supply = rt5665_supply_names[i]; - - ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5665->supplies), - rt5665->supplies); + ret = devm_regulator_bulk_get_enable(&i2c->dev, ARRAY_SIZE(rt5665_supply_names), + rt5665_supply_names); if (ret != 0) { dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); return ret; } - ret = regulator_bulk_enable(ARRAY_SIZE(rt5665->supplies), - rt5665->supplies); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); - return ret; - } - - rt5665->gpiod_ldo1_en = devm_gpiod_get_optional(&i2c->dev, "realtek,ldo1-en", GPIOD_OUT_HIGH); diff --git a/sound/soc/codecs/sma1307.c b/sound/soc/codecs/sma1307.c index f5c303d4bb62..498189ab691c 100644 --- a/sound/soc/codecs/sma1307.c +++ b/sound/soc/codecs/sma1307.c @@ -1705,7 +1705,7 @@ static void sma1307_check_fault_worker(struct work_struct *work) static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *file) { const struct firmware *fw; - int *data, size, offset, num_mode; + int size, offset, num_mode; int ret; ret = request_firmware(&fw, file, sma1307->dev); @@ -1722,7 +1722,7 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil return; } - data = kzalloc(fw->size, GFP_KERNEL); + int *data __free(kfree) = kzalloc(fw->size, GFP_KERNEL); if (!data) { release_firmware(fw); sma1307->set.status = false; @@ -1742,7 +1742,6 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil sma1307->set.header_size, GFP_KERNEL); if (!sma1307->set.header) { - kfree(data); sma1307->set.status = false; return; } @@ -1763,8 +1762,6 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil = devm_kzalloc(sma1307->dev, sma1307->set.def_size * sizeof(int), GFP_KERNEL); if (!sma1307->set.def) { - kfree(data); - kfree(sma1307->set.header); sma1307->set.status = false; return; } @@ -1782,9 +1779,6 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil sma1307->set.mode_size * 2 * sizeof(int), GFP_KERNEL); if (!sma1307->set.mode_set[i]) { - kfree(data); - kfree(sma1307->set.header); - kfree(sma1307->set.def); for (int j = 0; j < i; j++) kfree(sma1307->set.mode_set[j]); sma1307->set.status = false; @@ -1799,7 +1793,6 @@ static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *fil } } - kfree(data); sma1307->set.status = true; } diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index d259e1d4d83d..1c9df7c061bd 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -568,7 +568,7 @@ static const struct sdw_port_config wsa883x_pconfig[WSA883X_MAX_SWR_PORTS] = { }, [WSA883X_PORT_VISENSE] = { .num = WSA883X_PORT_VISENSE + 1, - .ch_mask = 0x3, + .ch_mask = 0x1, }, }; diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 8051483aa1ac..daada1a2a34c 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -891,7 +891,7 @@ static const struct sdw_port_config wsa884x_pconfig[WSA884X_MAX_SWR_PORTS] = { }, [WSA884X_PORT_VISENSE] = { .num = WSA884X_PORT_VISENSE + 1, - .ch_mask = 0x3, + .ch_mask = 0x1, }, [WSA884X_PORT_CPS] = { .num = WSA884X_PORT_CPS + 1, diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 4c4171bb3fbb..28001e9857d9 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -199,12 +199,10 @@ static void i2s_start(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 1); - /* I2S needs to enable IRQ to make a handshake with DMAC on the JH7110 SoC */ - if (dev->use_pio || dev->is_jh7110) - i2s_enable_irqs(dev, substream->stream, config->chan_nr); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_enable_dma(dev, substream->stream); + i2s_enable_irqs(dev, substream->stream, config->chan_nr); i2s_write_reg(dev->i2s_base, CER, 1); } @@ -218,11 +216,12 @@ static void i2s_stop(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 0); - if (dev->use_pio || dev->is_jh7110) - i2s_disable_irqs(dev, substream->stream, 8); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_disable_dma(dev, substream->stream); + i2s_disable_irqs(dev, substream->stream, 8); + + if (!dev->active) { i2s_write_reg(dev->i2s_base, CER, 0); i2s_write_reg(dev->i2s_base, IER, 0); diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index f501f47242fb..1bba48318e2d 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -156,11 +156,24 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, for_each_dpcm_be(rtd, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *substream_be; - struct snd_soc_dai *dai = snd_soc_rtd_to_cpu(be, 0); + struct snd_soc_dai *dai_cpu = snd_soc_rtd_to_cpu(be, 0); + struct snd_soc_dai *dai_codec = snd_soc_rtd_to_codec(be, 0); + struct snd_soc_dai *dai; if (dpcm->fe != rtd) continue; + /* + * With audio graph card, original cpu dai is changed to codec + * device in backend, so if cpu dai is dummy device in backend, + * get the codec dai device, which is the real hardware device + * connected. + */ + if (!snd_soc_dai_is_dummy(dai_cpu)) + dai = dai_cpu; + else + dai = dai_codec; + substream_be = snd_soc_dpcm_get_substream(be, stream); dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be); dev_be = dai->dev; diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c index b2979290c973..5614a8b909ed 100644 --- a/sound/soc/fsl/fsl_qmc_audio.c +++ b/sound/soc/fsl/fsl_qmc_audio.c @@ -250,6 +250,9 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component *component, switch (cmd) { case SNDRV_PCM_TRIGGER_START: bitmap_zero(prtd->chans_pending, 64); + prtd->buffer_ended = 0; + prtd->ch_dma_addr_current = prtd->ch_dma_addr_start; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { for (i = 0; i < prtd->channels; i++) prtd->qmc_dai->chans[i].prtd_tx = prtd; diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 905294682996..3686d468506b 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -772,6 +772,8 @@ static int imx_card_probe(struct platform_device *pdev) data->dapm_routes[i].sink = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%d %s", i + 1, "Playback"); + if (!data->dapm_routes[i].sink) + return -ENOMEM; data->dapm_routes[i].source = "CPU-Playback"; } } @@ -789,6 +791,8 @@ static int imx_card_probe(struct platform_device *pdev) data->dapm_routes[i].source = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%d %s", i + 1, "Capture"); + if (!data->dapm_routes[i].source) + return -ENOMEM; data->dapm_routes[i].sink = "CPU-Capture"; } } diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index 1daf0be3d100..de5f87600fac 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -301,7 +301,7 @@ static int imx_rpmsg_pcm_close(struct snd_soc_component *component, info->send_message(msg, info); - del_timer(&info->stream_timer[substream->stream].timer); + timer_delete(&info->stream_timer[substream->stream].timer); rtd->dai_link->ignore_suspend = 0; @@ -452,7 +452,7 @@ static int imx_rpmsg_terminate_all(struct snd_soc_component *component, info->msg[RX_POINTER].r_msg.param.buffer_offset = 0; } - del_timer(&info->stream_timer[substream->stream].timer); + timer_delete(&info->stream_timer[substream->stream].timer); return imx_rpmsg_insert_workqueue(substream, msg, info); } diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index ef0c1d125d66..cafb8c6198be 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -115,6 +115,78 @@ avs_path_find_variant(struct avs_dev *adev, return NULL; } +static struct acpi_nhlt_config * +avs_nhlt_config_or_default(struct avs_dev *adev, struct avs_tplg_module *t); + +int avs_path_set_constraint(struct avs_dev *adev, struct avs_tplg_path_template *template, + struct snd_pcm_hw_constraint_list *rate_list, + struct snd_pcm_hw_constraint_list *channels_list, + struct snd_pcm_hw_constraint_list *sample_bits_list) +{ + struct avs_tplg_path *path_template; + unsigned int *rlist, *clist, *slist; + size_t i; + + i = 0; + list_for_each_entry(path_template, &template->path_list, node) + i++; + + rlist = kcalloc(i, sizeof(rlist), GFP_KERNEL); + clist = kcalloc(i, sizeof(clist), GFP_KERNEL); + slist = kcalloc(i, sizeof(slist), GFP_KERNEL); + + i = 0; + list_for_each_entry(path_template, &template->path_list, node) { + struct avs_tplg_pipeline *pipeline_template; + + list_for_each_entry(pipeline_template, &path_template->ppl_list, node) { + struct avs_tplg_module *module_template; + + list_for_each_entry(module_template, &pipeline_template->mod_list, node) { + const guid_t *type = &module_template->cfg_ext->type; + struct acpi_nhlt_config *blob; + + if (!guid_equal(type, &AVS_COPIER_MOD_UUID) && + !guid_equal(type, &AVS_WOVHOSTM_MOD_UUID)) + continue; + + switch (module_template->cfg_ext->copier.dma_type) { + case AVS_DMA_DMIC_LINK_INPUT: + case AVS_DMA_I2S_LINK_OUTPUT: + case AVS_DMA_I2S_LINK_INPUT: + break; + default: + continue; + } + + blob = avs_nhlt_config_or_default(adev, module_template); + if (IS_ERR(blob)) + continue; + + rlist[i] = path_template->fe_fmt->sampling_freq; + clist[i] = path_template->fe_fmt->num_channels; + slist[i] = path_template->fe_fmt->bit_depth; + i++; + } + } + } + + if (i) { + rate_list->count = i; + rate_list->list = rlist; + channels_list->count = i; + channels_list->list = clist; + sample_bits_list->count = i; + sample_bits_list->list = slist; + } else { + kfree(rlist); + kfree(clist); + kfree(slist); + } + + return i; +} + static void avs_init_node_id(union avs_connector_node_id *node_id, struct avs_tplg_modcfg_ext *te, u32 dma_id) { diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h index 7ed7e94e0a56..c65ed84aa853 100644 --- a/sound/soc/intel/avs/path.h +++ b/sound/soc/intel/avs/path.h @@ -69,6 +69,11 @@ int avs_path_reset(struct avs_path *path); int avs_path_pause(struct avs_path *path); int avs_path_run(struct avs_path *path, int trigger); +int avs_path_set_constraint(struct avs_dev *adev, struct avs_tplg_path_template *template, + struct snd_pcm_hw_constraint_list *rate_list, + struct snd_pcm_hw_constraint_list *channels_list, + struct snd_pcm_hw_constraint_list *sample_bits_list); + int avs_peakvol_set_volume(struct avs_dev *adev, struct avs_path_module *mod, struct soc_mixer_control *mc, long *input); int avs_peakvol_set_mute(struct avs_dev *adev, struct avs_path_module *mod, diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index dac463390da1..d83ef504643b 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -31,6 +31,10 @@ struct avs_dma_data { struct hdac_ext_stream *host_stream; }; + struct snd_pcm_hw_constraint_list rate_list; + struct snd_pcm_hw_constraint_list channels_list; + struct snd_pcm_hw_constraint_list sample_bits_list; + struct work_struct period_elapsed_work; struct snd_pcm_substream *substream; }; @@ -74,6 +78,45 @@ void avs_period_elapsed(struct snd_pcm_substream *substream) schedule_work(&data->period_elapsed_work); } +static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule); +static int avs_hw_constraints_init(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hw_constraint_list *r, *c, *s; + struct avs_tplg_path_template *template; + struct avs_dma_data *data; + int ret; + + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + data = snd_soc_dai_get_dma_data(dai, substream); + r = &(data->rate_list); + c = &(data->channels_list); + s = &(data->sample_bits_list); + + template = avs_dai_find_path_template(dai, !rtd->dai_link->no_pcm, substream->stream); + ret = avs_path_set_constraint(data->adev, template, r, c, s); + if (ret <= 0) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, r); + if (ret < 0) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, c); + if (ret < 0) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, s); + if (ret < 0) + return ret; + + return 0; +} + static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); @@ -101,7 +144,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d if (rtd->dai_link->ignore_suspend) adev->num_lp_paths++; - return 0; + return avs_hw_constraints_init(substream, dai); } static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -114,6 +157,10 @@ static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc if (rtd->dai_link->ignore_suspend) data->adev->num_lp_paths--; + kfree(data->rate_list.list); + kfree(data->channels_list.list); + kfree(data->sample_bits_list.list); + snd_soc_dai_set_dma_data(dai, substream, NULL); kfree(data); } @@ -927,7 +974,8 @@ static int avs_component_probe(struct snd_soc_component *component) else mach->tplg_filename = devm_kasprintf(adev->dev, GFP_KERNEL, "hda-generic-tplg.bin"); - + if (!mach->tplg_filename) + return -ENOMEM; filename = kasprintf(GFP_KERNEL, "%s/%s", component->driver->topology_name_prefix, mach->tplg_filename); if (!filename) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 90dafa810b2e..095d08b3fc82 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -764,6 +764,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { static const struct snd_pci_quirk sof_sdw_ssid_quirk_table[] = { SND_PCI_QUIRK(0x1043, 0x1e13, "ASUS Zenbook S14", SOC_SDW_CODEC_MIC), + SND_PCI_QUIRK(0x1043, 0x1f43, "ASUS Zenbook S16", SOC_SDW_CODEC_MIC), {} }; diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 27a2bf9a6613..de3ec6f594c1 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -13,10 +13,11 @@ #include <linux/platform_device.h> #include <linux/regmap.h> #include <dt-bindings/sound/qcom,lpass.h> +#include <dt-bindings/sound/qcom,q6afe.h> #include "lpass-hdmi.h" #define LPASS_AHBIX_CLOCK_FREQUENCY 131072000 -#define LPASS_MAX_PORTS (LPASS_CDC_DMA_VA_TX8 + 1) +#define LPASS_MAX_PORTS (DISPLAY_PORT_RX_7 + 1) #define LPASS_MAX_MI2S_PORTS (8) #define LPASS_MAX_DMA_CHANNELS (8) #define LPASS_MAX_HDMI_DMA_CHANNELS (4) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index c9404b5934c7..2cd522108221 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -24,8 +24,8 @@ #define PLAYBACK_MIN_PERIOD_SIZE 128 #define CAPTURE_MIN_NUM_PERIODS 2 #define CAPTURE_MAX_NUM_PERIODS 8 -#define CAPTURE_MAX_PERIOD_SIZE 4096 -#define CAPTURE_MIN_PERIOD_SIZE 320 +#define CAPTURE_MAX_PERIOD_SIZE 65536 +#define CAPTURE_MIN_PERIOD_SIZE 6144 #define BUFFER_BYTES_MAX (PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE) #define BUFFER_BYTES_MIN (PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE) #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) @@ -64,12 +64,12 @@ struct q6apm_dai_rtd { phys_addr_t phys; unsigned int pcm_size; unsigned int pcm_count; - unsigned int pos; /* Buffer position */ unsigned int periods; unsigned int bytes_sent; unsigned int bytes_received; unsigned int copied_total; uint16_t bits_per_sample; + snd_pcm_uframes_t queue_ptr; bool next_track; enum stream_state state; struct q6apm_graph *graph; @@ -123,25 +123,16 @@ static void event_handler(uint32_t opcode, uint32_t token, void *payload, void * { struct q6apm_dai_rtd *prtd = priv; struct snd_pcm_substream *substream = prtd->substream; - unsigned long flags; switch (opcode) { case APM_CLIENT_EVENT_CMD_EOS_DONE: prtd->state = Q6APM_STREAM_STOPPED; break; case APM_CLIENT_EVENT_DATA_WRITE_DONE: - spin_lock_irqsave(&prtd->lock, flags); - prtd->pos += prtd->pcm_count; - spin_unlock_irqrestore(&prtd->lock, flags); snd_pcm_period_elapsed(substream); - if (prtd->state == Q6APM_STREAM_RUNNING) - q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, 0); break; case APM_CLIENT_EVENT_DATA_READ_DONE: - spin_lock_irqsave(&prtd->lock, flags); - prtd->pos += prtd->pcm_count; - spin_unlock_irqrestore(&prtd->lock, flags); snd_pcm_period_elapsed(substream); if (prtd->state == Q6APM_STREAM_RUNNING) q6apm_read(prtd->graph); @@ -248,7 +239,6 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, } prtd->pcm_count = snd_pcm_lib_period_bytes(substream); - prtd->pos = 0; /* rate and channels are sent to audio driver */ ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg); if (ret < 0) { @@ -294,6 +284,27 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, return 0; } +static int q6apm_dai_ack(struct snd_soc_component *component, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct q6apm_dai_rtd *prtd = runtime->private_data; + int i, ret = 0, avail_periods; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + avail_periods = (runtime->control->appl_ptr - prtd->queue_ptr)/runtime->period_size; + for (i = 0; i < avail_periods; i++) { + ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, NO_TIMESTAMP); + if (ret < 0) { + dev_err(component->dev, "Error queuing playback buffer %d\n", ret); + return ret; + } + prtd->queue_ptr += runtime->period_size; + } + } + + return ret; +} + static int q6apm_dai_trigger(struct snd_soc_component *component, struct snd_pcm_substream *substream, int cmd) { @@ -305,9 +316,6 @@ static int q6apm_dai_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - /* start writing buffers for playback only as we already queued capture buffers */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: /* TODO support be handled via SoftPause Module */ @@ -377,13 +385,14 @@ static int q6apm_dai_open(struct snd_soc_component *component, } } - ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + /* setup 10ms latency to accommodate DSP restrictions */ + ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 480); if (ret < 0) { dev_err(dev, "constraint for period bytes step ret = %d\n", ret); goto err; } - ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); + ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 480); if (ret < 0) { dev_err(dev, "constraint for buffer bytes step ret = %d\n", ret); goto err; @@ -428,16 +437,12 @@ static snd_pcm_uframes_t q6apm_dai_pointer(struct snd_soc_component *component, struct snd_pcm_runtime *runtime = substream->runtime; struct q6apm_dai_rtd *prtd = runtime->private_data; snd_pcm_uframes_t ptr; - unsigned long flags; - spin_lock_irqsave(&prtd->lock, flags); - if (prtd->pos == prtd->pcm_size) - prtd->pos = 0; - - ptr = bytes_to_frames(runtime, prtd->pos); - spin_unlock_irqrestore(&prtd->lock, flags); + ptr = q6apm_get_hw_pointer(prtd->graph, substream->stream) * runtime->period_size; + if (ptr) + return ptr - 1; - return ptr; + return 0; } static int q6apm_dai_hw_params(struct snd_soc_component *component, @@ -652,8 +657,6 @@ static int q6apm_dai_compr_set_params(struct snd_soc_component *component, prtd->pcm_size = runtime->fragments * runtime->fragment_size; prtd->bits_per_sample = 16; - prtd->pos = 0; - if (prtd->next_track != true) { memcpy(&prtd->codec, codec, sizeof(*codec)); @@ -836,6 +839,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = { .hw_params = q6apm_dai_hw_params, .pointer = q6apm_dai_pointer, .trigger = q6apm_dai_trigger, + .ack = q6apm_dai_ack, .compress_ops = &q6apm_dai_compress_ops, .use_dai_pcm_id = true, }; diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 11e252a70f69..b4ffa0f0b188 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -494,6 +494,19 @@ int q6apm_read(struct q6apm_graph *graph) } EXPORT_SYMBOL_GPL(q6apm_read); +int q6apm_get_hw_pointer(struct q6apm_graph *graph, int dir) +{ + struct audioreach_graph_data *data; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + data = &graph->rx_data; + else + data = &graph->tx_data; + + return (int)atomic_read(&data->hw_ptr); +} +EXPORT_SYMBOL_GPL(q6apm_get_hw_pointer); + static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) { struct data_cmd_rsp_rd_sh_mem_ep_data_buffer_done_v2 *rd_done; @@ -520,7 +533,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) done = data->payload; phys = graph->rx_data.buf[token].phys; mutex_unlock(&graph->lock); - + /* token numbering starts at 0 */ + atomic_set(&graph->rx_data.hw_ptr, token + 1); if (lower_32_bits(phys) == done->buf_addr_lsw && upper_32_bits(phys) == done->buf_addr_msw) { graph->result.opcode = hdr->opcode; @@ -553,6 +567,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) rd_done = data->payload; phys = graph->tx_data.buf[hdr->token].phys; mutex_unlock(&graph->lock); + /* token numbering starts at 0 */ + atomic_set(&graph->tx_data.hw_ptr, hdr->token + 1); if (upper_32_bits(phys) == rd_done->buf_addr_msw && lower_32_bits(phys) == rd_done->buf_addr_lsw) { diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h index c248c8d2b1ab..7ce08b401e31 100644 --- a/sound/soc/qcom/qdsp6/q6apm.h +++ b/sound/soc/qcom/qdsp6/q6apm.h @@ -2,6 +2,7 @@ #ifndef __Q6APM_H__ #define __Q6APM_H__ #include <linux/types.h> +#include <linux/atomic.h> #include <linux/slab.h> #include <linux/wait.h> #include <linux/kernel.h> @@ -77,6 +78,7 @@ struct audioreach_graph_data { uint32_t num_periods; uint32_t dsp_buf; uint32_t mem_map_handle; + atomic_t hw_ptr; }; struct audioreach_graph { @@ -150,4 +152,5 @@ int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples); int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples); int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph, uint32_t codec_id); +int q6apm_get_hw_pointer(struct q6apm_graph *graph, int dir); #endif /* __APM_GRAPH_ */ diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 045100c94352..a400c9a31fea 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -892,9 +892,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, if (ret < 0) { dev_err(dev, "q6asm_open_write failed\n"); - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; - return ret; + goto open_err; } } @@ -903,7 +901,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, prtd->session_id, dir); if (ret) { dev_err(dev, "Stream reg failed ret:%d\n", ret); - return ret; + goto q6_err; } ret = __q6asm_dai_compr_set_codec_params(component, stream, @@ -911,7 +909,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, prtd->stream_id); if (ret) { dev_err(dev, "codec param setup failed ret:%d\n", ret); - return ret; + goto q6_err; } ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, @@ -920,12 +918,21 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, if (ret < 0) { dev_err(dev, "Buffer Mapping failed ret:%d\n", ret); - return -ENOMEM; + ret = -ENOMEM; + goto q6_err; } prtd->state = Q6ASM_STREAM_RUNNING; return 0; + +q6_err: + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); + +open_err: + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return ret; } static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component, diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index fae3598fd601..dc1d21de4ab7 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -345,6 +345,7 @@ endif ## SND_SOC_SOF_HDA_GENERIC config SND_SOF_SOF_HDA_SDW_BPT tristate + select SND_HDA_EXT_CORE help This option is not user-selectable but automagically handled by 'select' statements at a higher level. diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index ccf8eefdca70..f64e8a6a9a33 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -991,6 +991,10 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) if (!sdev->dspless_mode_selected) { /* cancel any attempt for DSP D0I3 */ cancel_delayed_work_sync(&hda->d0i3_work); + + /* Cancel the microphone privacy work if mic privacy is active */ + if (hda->mic_privacy.active) + cancel_work_sync(&hda->mic_privacy.work); } /* stop hda controller and power dsp off */ @@ -1017,6 +1021,10 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) if (!sdev->dspless_mode_selected) { /* cancel any attempt for DSP D0I3 */ cancel_delayed_work_sync(&hda->d0i3_work); + + /* Cancel the microphone privacy work if mic privacy is active */ + if (hda->mic_privacy.active) + cancel_work_sync(&hda->mic_privacy.work); } if (target_state == SOF_DSP_PM_D0) { diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 6b1ada566476..b34e5fdf10f1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -968,6 +968,10 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) if (sdev->dspless_mode_selected) goto skip_disable_dsp; + /* Cancel the microphone privacy work if mic privacy is active */ + if (hda->mic_privacy.active) + cancel_work_sync(&hda->mic_privacy.work); + /* no need to check for error as the DSP will be disabled anyway */ if (chip && chip->power_down_dsp) chip->power_down_dsp(sdev); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 76154627fc17..108cad04879e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -487,6 +487,11 @@ enum sof_hda_D0_substate { SOF_HDA_DSP_PM_D0I3, /* low power D0 substate */ }; +struct sof_ace3_mic_privacy { + bool active; + struct work_struct work; +}; + /* represents DSP HDA controller frontend - i.e. host facing control */ struct sof_intel_hda_dev { bool imrboot_supported; @@ -542,6 +547,9 @@ struct sof_intel_hda_dev { /* Intel NHLT information */ struct nhlt_acpi_table *nhlt; + /* work queue for mic privacy state change notification sending */ + struct sof_ace3_mic_privacy mic_privacy; + /* * Pointing to the IPC message if immediate sending was not possible * because the downlink communication channel was BUSY at the time. diff --git a/sound/soc/sof/intel/ptl.c b/sound/soc/sof/intel/ptl.c index 8fa4bdceedd9..aa0b772178bc 100644 --- a/sound/soc/sof/intel/ptl.c +++ b/sound/soc/sof/intel/ptl.c @@ -27,22 +27,44 @@ static bool sof_ptl_check_mic_privacy_irq(struct snd_sof_dev *sdev, bool alt, return hdac_bus_eml_is_mic_privacy_changed(sof_to_bus(sdev), alt, elid); } +static void sof_ptl_mic_privacy_work(struct work_struct *work) +{ + struct sof_intel_hda_dev *hdev = container_of(work, + struct sof_intel_hda_dev, + mic_privacy.work); + struct hdac_bus *bus = &hdev->hbus.core; + struct snd_sof_dev *sdev = dev_get_drvdata(bus->dev); + bool state; + + /* + * The microphone privacy state is only available via Soundwire shim + * in PTL + * The work is only scheduled on change. + */ + state = hdac_bus_eml_get_mic_privacy_state(bus, 1, + AZX_REG_ML_LEPTR_ID_SDW); + sof_ipc4_mic_privacy_state_change(sdev, state); +} + static void sof_ptl_process_mic_privacy(struct snd_sof_dev *sdev, bool alt, int elid) { - bool state; + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; if (!alt || elid != AZX_REG_ML_LEPTR_ID_SDW) return; - state = hdac_bus_eml_get_mic_privacy_state(sof_to_bus(sdev), alt, elid); - - sof_ipc4_mic_privacy_state_change(sdev, state); + /* + * Schedule the work to read the microphone privacy state and send IPC + * message about the new state to the firmware + */ + schedule_work(&hdev->mic_privacy.work); } static void sof_ptl_set_mic_privacy(struct snd_sof_dev *sdev, struct sof_ipc4_intel_mic_privacy_cap *caps) { + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; u32 micpvcp; if (!caps || !caps->capabilities_length) @@ -58,6 +80,9 @@ static void sof_ptl_set_mic_privacy(struct snd_sof_dev *sdev, hdac_bus_eml_set_mic_privacy_mask(sof_to_bus(sdev), true, AZX_REG_ML_LEPTR_ID_SDW, PTL_MICPVCP_GET_SDW_MASK(micpvcp)); + + INIT_WORK(&hdev->mic_privacy.work, sof_ptl_mic_privacy_work); + hdev->mic_privacy.active = true; } int sof_ptl_set_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *dsp_ops) diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 8a4423646aea..9b8cb80ec81a 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -303,7 +303,7 @@ static void cx81801_close(struct tty_struct *tty) struct snd_soc_component *component = tty->disc_data; struct snd_soc_dapm_context *dapm; - del_timer_sync(&cx81801_timer); + timer_delete_sync(&cx81801_timer); /* Prevent the hook switch from further changing the DAPM pins */ INIT_LIST_HEAD(&ams_delta_hook_switch.pins); |