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

[alsa-devel] Applied "ASoC: support ROHM BD28623 codec" to the asoc tree
by Mark Brown 28 Feb '18

28 Feb '18

The patch ASoC: support ROHM BD28623 codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f6d4b052175d84c6f2b73e884f3ab8c83c5f4d45 Mon Sep 17 00:00:00 2001 From: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> Date: Fri, 23 Feb 2018 10:17:17 +0900 Subject: [PATCH] ASoC: support ROHM BD28623 codec This patch adds support of the ROHM BD28623MUV Class D speaker amplifier for Flat-panel TVs. This IC delivers an output power of 20W + 20W. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 8 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/bd28623.c | 242 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 252 insertions(+) create mode 100644 sound/soc/codecs/bd28623.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2b331f7266ab..4545f8fde220 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -45,6 +45,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_BT_SCO + select SND_SOC_BD28623 select SND_SOC_CQ0093VC select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C @@ -405,6 +406,13 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate +config SND_SOC_BD28623 + tristate "ROHM BD28623 CODEC" + help + Enable support for ROHM BD28623MUV Class D speaker amplifier. + This codec does not have any control buses such as I2C, it + detect format of I2S automatically. + config SND_SOC_BT_SCO tristate "Dummy BT SCO codec driver" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index da1571336f1e..6df2fb32fc7e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -35,6 +35,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o +snd-soc-bd28623-objs := bd28623.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o @@ -280,6 +281,7 @@ obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o +obj-$(CONFIG_SND_SOC_BD28623) += snd-soc-bd28623.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o diff --git a/sound/soc/codecs/bd28623.c b/sound/soc/codecs/bd28623.c new file mode 100644 index 000000000000..31904ef5c88b --- /dev/null +++ b/sound/soc/codecs/bd28623.c @@ -0,0 +1,242 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ROHM BD28623MUV class D speaker amplifier codec driver. +// +// Copyright (c) 2018 Socionext Inc. + +#include <linux/delay.h> +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#define BD28623_NUM_SUPPLIES 3 + +static const char *const bd28623_supply_names[BD28623_NUM_SUPPLIES] = { + "VCCA", + "VCCP1", + "VCCP2", +}; + +struct bd28623_priv { + struct device *dev; + struct regulator_bulk_data supplies[BD28623_NUM_SUPPLIES]; + struct gpio_desc *reset_gpio; + struct gpio_desc *mute_gpio; + + int switch_spk; +}; + +static const struct snd_soc_dapm_widget bd28623_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT1P"), + SND_SOC_DAPM_OUTPUT("OUT1N"), + SND_SOC_DAPM_OUTPUT("OUT2P"), + SND_SOC_DAPM_OUTPUT("OUT2N"), +}; + +static const struct snd_soc_dapm_route bd28623_routes[] = { + { "OUT1P", NULL, "DAC" }, + { "OUT1N", NULL, "DAC" }, + { "OUT2P", NULL, "DAC" }, + { "OUT2N", NULL, "DAC" }, +}; + +static int bd28623_power_on(struct bd28623_priv *bd) +{ + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(bd->supplies), bd->supplies); + if (ret) { + dev_err(bd->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + gpiod_set_value_cansleep(bd->reset_gpio, 0); + usleep_range(300000, 400000); + + return 0; +} + +static void bd28623_power_off(struct bd28623_priv *bd) +{ + gpiod_set_value_cansleep(bd->reset_gpio, 1); + + regulator_bulk_disable(ARRAY_SIZE(bd->supplies), bd->supplies); +} + +static int bd28623_get_switch_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = bd->switch_spk; + + return 0; +} + +static int bd28623_set_switch_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + if (bd->switch_spk == ucontrol->value.integer.value[0]) + return 0; + + bd->switch_spk = ucontrol->value.integer.value[0]; + + gpiod_set_value_cansleep(bd->mute_gpio, bd->switch_spk ? 0 : 1); + + return 0; +} + +static const struct snd_kcontrol_new bd28623_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Switch", 0, + bd28623_get_switch_spk, bd28623_set_switch_spk), +}; + +static int bd28623_codec_probe(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + int ret; + + bd->switch_spk = 1; + + ret = bd28623_power_on(bd); + if (ret) + return ret; + + gpiod_set_value_cansleep(bd->mute_gpio, bd->switch_spk ? 0 : 1); + + return 0; +} + +static void bd28623_codec_remove(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + bd28623_power_off(bd); +} + +static int bd28623_codec_suspend(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + + bd28623_power_off(bd); + + return 0; +} + +static int bd28623_codec_resume(struct snd_soc_component *component) +{ + struct bd28623_priv *bd = snd_soc_component_get_drvdata(component); + int ret; + + ret = bd28623_power_on(bd); + if (ret) + return ret; + + gpiod_set_value_cansleep(bd->mute_gpio, bd->switch_spk ? 0 : 1); + + return 0; +} + +static const struct snd_soc_component_driver soc_codec_bd = { + .probe = bd28623_codec_probe, + .remove = bd28623_codec_remove, + .suspend = bd28623_codec_suspend, + .resume = bd28623_codec_resume, + .dapm_widgets = bd28623_widgets, + .num_dapm_widgets = ARRAY_SIZE(bd28623_widgets), + .dapm_routes = bd28623_routes, + .num_dapm_routes = ARRAY_SIZE(bd28623_routes), + .controls = bd28623_controls, + .num_controls = ARRAY_SIZE(bd28623_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct snd_soc_dai_driver soc_dai_bd = { + .name = "bd28623-speaker", + .playback = { + .stream_name = "Playback", + .formats = SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_32000, + .channels_min = 2, + .channels_max = 2, + }, +}; + +static int bd28623_probe(struct platform_device *pdev) +{ + struct bd28623_priv *bd; + struct device *dev = &pdev->dev; + int i, ret; + + bd = devm_kzalloc(&pdev->dev, sizeof(struct bd28623_priv), GFP_KERNEL); + if (!bd) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(bd->supplies); i++) + bd->supplies[i].supply = bd28623_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(bd->supplies), + bd->supplies); + if (ret) { + dev_err(dev, "Failed to get supplies: %d\n", ret); + return ret; + } + + bd->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(bd->reset_gpio)) { + dev_err(dev, "Failed to request reset_gpio: %ld\n", + PTR_ERR(bd->reset_gpio)); + return PTR_ERR(bd->reset_gpio); + } + + bd->mute_gpio = devm_gpiod_get_optional(dev, "mute", + GPIOD_OUT_HIGH); + if (IS_ERR(bd->mute_gpio)) { + dev_err(dev, "Failed to request mute_gpio: %ld\n", + PTR_ERR(bd->mute_gpio)); + return PTR_ERR(bd->mute_gpio); + } + + platform_set_drvdata(pdev, bd); + bd->dev = dev; + + return devm_snd_soc_register_component(dev, &soc_codec_bd, + &soc_dai_bd, 1); +} + +static const struct of_device_id bd28623_of_match[] = { + { .compatible = "rohm,bd28623", }, + {} +}; +MODULE_DEVICE_TABLE(of, bd28623_of_match); + +static struct platform_driver bd28623_codec_driver = { + .driver = { + .name = "bd28623", + .of_match_table = of_match_ptr(bd28623_of_match), + }, + .probe = bd28623_probe, +}; +module_platform_driver(bd28623_codec_driver); + +MODULE_AUTHOR("Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com>"); +MODULE_DESCRIPTION("ROHM BD28623 speaker amplifier driver"); +MODULE_LICENSE("GPL v2"); -- 2.16.1

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[alsa-devel] [PATCH] ASoC: wm_adsp: For TLV controls only register TLV get/set
by Richard Fitzgerald 28 Feb '18

28 Feb '18

Normal 512-byte get/set of a TLV isn't supported but we were registering the normal get/set anyway and relying on omitting the SNDRV_CTL_ELEM_ACCESS_[READ|WRITE] flags to prevent them being called. Trouble is if this gets broken in the core ALSA code - as it has been since at least 4.14 - the standard get/set can be called unexpectedly and corrupt memory. There's no point providing functions that won't be called and it's a trivial change. The benefit is that if the ALSA core gets broken again we get a big fat immediate NULL dereference instead of a memory corruption timebomb. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9b27390cf4f1..82b0927e6ed7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1204,12 +1204,14 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->put = wm_coeff_put_acked; break; default: - kcontrol->get = wm_coeff_get; - kcontrol->put = wm_coeff_put; - - ctl->bytes_ext.max = ctl->len; - ctl->bytes_ext.get = wm_coeff_tlv_get; - ctl->bytes_ext.put = wm_coeff_tlv_put; + if (kcontrol->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + } else { + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + } break; } -- 2.11.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: x86: Fix missing spinlock and mutex initializations
by Takashi Iwai 28 Feb '18

28 Feb '18

The commit change for supporting the multiple ports moved involved some code shuffling, and there the initializations of spinlock and mutex in snd_intelhad object were dropped mistakenly. This patch adds the missing initializations again for each port. Fixes: b4eb0d522fcb ("ALSA: x86: Split snd_intelhad into card and PCM specific structures") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/intel_hdmi_audio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 96115c401292..cec62f9b7085 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1835,6 +1835,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) ctx->port = single_port ? -1 : port; ctx->pipe = -1; + spin_lock_init(&ctx->had_spinlock); + mutex_init(&ctx->mutex); INIT_WORK(&ctx->hdmi_audio_wq, had_audio_wq); ret = snd_pcm_new(card, INTEL_HAD, port, MAX_PB_STREAMS, -- 2.16.2

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[alsa-devel] [PATCH] ALSA: control: Fix a bunch of whitespace errors
by Richard Fitzgerald 28 Feb '18

28 Feb '18

Remove a bunch of trailing whitespace errors. They are fairly annoying if you have your editor set to strip trailing whitespace because you find you've introduced more changes than you were trying to make. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/core/control.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 8a77620a3854..69734b0eafd0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -105,7 +105,7 @@ static void snd_ctl_empty_read_queue(struct snd_ctl_file * ctl) { unsigned long flags; struct snd_kctl_event *cread; - + spin_lock_irqsave(&ctl->read_lock, flags); while (!list_empty(&ctl->events)) { cread = snd_kctl_event(ctl->events.next); @@ -159,7 +159,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, unsigned long flags; struct snd_ctl_file *ctl; struct snd_kctl_event *ev; - + if (snd_BUG_ON(!card || !id)) return; if (card->shutdown) @@ -213,7 +213,7 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, { unsigned int size; unsigned int idx; - + if (count == 0 || count > MAX_CONTROL_COUNT) return -EINVAL; @@ -238,7 +238,7 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, * @ncontrol: the initialization record * @private_data: the private data to set * - * Allocates a new struct snd_kcontrol instance and initialize from the given + * Allocates a new struct snd_kcontrol instance and initialize from the given * template. When the access field of ncontrol is 0, it's assumed as * READWRITE access. When the count field is 0, it's assumes as one. * @@ -251,7 +251,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, unsigned int count; unsigned int access; int err; - + if (snd_BUG_ON(!ncontrol || !ncontrol->info)) return NULL; @@ -753,7 +753,7 @@ static int snd_ctl_elem_list(struct snd_card *card, struct snd_ctl_elem_id id; unsigned int offset, space, jidx; int err = 0; - + if (copy_from_user(&list, _list, sizeof(list))) return -EFAULT; offset = list.offset; @@ -827,7 +827,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, struct snd_kcontrol_volatile *vd; unsigned int index_offset; int result; - + down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &info->id); if (kctl == NULL) { @@ -992,7 +992,7 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; int result; - + if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); @@ -1020,7 +1020,7 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; int result; - + if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); -- 2.11.0

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[alsa-devel] [PATCH v1 0/9] Enable HDA Codec support on Intel Platforms (Series2)
by Rakesh Ughreja 27 Feb '18

27 Feb '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. This patch carry forwards the work that is done in the previous series to enable HD Audio codecs on such platforms. This patch series adds ASoC HDA codec driver for Intel platforms. It is written by reusing the legacy HDA ALSA codec driver. Intention is to maximize the reuse and minimize the changes in the legacy HDA codec driver. Rakesh Ughreja (9): ASoC: Intel: Boards: Machine driver for Intel platforms ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda - split snd_hda_codec_new function ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init ALSA: hdac: add extended ops in the hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers ASoC: Intel: Boards: add support for HDA codecs include/sound/hdaudio.h | 9 + include/sound/hdaudio_ext.h | 6 +- sound/hda/ext/hdac_ext_bus.c | 12 +- sound/pci/hda/hda_bind.c | 6 + sound/pci/hda/hda_codec.c | 67 +++- sound/pci/hda/hda_codec.h | 2 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 448 +++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 23 ++ sound/soc/intel/Kconfig | 1 + sound/soc/intel/boards/Kconfig | 10 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 131 ++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 34 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 143 +++++++++ sound/soc/intel/skylake/skl-pcm.c | 50 ++- sound/soc/intel/skylake/skl.c | 98 +++++- sound/soc/intel/skylake/skl.h | 11 +- 19 files changed, 1011 insertions(+), 49 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c -- 2.7.4

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[alsa-devel] [PATCH] ALSA: control: Fix memory corruption risk in snd_ctl_elem_read
by Richard Fitzgerald 27 Feb '18

27 Feb '18

The patch "ALSA: control: code refactoring for ELEM_READ/ELEM_WRITE operations" introduced a potential for kernel memory corruption due to an incorrect if statement allowing non-readable controls to fall through and call the get function. For TLV controls a driver can omit SNDRV_CTL_ELEM_ACCESS_READ to ensure that only the TLV get function can be called. Instead the normal get() can be invoked unexpectedly and as the driver expects that this will only be called for controls <= 512 bytes, potentially try to copy >512 bytes into the 512 byte return array, so corrupting kernel memory. The problem is an attempt to refactor the snd_ctl_elem_read function to invert the logic so that it conditionally aborted if the control is unreadable instead of conditionally executing. But the if statement wasn't inverted correctly. The correct inversion of if (a && !b) is if (!a || b) Fixes: becf9e5d553c2 ("ALSA: control: code refactoring for ELEM_READ/ELEM_WRITE operations") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 0b3026d937b1..8a77620a3854 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -889,7 +889,7 @@ static int snd_ctl_elem_read(struct snd_card *card, index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) && kctl->get == NULL) + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL) return -EPERM; snd_ctl_build_ioff(&control->id, kctl, index_offset); -- 2.11.0

2 1

[alsa-devel] aplay\arecord compare utility
by Muni Sekhar 27 Feb '18

27 Feb '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, I’m using an alsa utilities(aplay & arecord) for sound loopback testing. From Linux host , sending & receiving the wav file. At end I want to compare the sent & received wav file. Does Linux has any utility to support this one? -- Thanks, Sekhar

2 3

[alsa-devel] [PATCH] ALSA: Use scnprintf() instead of snprintf()
by Jaejoong Kim 27 Feb '18

27 Feb '18

The show() method should use scnprintf() not snprintf() because snprintf() may returns a value that exceeds its second argument. Signed-off-by: Jaejoong Kim <climbbb.kim(a)gmail.com> --- sound/core/init.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 168ae03..4a51a06 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -670,7 +670,7 @@ card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%s\n", card->id); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->id); } static ssize_t @@ -710,7 +710,7 @@ card_number_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%i\n", card->number); + return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); -- 2.7.4

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[alsa-devel] No sound on MacbookPro 13,3 (A1707)
by Freaky 27 Feb '18

27 Feb '18

Hi there, we're trying to get sound working on a MacbookPro 13,3 (A1707). So far I've been unable to get the sound working. Did use the HDA analyzer and flipped the out-dir, enable and data on all 8 GPIO's, but none of them got the sound working on either the onboard speakers nor the jack output for headphone. Is there anyone willing to help on this issue? Driver development is a bit beyond my skills. I can provide hands, debug logs, etc. tho' :). Currently trying with a default Ubuntu 17.10.1 live USB, but a 4.15.3 kernel didn't help either. Alsa-info has been uploaded to http://www.alsa-project.org/db/?f=33cc432d3b648d1aa6adc997630817046656a086 The system was booted with an external USB-C dock from Dell, otherwise there's no network and no regular (USB-A) ports available (booting from stick) as the WiFi doesn't work and the device has only USB-C ports. The card 0 is from the dock - it's not relevant (and working). Before alsa-info ran there was also a USB headphone plugged in, that was card 1. Without those devices card 2 = 0 and 3 = 1 at boot. Issues are with what's listed as card 2 currently, the HDA-Intel - HDA Intel PCH. Any help is greatly appreciated. Some people already have some trackers for the current state of the devices. Not sure if they are of use, but might help: https://github.com/Dunedan/mbp-2016-linux/ https://gist.github.com/roadrunner2/1289542a748d9a104e7baec6a92f9cd7 Kind regards,

2 1

[alsa-devel] [PATCH v3] ASoC: imx-wm8962: Remove machine driver
by Fabio Estevam 27 Feb '18

27 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> fsl-asoc-card machine driver also handles wm8962, so there is really no need for keeping the dedicated imx-wm8962 driver anymore. Remove the imx-wm8962 machine driver. Suggested-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v2: - Also remove the dt-bindings .../devicetree/bindings/sound/imx-audio-wm8962.txt | 53 ---- sound/soc/fsl/Kconfig | 11 - sound/soc/fsl/Makefile | 2 - sound/soc/fsl/imx-wm8962.c | 312 --------------------- 4 files changed, 378 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt delete mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt deleted file mode 100644 index acea71b..0000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt +++ /dev/null @@ -1,53 +0,0 @@ -Freescale i.MX audio complex with WM8962 codec - -Required properties: - - - compatible : "fsl,imx-audio-wm8962" - - - model : The user-visible name of this sound complex - - - ssi-controller : The phandle of the i.MX SSI controller - - - audio-codec : The phandle of the WM8962 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, WM8962 - pins, and the jacks on the board: - - Power supplies: - * Mic Bias - - Board connectors: - * Mic Jack - * Headphone Jack - * Ext Spk - - - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) - - - mux-ext-port : The external port of the i.MX audio muxer - -Note: The AUDMUX port numbering should start at 1, which is consistent with -hardware manual. - -Example: - -sound { - compatible = "fsl,imx6q-sabresd-wm8962", - "fsl,imx-audio-wm8962"; - model = "wm8962-audio"; - ssi-controller = <&ssi2>; - audio-codec = <&codec>; - audio-routing = - "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "Ext Spk", "SPKOUTL", - "Ext Spk", "SPKOUTR", - "MICBIAS", "AMIC", - "IN3R", "MICBIAS", - "DMIC", "MICBIAS", - "DMICDAT", "DMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; -}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37f9b62..6ec19fb 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -231,17 +231,6 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface -config SND_SOC_IMX_WM8962 - tristate "SoC Audio support for i.MX boards with wm8962" - depends on OF && I2C && INPUT - select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI - help - Say Y if you want to add support for SoC audio on an i.MX board with - a wm8962 codec. - config SND_SOC_IMX_ES8328 tristate "SoC Audio support for i.MX boards with the ES8328 codec" depends on OF && (I2C || SPI) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c67bf11..de94fa0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -55,7 +55,6 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o -snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,6 +64,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o -obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c deleted file mode 100644 index 206b898..0000000 --- a/sound/soc/fsl/imx-wm8962.c +++ /dev/null @@ -1,312 +0,0 @@ -/* - * Copyright 2013 Freescale Semiconductor, Inc. - * - * Based on imx-sgtl5000.c - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ - -#include <linux/module.h> -#include <linux/of_platform.h> -#include <linux/i2c.h> -#include <linux/slab.h> -#include <linux/clk.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/soc-dapm.h> -#include <linux/pinctrl/consumer.h> - -#include "../codecs/wm8962.h" -#include "imx-audmux.h" - -#define DAI_NAME_SIZE 32 - -struct imx_wm8962_data { - struct snd_soc_dai_link dai; - struct snd_soc_card card; - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[DAI_NAME_SIZE]; - unsigned int clk_frequency; -}; - -struct imx_priv { - struct platform_device *pdev; - int sample_rate; - snd_pcm_format_t sample_format; -}; - -static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static int imx_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); - - priv->sample_rate = params_rate(params); - priv->sample_format = params_format(params); - - return 0; -} - -static const struct snd_soc_ops imx_hifi_ops = { - .hw_params = imx_hifi_hw_params, -}; - -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int imx_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(dev, "failed to set sysclk in %s\n", __func__); - - return ret; -} - -static int imx_wm8962_probe(struct platform_device *pdev) -{ - struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; - struct platform_device *ssi_pdev; - struct i2c_client *codec_dev; - struct imx_wm8962_data *data; - struct imx_priv *priv; - struct clk *codec_clk; - int int_port, ext_port; - int ret; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->pdev = pdev; - priv->sample_rate = 44100; - priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; - - ret = of_property_read_u32(np, "mux-int-port", &int_port); - if (ret) { - dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); - return ret; - } - ret = of_property_read_u32(np, "mux-ext-port", &ext_port); - if (ret) { - dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); - return ret; - } - - /* - * The port numbering in the hardware manual starts at 1, while - * the audmux API expects it starts at 0. - */ - int_port--; - ext_port--; - ret = imx_audmux_v2_configure_port(int_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TCLKDIR, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(&pdev->dev, "audmux internal port setup failed\n"); - return ret; - } - ret = imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(&pdev->dev, "audmux external port setup failed\n"); - return ret; - } - - ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); - codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); - if (!ssi_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); - ret = -EINVAL; - goto fail; - } - - ssi_pdev = of_find_device_by_node(ssi_np); - if (!ssi_pdev) { - dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; - goto fail; - } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev || !codec_dev->dev.driver) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } - - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - - codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(codec_clk)) { - ret = PTR_ERR(codec_clk); - dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); - goto fail; - } - - data->clk_frequency = clk_get_rate(codec_clk); - clk_put(codec_clk); - - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); - if (ret) - goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); - if (ret) - goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; - - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); - - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; - } - -fail: - of_node_put(ssi_np); - of_node_put(codec_np); - - return ret; -} - -static const struct of_device_id imx_wm8962_dt_ids[] = { - { .compatible = "fsl,imx-audio-wm8962", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); - -static struct platform_driver imx_wm8962_driver = { - .driver = { - .name = "imx-wm8962", - .pm = &snd_soc_pm_ops, - .of_match_table = imx_wm8962_dt_ids, - }, - .probe = imx_wm8962_probe, -}; -module_platform_driver(imx_wm8962_driver); - -MODULE_AUTHOR("Freescale Semiconductor, Inc."); -MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:imx-wm8962"); -- 2.7.4

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