July 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound/soc/codecs/tas5720.c: add ACPI support
by Nikolaus Voss 04 Jul '19

04 Jul '19

Add support for ACPI enumeration for tas5720 and tas5722. Use device_match API to unify access to driver data for DT and ACPI. Aggregate variant stuff into its own struct and directly reference it in variant data for i2c/of/acpi_device_id. Signed-off-by: Nikolaus Voss <nikolaus.voss(a)loewensteinmedical.de> --- sound/soc/codecs/tas5720.c | 215 +++++++++++++++++-------------------- 1 file changed, 99 insertions(+), 116 deletions(-) diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 37fab8f22800..ea973764c745 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -7,6 +7,7 @@ * Author: Andreas Dannenberg <dannenberg(a)ti.com> */ +#include <linux/acpi.h> #include <linux/module.h> #include <linux/errno.h> #include <linux/device.h> @@ -28,9 +29,10 @@ /* Define how often to check (and clear) the fault status register (in ms) */ #define TAS5720_FAULT_CHECK_INTERVAL 200 -enum tas572x_type { - TAS5720, - TAS5722, +struct tas5720_variant { + const int device_id; + const struct regmap_config reg_config; + const struct snd_soc_component_driver comp_drv; }; static const char * const tas5720_supply_names[] = { @@ -44,7 +46,7 @@ struct tas5720_data { struct snd_soc_component *component; struct regmap *regmap; struct i2c_client *tas5720_client; - enum tas572x_type devtype; + const struct tas5720_variant *variant; struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; struct delayed_work fault_check_work; unsigned int last_fault; @@ -179,17 +181,13 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, goto error_snd_soc_component_update_bits; /* Configure TDM slot width. This is only applicable to TAS5722. */ - switch (tas5720->devtype) { - case TAS5722: + if (tas5720->variant->device_id == TAS5722_DEVICE_ID) { ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, TAS5722_TDM_SLOT_16B, slot_width == 16 ? TAS5722_TDM_SLOT_16B : 0); if (ret < 0) goto error_snd_soc_component_update_bits; - break; - default: - break; } return 0; @@ -277,7 +275,7 @@ static void tas5720_fault_check_work(struct work_struct *work) static int tas5720_codec_probe(struct snd_soc_component *component) { struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component); - unsigned int device_id, expected_device_id; + unsigned int device_id; int ret; tas5720->component = component; @@ -301,21 +299,9 @@ static int tas5720_codec_probe(struct snd_soc_component *component) 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(component->dev, "unexpected private driver data\n"); - return -EINVAL; - } - - if (device_id != expected_device_id) + if (device_id != tas5720->variant->device_id) dev_warn(component->dev, "wrong device ID. expected: %u read: %u\n", - expected_device_id, device_id); + tas5720->variant->device_id, device_id); /* Set device to mute */ ret = snd_soc_component_update_bits(component, TAS5720_DIGITAL_CTRL2_REG, @@ -462,24 +448,6 @@ static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config tas5720_regmap_config = { - .reg_bits = 8, - .val_bits = 8, - - .max_register = TAS5720_MAX_REG, - .cache_type = REGCACHE_RBTREE, - .volatile_reg = tas5720_is_volatile_reg, -}; - -static const struct regmap_config tas5722_regmap_config = { - .reg_bits = 8, - .val_bits = 8, - - .max_register = TAS5722_MAX_REG, - .cache_type = REGCACHE_RBTREE, - .volatile_reg = tas5720_is_volatile_reg, -}; - /* * DAC analog gain. There are four discrete values to select from, ranging * from 19.2 dB to 26.3dB. @@ -558,40 +526,6 @@ static const struct snd_soc_dapm_route tas5720_audio_map[] = { { "OUT", NULL, "DAC" }, }; -static const struct snd_soc_component_driver soc_component_dev_tas5720 = { - .probe = tas5720_codec_probe, - .remove = tas5720_codec_remove, - .suspend = tas5720_suspend, - .resume = tas5720_resume, - .controls = tas5720_snd_controls, - .num_controls = ARRAY_SIZE(tas5720_snd_controls), - .dapm_widgets = tas5720_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), - .dapm_routes = tas5720_audio_map, - .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -static const struct snd_soc_component_driver soc_component_dev_tas5722 = { - .probe = tas5720_codec_probe, - .remove = tas5720_codec_remove, - .suspend = tas5720_suspend, - .resume = tas5720_resume, - .controls = tas5722_snd_controls, - .num_controls = ARRAY_SIZE(tas5722_snd_controls), - .dapm_widgets = tas5720_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), - .dapm_routes = tas5720_audio_map, - .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - /* PCM rates supported by the TAS5720 driver */ #define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) @@ -637,29 +571,25 @@ static int tas5720_probe(struct i2c_client *client, { struct device *dev = &client->dev; struct tas5720_data *data; - const struct regmap_config *regmap_config; + const struct tas5720_variant *type; int ret; int i; + type = device_get_match_data(&client->dev); + if (!type && id) + type = (const struct tas5720_variant *)id->driver_data; + + if (!type) + return -EINVAL; + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); if (!data) return -ENOMEM; data->tas5720_client = client; - data->devtype = id->driver_data; + data->variant = type; - switch (id->driver_data) { - case TAS5720: - regmap_config = &tas5720_regmap_config; - break; - case TAS5722: - regmap_config = &tas5722_regmap_config; - break; - default: - dev_err(dev, "unexpected private driver data\n"); - return -EINVAL; - } - data->regmap = devm_regmap_init_i2c(client, regmap_config); + data->regmap = devm_regmap_init_i2c(client, &type->reg_config); if (IS_ERR(data->regmap)) { ret = PTR_ERR(data->regmap); dev_err(dev, "failed to allocate register map: %d\n", ret); @@ -678,51 +608,104 @@ static int tas5720_probe(struct i2c_client *client, dev_set_drvdata(dev, data); - switch (id->driver_data) { - case TAS5720: - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5720, - tas5720_dai, - ARRAY_SIZE(tas5720_dai)); - break; - case TAS5722: - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5722, - tas5720_dai, - ARRAY_SIZE(tas5720_dai)); - break; - default: - dev_err(dev, "unexpected private driver data\n"); - return -EINVAL; - } - if (ret < 0) { - dev_err(dev, "failed to register component: %d\n", ret); - return ret; - } + ret = devm_snd_soc_register_component(&client->dev, + &type->comp_drv, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); - return 0; + if (ret < 0) + dev_err(dev, "failed to register component: %d\n", ret); + + return ret; } +static const struct tas5720_variant variant[] = { + { + .device_id = TAS5720_DEVICE_ID, + .reg_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS5720_MAX_REG, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5720_is_volatile_reg, + }, + .comp_drv = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5720_snd_controls, + .num_controls = ARRAY_SIZE(tas5720_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1 + }, + }, + { + .device_id = TAS5722_DEVICE_ID, + .reg_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS5722_MAX_REG, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5720_is_volatile_reg, + }, + .comp_drv = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5722_snd_controls, + .num_controls = ARRAY_SIZE(tas5722_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + }, + }, +}; + static const struct i2c_device_id tas5720_id[] = { - { "tas5720", TAS5720 }, - { "tas5722", TAS5722 }, + { "tas5720", (kernel_ulong_t)&variant[0] }, + { "tas5722", (kernel_ulong_t)&variant[1] }, { } }; 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", }, + { .compatible = "ti,tas5720", .data = &variant[0], }, + { .compatible = "ti,tas5722", .data = &variant[1], }, { }, }; MODULE_DEVICE_TABLE(of, tas5720_of_match); #endif +#if IS_ENABLED(CONFIG_ACPI) +static const struct acpi_device_id tas5720_acpi_match[] = { + { "10TI5720", (kernel_ulong_t)&variant[0] }, + { "10TI5722", (kernel_ulong_t)&variant[1] }, + { } +}; +MODULE_DEVICE_TABLE(acpi, tas5720_acpi_match); +#endif + static struct i2c_driver tas5720_i2c_driver = { .driver = { .name = "tas5720", .of_match_table = of_match_ptr(tas5720_of_match), + .acpi_match_table = ACPI_PTR(tas5720_acpi_match), }, .probe = tas5720_probe, .id_table = tas5720_id, -- 2.17.1

4 13

[alsa-devel] [PATCH V2 0/2] recover the channel swap after xrun
by shengjiu.wang＠nxp.com 03 Jul '19

03 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> recover the channel swap after xrun Shengjiu Wang (2): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun sound/soc/fsl/fsl_esai.c | 267 ++++++++++++++++++++++++++++----------- 1 file changed, 194 insertions(+), 73 deletions(-) Changes in v2 - add one patch for wrap operations to functions. - fix some coding style issue -- 2.21.0

3 5

[alsa-devel] when aplaymidi exits the sustained notes are played forever.
by GitHub issues - opened 03 Jul '19

03 Jul '19

alsa-project/alsa-utils issue #22 was opened from mercuriete: Transfered from here: https://github.com/OpenTTD/OpenTTD/issues/7637 ### Expected Result All notes are stoped before exit aplaymidi ### Actual Result Notes are sustained forever and I have to press the panic button on Qsynth ### Steps to Reproduce 1. Open Qsynth with fluid-soundfonts 2. do this: ```aplaymidi --port 128:0 file.mid``` 3. do control+c to kill aplaymidi 4. Qsynth plays a note forever. ### Possible fix Capture SIGINT and send midi off messages before exit properly. Issue URL : https://github.com/alsa-project/alsa-utils/issues/22 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH 1/2] ASoC: codecs: ad193x: Group register initialization at probe
by Codrin Ciubotariu 03 Jul '19

03 Jul '19

Create a structure with the register initialization values at probe and use it to initialize all the registers at once. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- The order of the initialization is changed, but it doesn't seem to matter. There is one checkpatch warning, let me know if you want to remove it: WARNING: line over 80 characters #32: FILE: sound/soc/codecs/ad193x.c:425: + { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ sound/soc/codecs/ad193x.c | 52 +++++++++++++++++++++++++-------------- 1 file changed, 33 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 3ebc0524f4b2..f944228f014e 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -413,6 +413,38 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { .ops = &ad193x_dai_ops, }; +struct ad193x_reg_default { + unsigned int reg; + unsigned int val; +}; + +/* codec register values to set after reset */ +static void ad193x_reg_default_init(struct ad193x_priv *ad193x) +{ + const struct ad193x_reg_default reg_init[] = { + { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ + { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ + { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ + { 4, 0x1A }, /* DAC_CTRL2: 48kHz de-emphasis, unmute dac */ + { 5, 0x00 }, /* DAC_CHNL_MUTE: unmute DAC channels */ + }; + const struct ad193x_reg_default reg_adc_init[] = { + { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ + { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ + }; + int i; + + for (i = 0; i < ARRAY_SIZE(reg_init); i++) + regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val); + + if (ad193x_has_adc(ad193x)) { + for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) { + regmap_write(ad193x->regmap, reg_adc_init[i].reg, + reg_adc_init[i].val); + } + } +} + static int ad193x_component_probe(struct snd_soc_component *component) { struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(component); @@ -420,25 +452,7 @@ static int ad193x_component_probe(struct snd_soc_component *component) int num, ret; /* default setting for ad193x */ - - /* unmute dac channels */ - regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0); - /* de-emphasis: 48kHz, powedown dac */ - regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); - /* dac in tdm mode */ - regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - - /* adc only */ - if (ad193x_has_adc(ad193x)) { - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); - } - - /* pll input: mclki/xi */ - regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ - regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + ad193x_reg_default_init(ad193x); /* adc only */ if (ad193x_has_adc(ad193x)) { -- 2.20.1

4 5

[alsa-devel] [PATCH v2 35/35] sound/soc/intel: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 03 Jul '19

03 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memcpy, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memcpy. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- Changes in v2: - Fix a typo in commit message (memset -> memcpy) - Split into two patches sound/soc/intel/atom/sst/sst_loader.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index ce11c36848c4..cc95af35c060 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -288,14 +288,13 @@ static int sst_cache_and_parse_fw(struct intel_sst_drv *sst, { int retval = 0; - sst->fw_in_mem = kzalloc(fw->size, GFP_KERNEL); + sst->fw_in_mem = kmemdup(fw->data, fw->size, GFP_KERNEL); if (!sst->fw_in_mem) { retval = -ENOMEM; goto end_release; } dev_dbg(sst->dev, "copied fw to %p", sst->fw_in_mem); dev_dbg(sst->dev, "phys: %lx", (unsigned long)virt_to_phys(sst->fw_in_mem)); - memcpy(sst->fw_in_mem, fw->data, fw->size); retval = sst_parse_fw_memcpy(sst, fw->size, &sst->memcpy_list); if (retval) { dev_err(sst->dev, "Failed to parse fw\n"); -- 2.11.0

1 0

[alsa-devel] [PATCH 30/30] sound/soc: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 03 Jul '19

03 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memset, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memset. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- sound/soc/codecs/wm0010.c | 4 +--- sound/soc/intel/atom/sst/sst_loader.c | 3 +-- 2 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 727d6703c905..807826f30f58 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -515,7 +515,7 @@ static int wm0010_stage2_load(struct snd_soc_component *component) dev_dbg(component->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); + img = kmemdup(&fw->data[0], fw->size, GFP_KERNEL | GFP_DMA); if (!img) { ret = -ENOMEM; goto abort2; @@ -527,8 +527,6 @@ static int wm0010_stage2_load(struct snd_soc_component *component) goto abort1; } - memcpy(img, &fw->data[0], fw->size); - spi_message_init(&m); memset(&t, 0, sizeof(t)); t.rx_buf = out; diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index ce11c36848c4..cc95af35c060 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -288,14 +288,13 @@ static int sst_cache_and_parse_fw(struct intel_sst_drv *sst, { int retval = 0; - sst->fw_in_mem = kzalloc(fw->size, GFP_KERNEL); + sst->fw_in_mem = kmemdup(fw->data, fw->size, GFP_KERNEL); if (!sst->fw_in_mem) { retval = -ENOMEM; goto end_release; } dev_dbg(sst->dev, "copied fw to %p", sst->fw_in_mem); dev_dbg(sst->dev, "phys: %lx", (unsigned long)virt_to_phys(sst->fw_in_mem)); - memcpy(sst->fw_in_mem, fw->data, fw->size); retval = sst_parse_fw_memcpy(sst, fw->size, &sst->memcpy_list); if (retval) { dev_err(sst->dev, "Failed to parse fw\n"); -- 2.11.0

2 2

[alsa-devel] [PATCH 29/30] sound/pci: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 03 Jul '19

03 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memset, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memset. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index b612a536a5a1..35bd3e7c8ce1 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2189,11 +2189,10 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); + commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { -- 2.11.0

1 0

[alsa-devel] [PATCH 0/3] pcm_file report EIO when fail to write to debug file
by Adam Miartus 03 Jul '19

03 Jul '19

Previously, I introduced a patch that may return EPIPE error when pcm_file plugin fails to write to output file for some reason. The purpose of reporting the error was to evade assert in code to allow API user error handling and recovery, or more clarity in debugging. Failing to write to a file usually means bad file descriptor, running out of free memory, insufficient file permissions or in case file is a pipe there may be other reasons preventing the write operation. All these are not easily recoverable by restaring playback as EPIPE error which is used to signal XRUN might suggest. Therefore, use EIO to signal a different kind of error. write_wav_header function is refactored a bit for more detailed error handling in case of failed write operation. Adam Miartus (3): pcm_file: use EIO instead of EPIPE when failing to write output file pcm_file: improve error checking in write_wav_header function pcm_file: in case of failed write clear file buffer variables src/pcm/pcm_file.c | 54 +++++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 41 insertions(+), 13 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: sunxi: sun50i-codec-analog: Add earpiece
by Luca Weiss 03 Jul '19

03 Jul '19

This adds the necessary registers and audio routes to play audio using the Earpiece, that's supported on the A64. Signed-off-by: Luca Weiss <luca(a)z3ntu.xyz> --- So, first of all: This is my first audio patch and I hope I didn't make too many mistakes :) , especially with the routes at the bottom of the patch. What I'm really unsure about, is how the enable & mute registers should be handled. Should I put both registers into a SOC_DOUBLE("Earpiece Playback Switch",...)? sound/soc/sunxi/sun50i-codec-analog.c | 51 +++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index d105c90c3706..6c19fea992c5 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -49,6 +49,15 @@ #define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 #define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 +#define SUN50I_ADDA_EARPIECE_CTRL0 0x03 +#define SUN50I_ADDA_EARPIECE_CTRL0_EAR_RAMP_TIME 4 +#define SUN50I_ADDA_EARPIECE_CTRL0_ESPSR 0 + +#define SUN50I_ADDA_EARPIECE_CTRL1 0x04 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN 7 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE 6 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL 0 + #define SUN50I_ADDA_LINEOUT_CTRL0 0x05 #define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 #define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 @@ -172,6 +181,10 @@ static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), ); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_earpiece_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); /* volume / mute controls */ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { @@ -225,6 +238,19 @@ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { SUN50I_ADDA_LINEOUT_CTRL0_LEN, SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + SOC_SINGLE_TLV("Earpiece Playback Volume", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL, 0x1f, 0, + sun50i_codec_earpiece_vol_scale), + + SOC_SINGLE("Earpiece Playback Switch (enable)", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN, 1, 0), + + SOC_SINGLE("Earpiece Playback Switch", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE, 1, 0), + }; static const char * const sun50i_codec_hp_src_enum_text[] = { @@ -257,6 +283,20 @@ static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { sun50i_codec_lineout_src_enum), }; +static const char * const sun50i_codec_earpiece_src_enum_text[] = { + "DACR", "DACL", "Right Analog Mixer", "Left Analog Mixer", +}; + +static SOC_ENUM_SINGLE_DECL(sun50i_codec_earpiece_src_enum, + SUN50I_ADDA_EARPIECE_CTRL0, + SUN50I_ADDA_EARPIECE_CTRL0_ESPSR, + sun50i_codec_earpiece_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_earpiece_src[] = { + SOC_DAPM_ENUM("Earpiece Source Playback Route", + sun50i_codec_earpiece_src_enum), +}; + static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { /* DAC */ SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, @@ -285,6 +325,10 @@ static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_MUX("Earpiece Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_earpiece_src), + SND_SOC_DAPM_OUTPUT("EARPIECE"), + /* Microphone inputs */ SND_SOC_DAPM_INPUT("MIC1"), @@ -388,6 +432,13 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, + + /* Earpiece Routes */ + { "Earpiece Source Playback Route", "DACL", "Left DAC" }, + { "Earpiece Source Playback Route", "DACR", "Right DAC" }, + { "Earpiece Source Playback Route", "Left Analog Mixer", "Left Mixer" }, + { "Earpiece Source Playback Route", "Right Analog Mixer", "Right Mixer" }, + { "EARPIECE", NULL, "Earpiece Source Playback Route" }, }; static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { -- 2.22.0

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[alsa-devel] [PATCH] ALSA: usb-audio: fix Line6 Helix audio format rates
by Nicola Lunghi 03 Jul '19

03 Jul '19

Line6 Helix and HX stomp don't support retrieving the number of clock sample rate. Add a quirk to return the default value of 48Khz. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 28 +++++++++++++++++++++++++--- 1 file changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index c02b51a82775..05442f6ada62 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -313,10 +313,32 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, tmp, sizeof(tmp)); if (ret < 0) { - dev_err(&dev->dev, - "%s(): unable to retrieve number of sample rates (clock %d)\n", + switch (chip->usb_id) { + /* LINE 6 HX pedals don't support getting the clock sample rate. + * Set the framerate to 48khz by default + */ + case USB_ID(0x0E41, 0x4244): /* HELIX */ + case USB_ID(0x0E41, 0x4246): /* HX STOMP */ + dev_warn(&dev->dev, + "%s(): line6 helix: unable to retrieve number of sample rates. Set it to default value (clock %d).\n", __func__, clock); - goto err; + fp->nr_rates = 1; + fp->rate_min = 48000; + fp->rate_max = 48000; + fp->rates = SNDRV_PCM_RATE_48000; + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + if (!fp->rate_table) { + ret = -ENOMEM; + goto err_free; + } + fp->rate_table[0] = 48000; + return 0; + default: + dev_err(&dev->dev, + "%s(): unable to retrieve number of sample rates (clock %d)\n", + __func__, clock); + goto err; + } } nr_triplets = (tmp[1] << 8) | tmp[0]; -- 2.19.1

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