January 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound/pci/hda: depends on instead of select for INPUT
by Randy Dunlap 15 Jan '18

15 Jan '18

From: Randy Dunlap <rdunlap(a)infradead.org> Drivers should not 'select' a subsystem. Instead they should depend on it. If the subsystem is disabled, the user probably did that for a purpose and one driver shouldn't be changing that. This also makes all sound/ drivers consistent w.r.t depending on INPUT instead of selecting it. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org (moderated for non-subscribers) --- sound/pci/hda/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-415-rc8.orig/sound/pci/hda/Kconfig +++ lnx-415-rc8/sound/pci/hda/Kconfig @@ -87,8 +87,8 @@ config SND_HDA_PATCH_LOADER config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" + depends on INPUT select SND_HDA_GENERIC - select INPUT help Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880.

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[alsa-devel] [PATCH] ALSA: seq: Process queue tempo/ppq change in a shot
by Takashi Iwai 15 Jan '18

15 Jan '18

The SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO ioctl sets the tempo and the ppq in a single call, while the current implementation updates each value one by one. This is a bit racy, and also suboptimal from the performance POV, as each call does re-acquire the lock and invokes the update of ALSA timer resolution. This patch reorganizes the code slightly so that we change both the tempo and the ppq in a shot. The skew value can be put into the same lock, but this is rather a rarely used feature and completely independent from the temp/ppq (it's evaluated only in the interrupt), so it's left as it was. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.c | 4 +--- sound/core/seq/seq_timer.c | 13 ++++++++----- sound/core/seq/seq_timer.h | 2 +- 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 79e0c5604ef8..0428e9061b47 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -497,9 +497,7 @@ int snd_seq_queue_timer_set_tempo(int queueid, int client, return -EPERM; } - result = snd_seq_timer_set_tempo(q->timer, info->tempo); - if (result >= 0) - result = snd_seq_timer_set_ppq(q->timer, info->ppq); + result = snd_seq_timer_set_tempo_ppq(q->timer, info->tempo, info->ppq); if (result >= 0 && info->skew_base > 0) result = snd_seq_timer_set_skew(q->timer, info->skew_value, info->skew_base); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index b80985fbc334..23167578231f 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -191,14 +191,15 @@ int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo) return 0; } -/* set current ppq */ -int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq) +/* set current tempo and ppq in a shot */ +int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq) { + int changed; unsigned long flags; if (snd_BUG_ON(!tmr)) return -EINVAL; - if (ppq <= 0) + if (tempo <= 0 || ppq <= 0) return -EINVAL; spin_lock_irqsave(&tmr->lock, flags); if (tmr->running && (ppq != tmr->ppq)) { @@ -208,9 +209,11 @@ int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq) pr_debug("ALSA: seq: cannot change ppq of a running timer\n"); return -EBUSY; } - + changed = (tempo != tmr->tempo) || (ppq != tmr->ppq); + tmr->tempo = tempo; tmr->ppq = ppq; - snd_seq_timer_set_tick_resolution(tmr); + if (changed) + snd_seq_timer_set_tick_resolution(tmr); spin_unlock_irqrestore(&tmr->lock, flags); return 0; } diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h index 9506b661fe5b..62f390671096 100644 --- a/sound/core/seq/seq_timer.h +++ b/sound/core/seq/seq_timer.h @@ -131,7 +131,7 @@ int snd_seq_timer_stop(struct snd_seq_timer *tmr); int snd_seq_timer_start(struct snd_seq_timer *tmr); int snd_seq_timer_continue(struct snd_seq_timer *tmr); int snd_seq_timer_set_tempo(struct snd_seq_timer *tmr, int tempo); -int snd_seq_timer_set_ppq(struct snd_seq_timer *tmr, int ppq); +int snd_seq_timer_set_tempo_ppq(struct snd_seq_timer *tmr, int tempo, int ppq); int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr, snd_seq_tick_time_t position); int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr, snd_seq_real_time_t position); int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, unsigned int base); -- 2.15.1

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[alsa-devel] [PATCH 1/4] ASoC: codecs: tas5720: add basic support for TAS5722 devices
by Andrew F. Davis 15 Jan '18

15 Jan '18

From: Andreas Dannenberg <dannenberg(a)ti.com> The TI TAS5722 digital amplifier is very similar to the TAS5720 from an overall and register map perspective. Therefore the existing driver can be extended easily to support this additional device. This commit allows TAS5722 devices to be used in a "subset" type of fashion, without exposing any of the additional features they offer. Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- .../devicetree/bindings/sound/tas5720.txt | 4 ++- sound/soc/codecs/tas5720.c | 38 ++++++++++++++++++---- sound/soc/codecs/tas5720.h | 1 + 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt index 40d94f82beb3..7481653fe8e3 100644 --- a/Documentation/devicetree/bindings/sound/tas5720.txt +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -6,10 +6,12 @@ audio playback. For more product information please see the links below: http://www.ti.com/product/TAS5720L http://www.ti.com/product/TAS5720M +http://www.ti.com/product/TAS5722L Required properties: -- compatible : "ti,tas5720" +- compatible : "ti,tas5720", + "ti,tas5722" - reg : I2C slave address - dvdd-supply : phandle to a 3.3-V supply for the digital circuitry - pvdd-supply : phandle to a supply used for the Class-D amp and the analog diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index a736a2a6976c..5def54d1336d 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -36,6 +36,11 @@ /* Define how often to check (and clear) the fault status register (in ms) */ #define TAS5720_FAULT_CHECK_INTERVAL 200 +enum tas572x_type { + TAS5720, + TAS5722, +}; + static const char * const tas5720_supply_names[] = { "dvdd", /* Digital power supply. Connect to 3.3-V supply. */ "pvdd", /* Class-D amp and analog power supply (connected). */ @@ -47,6 +52,7 @@ struct tas5720_data { struct snd_soc_codec *codec; struct regmap *regmap; struct i2c_client *tas5720_client; + enum tas572x_type devtype; struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; struct delayed_work fault_check_work; unsigned int last_fault; @@ -264,7 +270,7 @@ static void tas5720_fault_check_work(struct work_struct *work) static int tas5720_codec_probe(struct snd_soc_codec *codec) { struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); - unsigned int device_id; + unsigned int device_id, expected_device_id; int ret; tas5720->codec = codec; @@ -276,6 +282,11 @@ static int tas5720_codec_probe(struct snd_soc_codec *codec) return ret; } + /* + * Take a liberal approach to checking the device ID to allow the + * driver to be used even if the device ID does not match, however + * issue a warning if there is a mismatch. + */ ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id); if (ret < 0) { dev_err(codec->dev, "failed to read device ID register: %d\n", @@ -283,13 +294,22 @@ static int tas5720_codec_probe(struct snd_soc_codec *codec) goto probe_fail; } - if (device_id != TAS5720_DEVICE_ID) { - dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n", - TAS5720_DEVICE_ID, device_id); - ret = -ENODEV; - goto probe_fail; + switch (tas5720->devtype) { + case TAS5720: + expected_device_id = TAS5720_DEVICE_ID; + break; + case TAS5722: + expected_device_id = TAS5722_DEVICE_ID; + break; + default: + dev_err(codec->dev, "unexpected private driver data\n"); + return -EINVAL; } + if (device_id != expected_device_id) + dev_warn(codec->dev, "wrong device ID. expected: %u read: %u\n", + expected_device_id, device_id); + /* Set device to mute */ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, TAS5720_MUTE, TAS5720_MUTE); @@ -552,6 +572,8 @@ static int tas5720_probe(struct i2c_client *client, return -ENOMEM; data->tas5720_client = client; + data->devtype = id->driver_data; + data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config); if (IS_ERR(data->regmap)) { ret = PTR_ERR(data->regmap); @@ -592,7 +614,8 @@ static int tas5720_remove(struct i2c_client *client) } static const struct i2c_device_id tas5720_id[] = { - { "tas5720", 0 }, + { "tas5720", TAS5720 }, + { "tas5722", TAS5722 }, { } }; MODULE_DEVICE_TABLE(i2c, tas5720_id); @@ -600,6 +623,7 @@ MODULE_DEVICE_TABLE(i2c, tas5720_id); #if IS_ENABLED(CONFIG_OF) static const struct of_device_id tas5720_of_match[] = { { .compatible = "ti,tas5720", }, + { .compatible = "ti,tas5722", }, { }, }; MODULE_DEVICE_TABLE(of, tas5720_of_match); diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h index 3d077c779b12..bef802afcc69 100644 --- a/sound/soc/codecs/tas5720.h +++ b/sound/soc/codecs/tas5720.h @@ -32,6 +32,7 @@ /* TAS5720_DEVICE_ID_REG */ #define TAS5720_DEVICE_ID 0x01 +#define TAS5722_DEVICE_ID 0x12 /* TAS5720_POWER_CTRL_REG */ #define TAS5720_DIG_CLIP_MASK GENMASK(7, 2) -- 2.15.0

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[alsa-devel] ALSA device with slave clock
by Jussi Laako 15 Jan '18

15 Jan '18

Hi, I've been fighting with some devices for the case where ALSA device is a clock slave. I'm also working on some patches/quirks to make some of these things work. I have couple of questions in this domain I didn't find documented: 1) How is application supposed to use ALSA capture device when it doesn't know the sampling rate? For example with ASIO on Windows application sets sampling rate to 0 to indicate slave and then starts the device and retrieves the rate using ASIOGetSampleRate(). There are typically also reset request messages when the rate changes. 2) Some USB devices have indicator for lock status between set sampling rate and detected input rate. One such device I have is Motu 8D (UAC2 device): Simple mixer control 'Internal Clock Validity',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] As long as the status is valid, I can set sampling rate to the same, or something else. When the status becomes invalid due to rate mismatch, ALSA fails to set any of the sampling rates, even correct one and there's an error message in dmesg: "usb 3-14: clock source 1 is not valid, cannot use". So the clock source cannot be made valid again by changing sampling rate once it has become invalid, because it is currently invalid... 3) What is the correct way to deal with clock slave device from both driver and application perspective? So that application can retrieve the current sampling rate, instead of setting one... UAC2 has provisions for this, but this doesn't seem to be supported by the current driver. I have also some hardware where I would like to do the same with I2S - where clock generator is outside of the SoC and the SoC is just a slave (for example S/PDIF or AES/EBU input to a SoC where clock is generated from the input stream using PLL). Best regards, - Jussi

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[alsa-devel] ASoC: How to write new (atmel) driver?
by Ladislav Michl 14 Jan '18

14 Jan '18

Hi there! I've got new at91sam9g20 board with max9867 codec (*), so I wanted to get inspiration from other drivers in sound/soc/atmel. Indeed, there are three of them: atmel_wm8904.c sam9g20_wm8731.c sam9x5_wm8731.c Now, looking at sound/soc/tegra, sound/soc/mediatek, etc. most drivers seems to be created by copy, modify (mostly find&replace), commit. There are so many of them, that it is probably impossible to fix them all (consider for example patch bellow to be done several times for different drivers) (**) So is sound/soc/generic preferred solution these days? Btw, generic drivers are calling of_node_put before they are done dealing with that node. And if above is preferred solution what about all those already created drivers? Fix them all or covert to generic one by introducing DT fixups for all those over the time created DT properties? At least for atmel drivers that seems to be doable as atmel_ssc_dai needs to be modified to get it work under generic framework and keeping old interface would needlesly complicate things - unless I miss something obvious of course :) (*) for now I wrote driver using copy&modify, but that does not seem to be the right solution anymore (**) almost two decades ago I rather wrote OSS driver for the very same reasons, but now OSS is gone... diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 9db08826b32a..11e2c37a2e42 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -78,7 +78,6 @@ static const struct snd_soc_dapm_widget sam9x5_dapm_widgets[] = { static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct device_node *codec_np, *cpu_np; struct snd_soc_card *card; struct snd_soc_dai_link *dai; struct sam9x5_drvdata *priv; @@ -122,23 +121,20 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) goto out; } - codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); - if (!codec_np) { + dai->codec_of_node = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!dai->codec_of_node) { dev_err(&pdev->dev, "atmel,audio-codec node missing\n"); ret = -EINVAL; goto out; } - dai->codec_of_node = codec_np; - - cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); - if (!cpu_np) { + dai->cpu_of_node = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!dai->cpu_of_node) { dev_err(&pdev->dev, "atmel,ssc-controller node missing\n"); ret = -EINVAL; - goto out; + goto out_put_codec; } - dai->cpu_of_node = cpu_np; - dai->platform_of_node = cpu_np; + dai->platform_of_node = dai->cpu_of_node; priv->ssc_id = of_alias_get_id(cpu_np, "ssc"); @@ -146,12 +142,9 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) if (ret != 0) { dev_err(&pdev->dev, "Failed to set SSC %d for audio: %d\n", ret, priv->ssc_id); - goto out; + goto out_put_cpu; } - of_node_put(codec_np); - of_node_put(cpu_np); - ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "Platform device allocation failed\n"); @@ -164,6 +157,10 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) out_put_audio: atmel_ssc_put_audio(priv->ssc_id); +out_put_cpu: + of_node_put(dai->cpu_of_node); +out_put_codec: + of_node_put(dai->codec_of_node); out: return ret; }

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Re: [alsa-devel] [PATCH] ASoC: sgtl5000: fix powerup/down of micbias-voltage
by Fischer, Christian 13 Jan '18

13 Jan '18

Hi Mark, > Has your board connected the bias to the microphone widget it defines? > Unless your board says that the bias is in use on a given board it won't be enabled. Thanks for this tip! The problem is that the machine-driver mxs-sgtl5000 don't support audio-routing. I've prepared a patch that will be submitted this noon. Regards, Christian

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[alsa-devel] [PATCH] ASoC: mxs-sgtl5000: add audio-routing support
by Christian Fischer 12 Jan '18

12 Jan '18

Add dapm_widgets to machine-driver (from imx-sgtl5000). If the "audio-routing"-property is present at probing the dapm-widgets getting linked to the card. Signed-off-by: Christian Fischer <fischerc(a)swissphone.de> --- .../bindings/sound/mxs-audio-sgtl5000.txt | 33 +++++++++++++++++++--- sound/soc/mxs/mxs-sgtl5000.c | 20 +++++++++++++ 2 files changed, 49 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt index 601c518..4eb980b 100644 --- a/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/mxs-audio-sgtl5000.txt @@ -1,10 +1,31 @@ * Freescale MXS audio complex with SGTL5000 codec Required properties: -- compatible: "fsl,mxs-audio-sgtl5000" -- model: The user-visible name of this sound complex -- saif-controllers: The phandle list of the MXS SAIF controller -- audio-codec: The phandle of the SGTL5000 audio codec +- compatible : "fsl,mxs-audio-sgtl5000" +- model : The user-visible name of this sound complex +- saif-controllers : The phandle list of the MXS SAIF controller +- audio-codec : The phandle of the SGTL5000 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, SGTL5000 + pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + SGTL5000 pins: + * MIC_IN + * LINE_IN + * HP_OUT + * LINE_OUT + + Board connectors: + * Mic Jack + * Line In Jack + * Headphone Jack + * Line Out Jack + * Ext Spk Example: @@ -14,4 +35,8 @@ sound { model = "imx28-evk-sgtl5000"; saif-controllers = <&saif0 &saif1>; audio-codec = <&sgtl5000>; + audio-routing = + "MIC_IN", "Mic Jack", + "Mic Jack", "Mic Bias", + "Headphone Jack", "HP_OUT"; }; diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 2ed3240..b593f76 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -93,6 +93,14 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { }, }; +static const struct snd_soc_dapm_widget mxs_sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Line Out Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + static struct snd_soc_card mxs_sgtl5000 = { .name = "mxs_sgtl5000", .owner = THIS_MODULE, @@ -141,6 +149,18 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) card->dev = &pdev->dev; + if (of_find_property(np, "audio-routing", NULL)) { + card->dapm_widgets = mxs_sgtl5000_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(mxs_sgtl5000_dapm_widgets); + + ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing (%d)\n", + ret); + return ret; + } + } + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", -- 2.7.4

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[alsa-devel] [PATCH v4] ASoC: Intel - Convert to use acpi_dev_get_first_match_name()
by Andy Shevchenko 12 Jan '18

12 Jan '18

Instead of home grown snd_soc_acpi_find_name_from_hid() use acpi_dev_get_first_match_name(). Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- - rebase on top of topic/intel branch of ASoC tree; The patch has dependency on Pierre's "ASoC: Intel: bytcht_es8316: fix HID handling" submitted earlier and pushed AFAICS to sound.git/topic/intel. include/sound/soc-acpi.h | 7 ------- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/soc-acpi.c | 33 --------------------------------- 8 files changed, 6 insertions(+), 46 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 83320af8def2..082224275f52 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -31,16 +31,9 @@ struct snd_soc_acpi_package_context { #define SND_ACPI_I2C_ID_LEN (4 + ACPI_ID_LEN + 3 + 1) #if IS_ENABLED(CONFIG_ACPI) -/* translation fron HID to I2C name, needed for DAI codec_name */ -const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], struct snd_soc_acpi_package_context *ctx); #else -static inline const char * -snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) -{ - return NULL; -} static inline bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN], struct snd_soc_acpi_package_context *ctx) diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 6219c04d4731..2179dedb28ad 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -243,7 +243,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 079f35cd4eaf..305e7f4fe55a 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -258,7 +258,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4548f75498d0..b6a1cfeec830 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -762,7 +762,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(byt_rt5640_codec_name, sizeof(byt_rt5640_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 5a6b7dedb773..456526a93dd5 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -539,7 +539,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(byt_rt5651_codec_name, sizeof(byt_rt5651_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index cef6a8c31c8d..976ea6bf9539 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -566,7 +566,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 1f3d38dc4fcb..c14a52d2f714 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -396,7 +396,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* fixup codec name based on HID */ if (mach) { - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(drv->codec_name, sizeof(drv->codec_name), "i2c-%s", i2c_name); diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 7f43c9bf3d09..3d7e1ff79139 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -16,39 +16,6 @@ #include <sound/soc-acpi.h> -static acpi_status snd_soc_acpi_find_name(acpi_handle handle, u32 level, - void *context, void **ret) -{ - struct acpi_device *adev; - const char *name = NULL; - - if (acpi_bus_get_device(handle, &adev)) - return AE_OK; - - if (adev->status.present && adev->status.functional) { - name = acpi_dev_name(adev); - *(const char **)ret = name; - return AE_CTRL_TERMINATE; - } - - return AE_OK; -} - -const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) -{ - const char *name = NULL; - acpi_status status; - - status = acpi_get_devices(hid, snd_soc_acpi_find_name, NULL, - (void **)&name); - - if (ACPI_FAILURE(status) || name[0] == '\0') - return NULL; - - return name; -} -EXPORT_SYMBOL_GPL(snd_soc_acpi_find_name_from_hid); - struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { -- 2.15.1

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[alsa-devel] [PATCH 0/3] ASoC: Intel: ACPI-related fixes
by Pierre-Louis Bossart 12 Jan '18

12 Jan '18

Here are 3 patches that were provided earlier on the mailing list or in Bugzilla entries, I rebased and ported them on top of the latest ACPI/KConfig fixes. Jeremy Cline (1): ASoC: Replace snd_soc_acpi_check_hid with acpi_dev_present Pierre-Louis Bossart (2): ASoC: acpi: fix machine driver selection based on quirk ASoC: Intel: bytcht_es8316: fix HID handling include/sound/soc-acpi.h | 3 --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bytcht_es8316.c | 26 ++++++++++++++++++++++- sound/soc/soc-acpi.c | 38 ++++------------------------------ 4 files changed, 30 insertions(+), 38 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH v3 0/2] ASoC: Intel: ACPI-related fixes
by Pierre-Louis Bossart 12 Jan '18

12 Jan '18

The first two patches in the v2 series were merged, but these last two were not. As agreed with Andy and Vinod, there will be a follow-up patch to replace snd_soc_acpi_find_name_from_hid() with acpi_dev_get_dev_name(), as well as the introduction of a helper for the search loops. Changes since v2: rename constant per Vinod's suggestion removed redundant comment on name Changes since v1: cleaned-up quirk handling, fixed merge mistake added Andy's Reviewed-by tag on patch 2 added 4th patch to add new constant dependent on ACPI_ID_LEN Pierre-Louis Bossart (2): ASoC: Intel: bytcht_es8316: fix HID handling ASoC: acpi: remove hard-coded i2c-device name length include/sound/soc-acpi.h | 3 +++ sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 26 +++++++++++++++++++++++++- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- 8 files changed, 35 insertions(+), 7 deletions(-) -- 2.14.1

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