Alsa-devel August 2018

alsa-devel@alsa-project.org

109 participants
267 discussions

[alsa-devel] [PATCH] ASoC: core: add support to card rebind
by Srinivas Kandagatla 10 Sep '18

10 Sep '18

Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi Mark, This time I tried doing this without component framework, by maintaining list of ubind card in ASoC core that are still not unregistered. This list is tried to bind for every successful add of component. It looks much better than the component framework one! thanks, srini include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 86 ++++++++++++++++++++++++++++++-------------- 2 files changed, 61 insertions(+), 27 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a4915148f739..d9a8aea213cf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1092,6 +1092,7 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head component_dev_list; + struct list_head list; struct list_head widgets; struct list_head paths; @@ -1359,6 +1360,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_list); INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); + INIT_LIST_HEAD(&card->list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6d33634b934b..795796978fc8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -52,6 +52,7 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); +static LIST_HEAD(unbind_card_list); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -2678,6 +2679,34 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_bind_card(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + int ret; + + ret = snd_soc_instantiate_card(card); + if (ret != 0) + return ret; + + /* deactivate pins to sleep state */ + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + + return ret; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2687,7 +2716,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; - struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2719,28 +2747,23 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int j; - - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } + return snd_soc_bind_card(card); +} +EXPORT_SYMBOL_GPL(snd_soc_register_card); - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); } - - return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core @@ -2750,12 +2773,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) { - card->instantiated = false; - snd_soc_dapm_shutdown(card); - soc_cleanup_card_resources(card); - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - } + snd_soc_unbind_card(card, true); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -3101,7 +3120,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) struct snd_soc_card *card = component->card; if (card) - snd_soc_unregister_card(card); + snd_soc_unbind_card(card, false); list_del(&component->list); } @@ -3141,6 +3160,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream) stream->formats |= endianness_format_map[i]; } +static void snd_soc_try_rebind_card(void) +{ + struct snd_soc_card *card, *c; + + if (!list_empty(&unbind_card_list)) { + list_for_each_entry_safe(card, c, &unbind_card_list, list) { + if (!snd_soc_bind_card(card)) + list_del(&card->list); + } + } +} + int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -3168,6 +3199,7 @@ int snd_soc_add_component(struct device *dev, } snd_soc_component_add(component); + snd_soc_try_rebind_card(); return 0; -- 2.18.0

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[alsa-devel] [PATCH v2] ASoC: Intel: hdac_hdmi: Limit sampling rates at dai creation
by Yong Zhi 07 Sep '18

07 Sep '18

Playback of 44.1Khz contents with HDMI plugged returns "Invalid pipe config" because HDMI paths in the FW topology are configured to operate at 48Khz. This patch filters out sampling rates not supported at hdac_hdmi_create_dais() to let user space SRC to do the converting. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- v2: - Add Reviewed-by tag. - Use (and tested) Takashi's sample code in his v1 review. sound/soc/codecs/hdac_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..590dc1e33a60 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, if (ret) return ret; + /* Filter out 44.1, 88.2 and 176.4Khz */ + rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + if (!rates) + return -EINVAL; + sprintf(dai_name, "intel-hdmi-hifi%d", i+1); hdmi_dais[i].name = devm_kstrdup(&hdev->dev, dai_name, GFP_KERNEL); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: Add device tree binding for r8a77990
by Yoshihiro Kaneko 06 Sep '18

06 Sep '18

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> This patch adds the device tree binding of the r8a77990 SoC. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Yoshihiro Kaneko <ykaneko0929(a)gmail.com> --- This patch is based on the devel branch of Simon Horman's renesas tree. Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790..07aba8d 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77990" (R-Car E3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 1.9.1

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[alsa-devel] [PATCH 0/1] Fix ADI axi-i2s + Xilinx AXI-DMA capture
by Luca Ceresoli 06 Sep '18

06 Sep '18

Hi, here is a fix for a nasty audio capture problem when the axi-i2s output stream is fed to a Xilinx AXI-DMA. The commit is simple and hopefully well described, but I am not 100% sure the solution is the correct one. If it is not, I'll be glad to know which one is the best. It solved my problem anyway. Luca Luca Ceresoli (1): axi-i2s: set period size register sound/soc/adi/axi-i2s.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) -- 2.17.1

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Add MICBIAS off setting
by Andrew F. Davis 04 Sep '18

04 Sep '18

Leaving microphone bias off is a valid setting and even used in the DT binding document example. Add this setting here and document the same. Although it may not make much sense to enable a microphone here without any bias, it is a valid setting that can be chosen by DT and may be needed for some boards. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 1 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 1 + 3 files changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt index 5b3c33bb99e5..411cc46a2c58 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -24,6 +24,7 @@ Optional properties: - reset-gpios - GPIO specification for the active low RESET input. - ai31xx-micbias-vg - MicBias Voltage setting + 0 or MICBIAS_OFF - MICBIAS output is powered off 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h index c6895a18a455..069484070fcf 100644 --- a/include/dt-bindings/sound/tlv320aic31xx-micbias.h +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -2,6 +2,7 @@ #ifndef __DT_TLV320AIC31XX_MICBIAS_H #define __DT_TLV320AIC31XX_MICBIAS_H +#define MICBIAS_OFF 0 #define MICBIAS_2_0V 1 #define MICBIAS_2_5V 2 #define MICBIAS_AVDDV 3 diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index bf92d36b8f8a..7d87df518fed 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1421,6 +1421,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, fwnode_property_read_u32(aic31xx->dev->fwnode, "ai31xx-micbias-vg", &micbias_value); switch (micbias_value) { + case MICBIAS_OFF: case MICBIAS_2_0V: case MICBIAS_2_5V: case MICBIAS_AVDDV: -- 2.18.0

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Remove regulator notification handling
by Andrew F. Davis 04 Sep '18

04 Sep '18

The notification handler sets the reset line but never re-enables it as no one is listening for an enable event, this is certainly broken and was most likely just copied from other CODECs, remove this code. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 55 +------------------------------- 1 file changed, 1 insertion(+), 54 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7d87df518fed..f8d0f3ebed9c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -147,11 +147,6 @@ static const char * const aic31xx_supply_names[] = { #define AIC31XX_NUM_SUPPLIES ARRAY_SIZE(aic31xx_supply_names) -struct aic31xx_disable_nb { - struct notifier_block nb; - struct aic31xx_priv *aic31xx; -}; - struct aic31xx_priv { struct snd_soc_component *component; u8 i2c_regs_status; @@ -162,7 +157,6 @@ struct aic31xx_priv { int micbias_vg; struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; - struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1116,28 +1110,6 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static int aic31xx_regulator_event(struct notifier_block *nb, - unsigned long event, void *data) -{ - struct aic31xx_disable_nb *disable_nb = - container_of(nb, struct aic31xx_disable_nb, nb); - struct aic31xx_priv *aic31xx = disable_nb->aic31xx; - - if (event & REGULATOR_EVENT_DISABLE) { - /* - * Put codec to reset and as at least one of the - * supplies was disabled. - */ - if (aic31xx->gpio_reset) - gpiod_set_value(aic31xx->gpio_reset, 1); - - regcache_mark_dirty(aic31xx->regmap); - dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); - } - - return 0; -} - static int aic31xx_reset(struct aic31xx_priv *aic31xx) { int ret = 0; @@ -1263,26 +1235,12 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i, ret; + int ret; dev_dbg(aic31xx->dev, "## %s\n", __func__); aic31xx->component = component; - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { - aic31xx->disable_nb[i].nb.notifier_call = - aic31xx_regulator_event; - aic31xx->disable_nb[i].aic31xx = aic31xx; - ret = regulator_register_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); - if (ret) { - dev_err(component->dev, - "Failed to request regulator notifier: %d\n", - ret); - return ret; - } - } - regcache_cache_only(aic31xx->regmap, true); regcache_mark_dirty(aic31xx->regmap); @@ -1297,19 +1255,8 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) return 0; } -static void aic31xx_codec_remove(struct snd_soc_component *component) -{ - struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i; - - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) - regulator_unregister_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); -} - static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, - .remove = aic31xx_codec_remove, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), -- 2.18.0

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[alsa-devel] [PATCH 1/4] ASoC: tlv320aic31xx: Add short circuit detection support
by Andrew F. Davis 04 Sep '18

04 Sep '18

These devices support detecting and reporting short circuits across the output stages. Add support for reporting these issue. Do this by registering an interrupt if available and enabling this error to trigger that interrupt in the device. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 55 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 16 ++++++++++ 2 files changed, 71 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8d0f3ebed9c..d8783af7d31c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -161,6 +161,7 @@ struct aic31xx_priv { u8 p_div; int rate_div_line; bool master_dapm_route_applied; + int irq; }; struct aic31xx_rate_divs { @@ -1338,6 +1339,40 @@ static const struct acpi_device_id aic31xx_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); #endif +static irqreturn_t aic31xx_irq(int irq, void *data) +{ + struct aic31xx_priv *aic31xx = data; + struct device *dev = aic31xx->dev; + unsigned int value; + bool handled = false; + int ret; + + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_HPLSCDETECT) + dev_err(dev, "Short circuit on Left output is detected\n"); + if (value & AIC31XX_HPRSCDETECT) + dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & ~(AIC31XX_HPLSCDETECT | + AIC31XX_HPRSCDETECT)) + dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); + +exit: + if (handled) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1360,6 +1395,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } aic31xx->dev = &i2c->dev; + aic31xx->irq = i2c->irq; aic31xx->codec_type = id->driver_data; @@ -1404,6 +1440,25 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } + if (aic31xx->irq > 0) { + regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1, + AIC31XX_GPIO1_FUNC_MASK, + AIC31XX_GPIO1_INT1 << + AIC31XX_GPIO1_FUNC_SHIFT); + + regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_SC); + + ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, + NULL, aic31xx_irq, + IRQF_ONESHOT, "aic31xx-irq", + aic31xx); + if (ret) { + dev_err(aic31xx->dev, "Unable to request IRQ\n"); + return ret; + } + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0b587585b38b..52e171988906 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -191,6 +191,22 @@ struct aic31xx_pdata { #define AIC31XX_SC BIT(3) #define AIC31XX_ENGINE BIT(2) +/* AIC31XX_GPIO1 */ +#define AIC31XX_GPIO1_FUNC_MASK GENMASK(5, 2) +#define AIC31XX_GPIO1_FUNC_SHIFT 2 +#define AIC31XX_GPIO1_DISABLED 0x00 +#define AIC31XX_GPIO1_INPUT 0x01 +#define AIC31XX_GPIO1_GPI 0x02 +#define AIC31XX_GPIO1_GPO 0x03 +#define AIC31XX_GPIO1_CLKOUT 0x04 +#define AIC31XX_GPIO1_INT1 0x05 +#define AIC31XX_GPIO1_INT2 0x06 +#define AIC31XX_GPIO1_ADC_WCLK 0x07 +#define AIC31XX_GPIO1_SBCLK 0x08 +#define AIC31XX_GPIO1_SWCLK 0x09 +#define AIC31XX_GPIO1_ADC_MOD_CLK 0x10 +#define AIC31XX_GPIO1_SDOUT 0x11 + /* AIC31XX_DACSETUP */ #define AIC31XX_SOFTSTEP_MASK GENMASK(1, 0) -- 2.18.0

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[alsa-devel] Possibility of routing at the kernel level
by Mark Farrugia 04 Sep '18

04 Sep '18

Hi all, I am currently using the ALSA loopback driver to create a "virtual" sound card to abstract underlying real sound cards. I capture from the loopback in user-space, and then subsequently playback to a real sound card. The issue with this is that there is a significant latency introduced (~100ms) due to the number of transitions of the stream from user to kernel space. I am seeking a way to avoid this. My question is this: is there any way of routing PCM substreams at the kernel level, so that we can avoid the need to pipe audio back up to user-space with the loopback capturer? If there is no current solution that would achieve this, I would proposition to design and implement a system that uses a top-level ALSA driver, which then routes dynamically to one or many subscribed lower-level "audio path" drivers. The system would also have the option to loopback upon itself. Does anyone see any reason that I couldn't (or shouldn't) attempt such a solution? All feedback is welcome. Regards, Mark

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[alsa-devel] [PATCH v2] sound: Convert to using %pOFn instead of device_node.name
by Rob Herring 03 Sep '18

03 Sep '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - Fix error in strncmp sound/aoa/soundbus/i2sbus/core.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 000b58522106..bd7c5029fc59 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child = NULL, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - static const char *rnames[] = { "i2sbus: %s (control)", - "i2sbus: %s (tx)", - "i2sbus: %s (rx)" }; + char node_name[6]; + static const char *rnames[] = { "i2sbus: %pOFn (control)", + "i2sbus: %pOFn (tx)", + "i2sbus: %pOFn (rx)" }; static irq_handler_t ints[] = { i2sbus_bus_intr, i2sbus_tx_intr, i2sbus_rx_intr }; - if (strlen(np->name) != 5) + if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5) return 0; - if (strncmp(np->name, "i2s-", 4)) + if (strncmp(node_name, "i2s-", 4)) return 0; dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL); @@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio, dev->sound.pcmid = -1; dev->macio = macio; dev->control = control; - dev->bus_number = np->name[4] - 'a'; + dev->bus_number = node_name[4] - 'a'; INIT_LIST_HEAD(&dev->sound.codec_list); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { dev->interrupts[i] = -1; snprintf(dev->rnames[i], sizeof(dev->rnames[i]), - rnames[i], np->name); + rnames[i], np); } for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { int irq = irq_of_parse_and_map(np, i); -- 2.17.1

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[alsa-devel] [PATCH] ASoC: core: Don't schedule DAPM work if already in target state
by Jon Hunter 03 Sep '18

03 Sep '18

When dapm_power_widgets() is called, the dapm_pre_sequence_async() and dapm_post_sequence_async() functions are scheduled for all DAPM contexts (apart from the card DAPM context) regardless of whether the DAPM context is already in the desired state. The overhead of this is not insignificant and the more DAPM contexts there are the more overhead there is. For example, on the Tegra124 Jetson TK1, when profiling the time taken to execute the dapm_power_widgets() the following times were observed. Times for function dapm_power_widgets() are (us): Min 23, Ave 190, Max 434, Count 39 Here 'Count' is the number of times that dapm_power_widgets() has been called. Please note that the above time were measured using ktime_get() to log the time on entry and exit from dapm_power_widgets(). So it should be noted that these times may not be purely the time take to execute this function if it is preempted. However, after applying this patch and measuring the time taken to execute dapm_power_widgets() again a significant improvement is seen as shown below. Times for function dapm_power_widgets() are (us): Min 4, Ave 16, Max 82, Count 39 Therefore, optimise the dapm_power_widgets() function by only scheduling the dapm_pre/post_sequence_async() work if the DAPM context is not in the desired state. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 461d951917c0..2e9231aeabb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1953,7 +1953,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); } @@ -1977,7 +1977,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run all the bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); } -- 2.7.4

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Alsa-devel August 2018