October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/2] ASoC: meson: axg: fix TDM channel order sync
by Jerome Brunet 23 Oct '21

23 Oct '21

On the Amlogic AXG series, the TODDR FIFO may get out of sync with the TDM decoder if the decoder is started before the FIFO. The channel appears shifted in memory in an unpredictable way. To fix this, the trick is to start the FIFO before the TDM decoder. This way the FIFO is already waiting when the 1st channel is produced and it is correctly placed in memory. Jerome Brunet (2): ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++++++----- 2 files changed, 22 insertions(+), 5 deletions(-) -- 2.33.0

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[PATCH] ASoC: rt5682s: Downsizing the DAC volume scale
by derek.fang＠realtek.com 23 Oct '21

23 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> Use 0.75db/step of DAC volume instead of 1.5 to get a more smooth volume curve. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 9dd1796f3339..f6435b233a51 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1003,7 +1003,7 @@ static int rt5682s_set_jack_detect(struct snd_soc_component *component, return 0; } -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9562, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); @@ -1011,7 +1011,7 @@ static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); static const struct snd_kcontrol_new rt5682s_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682S_DAC1_DIG_VOL, - RT5682S_L_VOL_SFT + 2, RT5682S_R_VOL_SFT + 2, 63, 0, dac_vol_tlv), + RT5682S_L_VOL_SFT + 1, RT5682S_R_VOL_SFT + 1, 127, 0, dac_vol_tlv), /* CBJ Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682S_REC_MIXER, -- 2.17.1

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[PATCH] ASoC: cs42l42: Remove unused runtime_suspend/runtime_resume callbacks
by Richard Fitzgerald 23 Oct '21

23 Oct '21

The driver has runtime_suspend and runtime_resume callbacks, but pm_runtime is never enabled so these functions won't be called. They could not be used anyway because the runtime_suspend would cause jack detect to stop working. These functions are unused - delete them. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 51 +--------------------------------------------- 1 file changed, 1 insertion(+), 50 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 5e4d6791756b..0dbe4e23194b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -25,7 +25,6 @@ #include <linux/regulator/consumer.h> #include <linux/gpio/consumer.h> #include <linux/of_device.h> -#include <linux/pm_runtime.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2175,59 +2174,12 @@ static int cs42l42_i2c_remove(struct i2c_client *i2c_client) if (i2c_client->irq) devm_free_irq(&i2c_client->dev, i2c_client->irq, cs42l42); - pm_runtime_suspend(&i2c_client->dev); - pm_runtime_disable(&i2c_client->dev); - - return 0; -} - -#ifdef CONFIG_PM -static int cs42l42_runtime_suspend(struct device *dev) -{ - struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); - - regcache_cache_only(cs42l42->regmap, true); - regcache_mark_dirty(cs42l42->regmap); - - /* Hold down reset */ gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); - - /* remove power */ - regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), - cs42l42->supplies); + regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), cs42l42->supplies); return 0; } -static int cs42l42_runtime_resume(struct device *dev) -{ - struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); - int ret; - - /* Enable power */ - ret = regulator_bulk_enable(ARRAY_SIZE(cs42l42->supplies), - cs42l42->supplies); - if (ret != 0) { - dev_err(dev, "Failed to enable supplies: %d\n", - ret); - return ret; - } - - gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); - usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); - - regcache_cache_only(cs42l42->regmap, false); - regcache_sync(cs42l42->regmap); - - return 0; -} -#endif - -static const struct dev_pm_ops cs42l42_runtime_pm = { - SET_RUNTIME_PM_OPS(cs42l42_runtime_suspend, cs42l42_runtime_resume, - NULL) -}; - #ifdef CONFIG_OF static const struct of_device_id cs42l42_of_match[] = { { .compatible = "cirrus,cs42l42", }, @@ -2254,7 +2206,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l42_id); static struct i2c_driver cs42l42_i2c_driver = { .driver = { .name = "cs42l42", - .pm = &cs42l42_runtime_pm, .of_match_table = of_match_ptr(cs42l42_of_match), .acpi_match_table = ACPI_PTR(cs42l42_acpi_match), }, -- 2.11.0

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[PATCH] ASoC: fix unmet dependency on GPIOLIB
by Julian Braha 23 Oct '21

23 Oct '21

When SND_SOC_SC7180 or SND_SOC_STORM is selected, and GPIOLIB is not selected, Kbuild gives the following warning: WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_STORM [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] - SND_SOC_SC7180 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && I2C [=y] This is because SND_SOC_MAX98357A is selected by SND_SOC_STORM and SND_SOC_SC7180, but these config options do not select or depend on GPIOLIB, despite SND_SOC_MAX98357A depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/qcom/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index cc7c1de2f1d9..8c5c1ed54bc7 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -38,6 +38,7 @@ config SND_SOC_LPASS_SC7180 config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" + depends on GPIOLIB select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -141,7 +142,7 @@ config SND_SOC_SM8250 config SND_SOC_SC7180 tristate "SoC Machine driver for SC7180 boards" - depends on I2C + depends on I2C && GPIOLIB select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A -- 2.25.1

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wrong test in alsa-info.sh
by GitHub issues - opened 22 Oct '21

22 Oct '21

alsa-project/alsa-utils issue #115 was opened from mhampl: In line 691 of alsa_info.sh script (version 0.5.0) a test for $TEMPDIR/alsa-usbstream.tmp is needed instead of $TEMPDIR/lsusb.tmp Issue URL : https://github.com/alsa-project/alsa-utils/issues/115 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: cs42l51: improve error handling in cs42l51_remove()
by Uwe Kleine-König 22 Oct '21

22 Oct '21

When disabling a regulator fails while the device goes away, it doesn't make sense to reenable the already disabled regulators, so use regulator_bulk_force_disable() instead of regulator_bulk_disable(). In the error case returning an error code is also hardly sensible because the only effect is that the i2c core emits a generic error message. Replace this by a more specific message and make sure the i2c remove callback returns 0 instead. While at it also change cs42l51_remove() to return void as it doesn't yield any meaningful return value any more. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, a bit unsure if regulator_bulk_force_disable() is indeed the right function here, its documentation specifies a different usecase. My motivation for this change was to make it obvious in cs42l51_i2c_remove() that there is no error to handle to eventually make i2c remove callbacks return void, too. Best regards Uwe sound/soc/codecs/cs42l51-i2c.c | 4 +++- sound/soc/codecs/cs42l51.c | 11 ++++++++--- sound/soc/codecs/cs42l51.h | 2 +- 3 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 70260e0a8f09..3cb21a2ba29f 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -31,7 +31,9 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c, static int cs42l51_i2c_remove(struct i2c_client *i2c) { - return cs42l51_remove(&i2c->dev); + cs42l51_remove(&i2c->dev); + + return 0; } static const struct dev_pm_ops cs42l51_pm_ops = { diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..f3667693e2ea 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -793,14 +793,19 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(cs42l51_probe); -int cs42l51_remove(struct device *dev) +void cs42l51_remove(struct device *dev) { struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + int ret; gpiod_set_value_cansleep(cs42l51->reset_gpio, 1); - return regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), - cs42l51->supplies); + ret = regulator_bulk_force_disable(ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); + if (ret) + dev_warn(dev, "Failed to disable all regulators (%pe)\n", + ERR_PTR(ret)); + } EXPORT_SYMBOL_GPL(cs42l51_remove); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 9d06cf7f8876..a79343e8a54e 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -13,7 +13,7 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); -int cs42l51_remove(struct device *dev); +void cs42l51_remove(struct device *dev); int __maybe_unused cs42l51_suspend(struct device *dev); int __maybe_unused cs42l51_resume(struct device *dev); extern const struct of_device_id cs42l51_of_match[]; -- 2.30.2

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[PATCH] ASoC: Intel: Unify HDAudio-ext bus initialization
by Cezary Rojewski 21 Oct '21

21 Oct '21

HDAudio-extended bus initialization parts are scattered throughout Intel ADSP drivers code. Gather them up in snd_hda_ext_bus_init() to provide unified initialization point. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/hdaudio_ext.h | 9 ++++++--- sound/hda/ext/hdac_ext_bus.c | 29 ++++++++++++++++++++--------- sound/soc/intel/skylake/skl.c | 9 +-------- sound/soc/sof/intel/hda-bus.c | 16 +++++++++------- sound/soc/sof/intel/hda.c | 5 +++-- sound/soc/sof/intel/hda.h | 3 ++- 6 files changed, 41 insertions(+), 30 deletions(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 375581634143..d1f6e9f7c057 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -2,11 +2,14 @@ #ifndef __SOUND_HDAUDIO_EXT_H #define __SOUND_HDAUDIO_EXT_H +#include <linux/pci.h> #include <sound/hdaudio.h> +#include <sound/hda_codec.h> -int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_bus_ops *ops, - const struct hdac_ext_bus_ops *ext_ops); +int snd_hda_ext_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const struct hdac_bus_ops *ops, + const struct hdac_ext_bus_ops *ext_ops, + const char *model); void snd_hdac_ext_bus_exit(struct hdac_bus *bus); int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 765c40a6ccba..a89e2e80ea4c 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -10,15 +10,17 @@ */ #include <linux/module.h> +#include <linux/pci.h> #include <linux/slab.h> #include <linux/io.h> #include <sound/hdaudio_ext.h> +#include <sound/hda_codec.h> MODULE_DESCRIPTION("HDA extended core"); MODULE_LICENSE("GPL v2"); /** - * snd_hdac_ext_bus_init - initialize a HD-audio extended bus + * snd_hda_ext_bus_init - initialize a HD-audio extended bus * @bus: the pointer to HDAC bus object * @dev: device pointer * @ops: bus verb operators @@ -26,28 +28,37 @@ MODULE_LICENSE("GPL v2"); * * Returns 0 if successful, or a negative error code. */ -int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_bus_ops *ops, - const struct hdac_ext_bus_ops *ext_ops) +int snd_hda_ext_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const struct hdac_bus_ops *ops, + const struct hdac_ext_bus_ops *ext_ops, + const char *model) { + struct hdac_bus *base = &bus->core; int ret; - ret = snd_hdac_bus_init(bus, dev, ops); + ret = snd_hdac_bus_init(base, &pdev->dev, ops); if (ret < 0) return ret; - bus->ext_ops = ext_ops; + base->ext_ops = ext_ops; /* FIXME: * Currently only one bus is supported, if there is device with more * buses, bus->idx should be greater than 0, but there needs to be a * reliable way to always assign same number. */ - bus->idx = 0; - bus->cmd_dma_state = true; + base->idx = 0; + base->cmd_dma_state = true; + base->use_posbuf = 1; + base->bdl_pos_adj = 0; + base->sync_write = 1; + bus->pci = pdev; + bus->modelname = model; + bus->mixer_assigned = -1; + mutex_init(&bus->prepare_mutex); return 0; } -EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); +EXPORT_SYMBOL_GPL(snd_hda_ext_bus_init); /** * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 5b1a15e39912..95de41d14e56 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -886,16 +886,9 @@ static int skl_create(struct pci_dev *pci, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif - snd_hdac_ext_bus_init(bus, &pci->dev, NULL, ext_ops); - bus->use_posbuf = 1; + snd_hda_ext_bus_init(hbus, pci, NULL, ext_ops, "sklbus"); skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); - bus->bdl_pos_adj = 0; - - mutex_init(&hbus->prepare_mutex); - hbus->pci = pci; - hbus->mixer_assigned = -1; - hbus->modelname = "sklbus"; *rskl = skl; diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 30025d3c16b6..5d5081f80e88 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -8,6 +8,7 @@ // Authors: Keyon Jie <yang.jie(a)linux.intel.com> #include <linux/io.h> +#include <linux/pci.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include "../sof-priv.h" @@ -53,24 +54,25 @@ static const struct hdac_bus_ops bus_core_ops = { /* * This can be used for both with/without hda link support. */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev) +void sof_hda_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const char *model) { #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - snd_hdac_ext_bus_init(bus, dev, &bus_core_ops, sof_hda_ext_ops); + snd_hda_ext_bus_init(bus, pdev, &bus_core_ops, sof_hda_ext_ops, model); #else /* CONFIG_SND_SOC_SOF_HDA */ memset(bus, 0, sizeof(*bus)); - bus->dev = dev; + bus->core.dev = &pdev->dev; - INIT_LIST_HEAD(&bus->stream_list); + INIT_LIST_HEAD(&bus->core.stream_list); - bus->irq = -1; + bus->core.irq = -1; /* * There is only one HDA bus atm. keep the index as 0. * Need to fix when there are more than one HDA bus. */ - bus->idx = 0; + bus->core.idx = 0; - spin_lock_init(&bus->reg_lock); + spin_lock_init(&bus->core.reg_lock); #endif /* CONFIG_SND_SOC_SOF_HDA */ } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index fbc2421c77f8..03a68d286c7c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -609,8 +609,9 @@ static int hda_init(struct snd_sof_dev *sdev) bus = sof_to_bus(sdev); /* HDA bus init */ - sof_hda_bus_init(bus, &pci->dev); + sof_hda_bus_init(hbus, pci, hda_model); +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) bus->use_posbuf = 1; bus->bdl_pos_adj = 0; bus->sync_write = 1; @@ -619,7 +620,7 @@ static int hda_init(struct snd_sof_dev *sdev) hbus->pci = pci; hbus->mixer_assigned = -1; hbus->modelname = hda_model; - +#endif /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); bus->remap_addr = pci_ioremap_bar(pci, 0); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 1195018a1f4f..a4ec88f36512 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -635,7 +635,8 @@ void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev); /* * HDA bus operations. */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev); +void sof_hda_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const char *model); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* -- 2.25.1

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[PATCH] Revert "ASoC: rockchip: add config for rockchip dmaengine pcm register"
by Sugar Zhang 21 Oct '21

21 Oct '21

This reverts commit 75b31192fe6ad20b42276b20ee3bdf1493216d63. The original purpose of customized pcm was to config prealloc buffer size flexibly. but, we can do the same thing by soc-generic-dmaengine-pcm. And the generic one can generated the better config by querying DMA capabilities from dmaengine driver rather than the Hard-Coded one. e.g. the customized one: static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, ... the generic one: ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause && dma_caps.cmd_resume) hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw.info |= SNDRV_PCM_INFO_BATCH; ... So, let's revert back to use the generic dmaengine pcm. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/Makefile | 3 +-- sound/soc/rockchip/rockchip_i2s.c | 3 +-- sound/soc/rockchip/rockchip_pcm.c | 44 --------------------------------------- sound/soc/rockchip/rockchip_pcm.h | 11 ---------- 4 files changed, 2 insertions(+), 59 deletions(-) delete mode 100644 sound/soc/rockchip/rockchip_pcm.c delete mode 100644 sound/soc/rockchip/rockchip_pcm.h diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 65e814d..05b078e 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,11 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o -snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 7e89f5b..a6d7656 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -20,7 +20,6 @@ #include <sound/dmaengine_pcm.h> #include "rockchip_i2s.h" -#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -756,7 +755,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = rockchip_pcm_platform_register(&pdev->dev); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); goto err_suspend; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c deleted file mode 100644 index 02254e4..0000000 --- a/sound/soc/rockchip/rockchip_pcm.c +++ /dev/null @@ -1,44 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#include <linux/device.h> -#include <linux/init.h> -#include <linux/module.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/dmaengine_pcm.h> - -#include "rockchip_pcm.h" - -static const struct snd_pcm_hardware snd_rockchip_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_INTERLEAVED, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 52, - .buffer_bytes_max = 64 * 1024, - .fifo_size = 32, -}; - -static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { - .pcm_hardware = &snd_rockchip_hardware, - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .prealloc_buffer_size = 32 * 1024, -}; - -int rockchip_pcm_platform_register(struct device *dev) -{ - return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); -} -EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); - -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h deleted file mode 100644 index 7f00e2c..0000000 --- a/sound/soc/rockchip/rockchip_pcm.h +++ /dev/null @@ -1,11 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#ifndef _ROCKCHIP_PCM_H -#define _ROCKCHIP_PCM_H - -int rockchip_pcm_platform_register(struct device *dev); - -#endif -- 2.7.4

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[PATCH 0/2] ASoC: qcom: sm8250: add support for TX and RX Macro dais
by Srinivas Kandagatla 21 Oct '21

21 Oct '21

This patchset adds support for testing WCD938X connected via TX and RX Macros on SM8250 MTP. Srinivas Kandagatla (2): ASoC: qcom: sm8250: add support for TX and RX Macro dais ASoC: qcom: sm8250: Add Jack support sound/soc/qcom/sm8250.c | 79 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 79 insertions(+) -- 2.21.0

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a52 plugin not working with PipeWire
by GitHub issues - edited 21 Oct '21

21 Oct '21

alsa-project/alsa-plugins issue #37 was edited from sjnewbury: Using the lastest code since v1.2.5 I can successfully get AC3 output through HDMI when using speaker-test. Defaults use a huge buffer and gives terrible latency, but it works. PulseAudio also works fine with fairly low-latency using "tsched=1 tsched_buffer_size=32768 tsched_buffer_watermark=8192 fixed_latency_range=yes mmap=1". This doesn't need the rate kludge any more. When I try pipewire using mmap I get the below messages in the log, no audio: `spa.alsa: alsa-pcm a52hdmi: mmap_commit wrote 512 instead of 1024` With mmap disabled, no errors are reported, but still no sound. Is there a configuration for pipewire which works with the a52 plugin, a pipewire bug, or is this another a52 plugin bug? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/37 Repository URL: https://github.com/alsa-project/alsa-plugins

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