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[alsa-devel] [PATCH] ALSA: hda_intel: add position_fix quirk for Asus K53E
by Catalin Iacob 12 Sep '12

12 Sep '12

Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including repeated sounds on my Asus laptop. My hardware is Cougar Point which the commit log of c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO probably works in general but apparently it doesn't on Asus K53E therefore the need for the quirk. Signed-off-by: Catalin Iacob <iacobcatalin(a)gmail.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c6..228cdf9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2701,6 +2701,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), -- 1.7.11.5

2 4

[alsa-devel] [PATCH 1/2] ASoC: wm8737: Move regulator acquisition to device registration
by Mark Brown 12 Sep '12

12 Sep '12

This is better style as we acquire resources we will need before we go into the ASoC card probe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8737.c | 41 +++++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 8c25442..3bf5bc7 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -557,7 +557,7 @@ static int wm8737_resume(struct snd_soc_codec *codec) static int wm8737_probe(struct snd_soc_codec *codec) { struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - int ret, i; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8737->control_type); if (ret != 0) { @@ -565,16 +565,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) - wm8737->supplies[i].supply = wm8737_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8737->supplies), - wm8737->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); if (ret != 0) { @@ -607,17 +597,12 @@ static int wm8737_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); - return ret; } static int wm8737_remove(struct snd_soc_codec *codec) { - struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); return 0; } @@ -645,13 +630,23 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&i2c->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8737); wm8737->control_type = SND_SOC_I2C; @@ -691,13 +686,23 @@ static struct i2c_driver wm8737_i2c_driver = { static int __devinit wm8737_spi_probe(struct spi_device *spi) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&spi->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&spi->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + wm8737->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8737); -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm8728: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8728.c | 45 ++++++++++++++++++++++++++++----------------- 1 file changed, 28 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f274c25..00a12a0 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -17,6 +17,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -35,16 +36,16 @@ * the volume update bits, mute the output and enable infinite zero * detect. */ -static const u16 wm8728_reg_defaults[] = { - 0x1ff, - 0x1ff, - 0x001, - 0x100, +static const struct reg_default wm8728_reg_defaults[] = { + { 0, 0x1ff }, + { 1, 0x1ff }, + { 2, 0x001 }, + { 3, 0x100 }, }; /* codec private data */ struct wm8728_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); @@ -162,8 +163,8 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8728_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); u16 reg; - int i; switch (level) { case SND_SOC_BIAS_ON: @@ -175,9 +176,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); /* ..then sync in the register cache. */ - for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) - snd_soc_write(codec, i, - snd_soc_read(codec, i)); + regcache_sync(wm8728->regmap); } break; @@ -229,10 +228,9 @@ static int wm8728_resume(struct snd_soc_codec *codec) static int wm8728_probe(struct snd_soc_codec *codec) { - struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8728->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", ret); @@ -257,9 +255,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .suspend = wm8728_suspend, .resume = wm8728_resume, .set_bias_level = wm8728_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8728_reg_defaults, .controls = wm8728_snd_controls, .num_controls = ARRAY_SIZE(wm8728_snd_controls), .dapm_widgets = wm8728_dapm_widgets, @@ -274,6 +269,16 @@ static const struct of_device_id wm8728_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8728_of_match); +static const struct regmap_config wm8728_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8728_IFCTL, + + .reg_defaults = wm8728_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8728_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8728_spi_probe(struct spi_device *spi) { @@ -285,7 +290,10 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi) if (wm8728 == NULL) return -ENOMEM; - wm8728->control_type = SND_SOC_SPI; + wm8728->regmap = devm_regmap_init_spi(spi, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + spi_set_drvdata(spi, wm8728); ret = snd_soc_register_codec(&spi->dev, @@ -324,8 +332,11 @@ static __devinit int wm8728_i2c_probe(struct i2c_client *i2c, if (wm8728 == NULL) return -ENOMEM; + wm8728->regmap = devm_regmap_init_i2c(i2c, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + i2c_set_clientdata(i2c, wm8728); - wm8728->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8728, &wm8728_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8711: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8711.c | 38 ++++++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 1b1209a..5c165d0 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -32,7 +33,7 @@ /* codec private data */ struct wm8711_priv { - enum snd_soc_control_type bus_type; + struct regmap *regmap; unsigned int sysclk; }; @@ -42,9 +43,9 @@ struct wm8711_priv { * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { - 0x0079, 0x0079, 0x000a, 0x0008, - 0x009f, 0x000a, 0x0000, 0x0000 +static const struct reg_default wm8711_reg_defaults[] = { + { 0, 0x0079 }, { 1, 0x0079 }, { 2, 0x000a }, { 3, 0x0008 }, + { 4, 0x009f }, { 5, 0x000a }, { 6, 0x0000 }, { 7, 0x0000 }, }; #define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) @@ -289,6 +290,7 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8711_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f; switch (level) { @@ -299,7 +301,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - snd_soc_cache_sync(codec); + regcache_sync(wm8711->regmap); snd_soc_write(codec, WM8711_PWR, reg | 0x0040); break; @@ -353,10 +355,9 @@ static int wm8711_resume(struct snd_soc_codec *codec) static int wm8711_probe(struct snd_soc_codec *codec) { - struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -391,9 +392,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .suspend = wm8711_suspend, .resume = wm8711_resume, .set_bias_level = wm8711_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8711_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8711_reg, .controls = wm8711_snd_controls, .num_controls = ARRAY_SIZE(wm8711_snd_controls), .dapm_widgets = wm8711_dapm_widgets, @@ -408,6 +406,16 @@ static const struct of_device_id wm8711_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8711_of_match); +static const struct regmap_config wm8711_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8711_RESET, + + .reg_defaults = wm8711_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8711_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -419,8 +427,11 @@ static int __devinit wm8711_spi_probe(struct spi_device *spi) if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_spi(spi, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + spi_set_drvdata(spi, wm8711); - wm8711->bus_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); @@ -458,8 +469,11 @@ static __devinit int wm8711_i2c_probe(struct i2c_client *client, if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_i2c(client, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + i2c_set_clientdata(client, wm8711); - wm8711->bus_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8580: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 104 +++++++++++++++++++++++++++++++++------------ 1 file changed, 78 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index cc198df..2a07c70 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008-11 Wolfson Microelectronics PLC. + * Copyright 2008-12 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -23,6 +23,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -157,21 +158,60 @@ * We can't read the WM8580 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8580_reg[] = { - 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ - 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ - 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ - 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R27*/ - 0x01f0, 0x0040, 0x0000, 0x0000, /*R31(0x1F)*/ - 0x0000, 0x0000, 0x0031, 0x000b, /*R35*/ - 0x0039, 0x0000, 0x0010, 0x0032, /*R39*/ - 0x0054, 0x0076, 0x0098, 0x0000, /*R43(0x2B)*/ - 0x0000, 0x0000, 0x0000, 0x0000, /*R47*/ - 0x0000, 0x0000, 0x005e, 0x003e, /*R51(0x33)*/ - 0x0000, 0x0000 /*R53*/ +static const struct reg_default wm8580_reg_defaults[] = { + { 0, 0x0121 }, + { 1, 0x017e }, + { 2, 0x007d }, + { 3, 0x0014 }, + { 4, 0x0121 }, + { 5, 0x017e }, + { 6, 0x007d }, + { 7, 0x0194 }, + { 8, 0x0010 }, + { 9, 0x0002 }, + { 10, 0x0002 }, + { 11, 0x00c2 }, + { 12, 0x0182 }, + { 13, 0x0082 }, + { 14, 0x000a }, + { 15, 0x0024 }, + { 16, 0x0009 }, + { 17, 0x0000 }, + { 18, 0x00ff }, + { 19, 0x0000 }, + { 20, 0x00ff }, + { 21, 0x00ff }, + { 22, 0x00ff }, + { 23, 0x00ff }, + { 24, 0x00ff }, + { 25, 0x00ff }, + { 26, 0x00ff }, + { 27, 0x00ff }, + { 28, 0x01f0 }, + { 29, 0x0040 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0031 }, + { 35, 0x000b }, + { 36, 0x0039 }, + { 37, 0x0000 }, + { 38, 0x0010 }, + { 39, 0x0032 }, + { 40, 0x0054 }, + { 41, 0x0076 }, + { 42, 0x0098 }, + { 43, 0x0000 }, + { 44, 0x0000 }, + { 45, 0x0000 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0000 }, + { 50, 0x005e }, + { 51, 0x003e }, + { 52, 0x0000 }, }; struct pll_state { @@ -188,7 +228,7 @@ static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { /* codec private data */ struct wm8580_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES]; struct pll_state a; struct pll_state b; @@ -203,14 +243,16 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; + struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); unsigned int reg = mc->reg; unsigned int reg2 = mc->rreg; int ret; - /* Clear the register cache so we write without VU set */ - reg_cache[reg] = 0; - reg_cache[reg2] = 0; + /* Clear the register cache VU so we write without VU set */ + regcache_cache_only(wm8580->regmap, true); + regmap_update_bits(wm8580->regmap, reg, 0x100, 0x000); + regmap_update_bits(wm8580->regmap, reg2, 0x100, 0x000); + regcache_cache_only(wm8580->regmap, false); ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) @@ -817,7 +859,7 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0,i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8580->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -875,9 +917,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8580_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8580_reg, .controls = wm8580_snd_controls, .num_controls = ARRAY_SIZE(wm8580_snd_controls), @@ -892,6 +931,16 @@ static const struct of_device_id wm8580_of_match[] = { { }, }; +static const struct regmap_config wm8580_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8580_MAX_REGISTER, + + .reg_defaults = wm8580_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8580_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -904,8 +953,11 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, if (wm8580 == NULL) return -ENOMEM; + wm8580->regmap = devm_regmap_init_i2c(i2c, &wm8580_regmap); + if (IS_ERR(wm8580->regmap)) + return PTR_ERR(wm8580->regmap); + i2c_set_clientdata(i2c, wm8580); - wm8580->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8580, wm8580_dai, ARRAY_SIZE(wm8580_dai)); -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm8510: Convert to direct regmap API usage
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 105 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 80 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 98f28a0..d9a3ca6 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/of_device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -33,22 +34,63 @@ * We can't read the WM8510 register space when we are * using 2 wire for device control, so we cache them instead. */ -static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0050, 0x0000, 0x0140, 0x0000, - 0x0000, 0x0000, 0x0000, 0x00ff, - 0x0000, 0x0000, 0x0100, 0x00ff, - 0x0000, 0x0000, 0x012c, 0x002c, - 0x002c, 0x002c, 0x002c, 0x0000, - 0x0032, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0038, 0x000b, 0x0032, 0x0000, - 0x0008, 0x000c, 0x0093, 0x00e9, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0003, 0x0010, 0x0000, 0x0000, - 0x0000, 0x0002, 0x0001, 0x0000, - 0x0000, 0x0000, 0x0039, 0x0000, - 0x0001, +static const struct reg_default wm8510_reg_defaults[] = { + { 1, 0x0000 }, + { 2, 0x0000 }, + { 3, 0x0000 }, + { 4, 0x0050 }, + { 5, 0x0000 }, + { 6, 0x0140 }, + { 7, 0x0000 }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x0000 }, + { 11, 0x00ff }, + { 12, 0x0000 }, + { 13, 0x0000 }, + { 14, 0x0100 }, + { 15, 0x00ff }, + { 16, 0x0000 }, + { 17, 0x0000 }, + { 18, 0x012c }, + { 19, 0x002c }, + { 20, 0x002c }, + { 21, 0x002c }, + { 22, 0x002c }, + { 23, 0x0000 }, + { 24, 0x0032 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0038 }, + { 33, 0x000b }, + { 34, 0x0032 }, + { 35, 0x0000 }, + { 36, 0x0008 }, + { 37, 0x000c }, + { 38, 0x0093 }, + { 39, 0x00e9 }, + { 40, 0x0000 }, + { 41, 0x0000 }, + { 42, 0x0000 }, + { 43, 0x0000 }, + { 44, 0x0003 }, + { 45, 0x0010 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0002 }, + { 50, 0x0001 }, + { 51, 0x0000 }, + { 52, 0x0000 }, + { 53, 0x0000 }, + { 54, 0x0039 }, + { 55, 0x0000 }, + { 56, 0x0001 }, }; #define WM8510_POWER1_BIASEN 0x08 @@ -58,7 +100,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { /* codec private data */ struct wm8510_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; @@ -454,6 +496,7 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3; switch (level) { @@ -467,7 +510,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8510->regmap); /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); @@ -536,10 +579,9 @@ static int wm8510_resume(struct snd_soc_codec *codec) static int wm8510_probe(struct snd_soc_codec *codec) { - struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8510->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); return ret; @@ -569,9 +611,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .suspend = wm8510_suspend, .resume = wm8510_resume, .set_bias_level = wm8510_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8510_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default =wm8510_reg, .controls = wm8510_snd_controls, .num_controls = ARRAY_SIZE(wm8510_snd_controls), @@ -586,6 +625,16 @@ static const struct of_device_id wm8510_of_match[] = { { }, }; +static const struct regmap_config wm8510_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8510_MONOMIX, + + .reg_defaults = wm8510_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8510_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -597,7 +646,10 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) if (wm8510 == NULL) return -ENOMEM; - wm8510->control_type = SND_SOC_SPI; + wm8510->regmap = devm_regmap_init_spi(spi, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + spi_set_drvdata(spi, wm8510); ret = snd_soc_register_codec(&spi->dev, @@ -635,8 +687,11 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, if (wm8510 == NULL) return -ENOMEM; + wm8510->regmap = devm_regmap_init_i2c(i2c, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + i2c_set_clientdata(i2c, wm8510); - wm8510->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/7] ASoC: wm8510: Convert to devm_kzalloc()
by Mark Brown 12 Sep '12

12 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 56a0495..98f28a0 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -592,7 +592,8 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&spi->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -601,8 +602,7 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } @@ -630,7 +630,8 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&i2c->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -639,8 +640,7 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } -- 1.7.10.4

1 6

[alsa-devel] [PATCH] ALSA: hda - Avoid BDL position workaround when no_period_wakeup is set
by Takashi Iwai 11 Sep '12

11 Sep '12

Originally the bogus period at BDL head was introduced as a workaround for the mismatching position update at the period boundary, typically seen on dmix. However, for applications like PulseAudio that don't require period wake ups, this workaround is just superfluous. Thus better to disable it when no_period_wakeup is given in hw_params. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6a19f6a..76ec242 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -415,6 +415,7 @@ struct azx_dev { */ unsigned int insufficient :1; unsigned int wc_marked:1; + unsigned int no_period_wakeup:1; }; /* CORB/RIRB */ @@ -1419,7 +1420,7 @@ static int azx_setup_periods(struct azx *chip, ofs = 0; azx_dev->frags = 0; pos_adj = bdl_pos_adj[chip->dev_index]; - if (pos_adj > 0) { + if (!azx_dev->no_period_wakeup && pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; int pos_align = pos_adj; pos_adj = (pos_adj * runtime->rate + 47999) / 48000; @@ -1435,8 +1436,7 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, - &bdl, ofs, pos_adj, - !substream->runtime->no_period_wakeup); + &bdl, ofs, pos_adj, true); if (ofs < 0) goto error; } @@ -1449,7 +1449,7 @@ static int azx_setup_periods(struct azx *chip, else ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes, - !substream->runtime->no_period_wakeup); + !azx_dev->no_period_wakeup); if (ofs < 0) goto error; } @@ -1922,10 +1922,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || - format_val != azx_dev->format_val) { + format_val != azx_dev->format_val || + runtime->no_period_wakeup != azx_dev->no_period_wakeup) { azx_dev->bufsize = bufsize; azx_dev->period_bytes = period_bytes; azx_dev->format_val = format_val; + azx_dev->no_period_wakeup = runtime->no_period_wakeup; err = azx_setup_periods(chip, substream, azx_dev); if (err < 0) return err; -- 1.7.11.5

1 0

[alsa-devel] [patch] ALSA: compress_core: fix open flags test in snd_compr_open()
by Dan Carpenter 11 Sep '12

11 Sep '12

O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always false and it will never do compress capture. The test for O_WRONLY is also slightly off. The original test would consider "->flags = (O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid. I've also removed the pr_err() because that could flood dmesg. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d..eb60cb8 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f) int maj = imajor(inode); int ret; - if (f->f_flags & O_WRONLY) + if ((f->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; - else if (f->f_flags & O_RDONLY) + else if ((f->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; - else { - pr_err("invalid direction\n"); + else return -EINVAL; - } if (maj == snd_major) compr = snd_lookup_minor_data(iminor(inode),

2 1

[alsa-devel] [PATCH 06/17] ARM: OMAP: Move gpio.h to include/linux/platform_data
by Tony Lindgren 11 Sep '12

11 Sep '12

This way we can remove includes of plat/gpio.h which won't work with the single zImage support. Note that we also remove the cpu_class_is_omap2() check in gpio-omap.c as the drivers should not call it as we need to make it local to arch/arm/mach-omap2 for single zImage support. While at it, arrange the related includes in the standard way. Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: linux-mtd(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- arch/arm/mach-omap1/board-ams-delta.c | 1 + arch/arm/mach-omap1/board-osk.c | 1 + arch/arm/mach-omap1/gpio15xx.c | 1 + arch/arm/mach-omap1/gpio16xx.c | 1 + arch/arm/mach-omap1/gpio7xx.c | 1 + arch/arm/mach-omap1/include/mach/gpio.h | 2 -- arch/arm/mach-omap1/leds-h2p2-debug.c | 1 + arch/arm/mach-omap1/leds.c | 1 + arch/arm/mach-omap2/board-am3517evm.c | 1 + arch/arm/mach-omap2/board-cm-t35.c | 1 + arch/arm/mach-omap2/board-zoom-display.c | 2 ++ arch/arm/mach-omap2/board-zoom-peripherals.c | 1 + arch/arm/mach-omap2/gpio.c | 1 + arch/arm/mach-omap2/hsmmc.c | 2 ++ arch/arm/mach-omap2/include/mach/gpio.h | 2 -- arch/arm/mach-omap2/msdi.c | 1 + arch/arm/mach-omap2/omap_hwmod_2420_data.c | 1 - arch/arm/mach-omap2/omap_hwmod_2430_data.c | 1 - arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c | 2 +- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 2 +- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/mach-omap2/pm24xx.c | 1 + arch/arm/mach-omap2/pm34xx.c | 2 ++ arch/arm/plat-omap/debug-leds.c | 1 + drivers/gpio/gpio-omap.c | 15 ++++++++------- drivers/mtd/nand/ams-delta.c | 8 ++++++-- include/linux/platform_data/gpio-omap.h | 15 ++------------- sound/soc/omap/sdp3430.c | 1 + 28 files changed, 40 insertions(+), 31 deletions(-) rename arch/arm/plat-omap/include/plat/gpio.h => include/linux/platform_data/gpio-omap.h (94%) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index c162369..6f192c4 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -26,6 +26,7 @@ #include <linux/export.h> #include <linux/omapfb.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <media/soc_camera.h> diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 569b687..3b2d907 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -39,6 +39,7 @@ #include <linux/mtd/partitions.h> #include <linux/mtd/physmap.h> #include <linux/i2c/tps65010.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/omap1_bl.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap1/gpio15xx.c b/arch/arm/mach-omap1/gpio15xx.c index ebef15e..98e6f39 100644 --- a/arch/arm/mach-omap1/gpio15xx.c +++ b/arch/arm/mach-omap1/gpio15xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1_MPUIO_VBASE OMAP1_MPUIO_BASE #define OMAP1510_GPIO_BASE 0xFFFCE000 diff --git a/arch/arm/mach-omap1/gpio16xx.c b/arch/arm/mach-omap1/gpio16xx.c index 2a48cd2..33f4192 100644 --- a/arch/arm/mach-omap1/gpio16xx.c +++ b/arch/arm/mach-omap1/gpio16xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1610_GPIO1_BASE 0xfffbe400 #define OMAP1610_GPIO2_BASE 0xfffbec00 diff --git a/arch/arm/mach-omap1/gpio7xx.c b/arch/arm/mach-omap1/gpio7xx.c index acf12b7..958ce9a 100644 --- a/arch/arm/mach-omap1/gpio7xx.c +++ b/arch/arm/mach-omap1/gpio7xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP7XX_GPIO1_BASE 0xfffbc000 #define OMAP7XX_GPIO2_BASE 0xfffbc800 diff --git a/arch/arm/mach-omap1/include/mach/gpio.h b/arch/arm/mach-omap1/include/mach/gpio.h index e737706..ebf86c0 100644 --- a/arch/arm/mach-omap1/include/mach/gpio.h +++ b/arch/arm/mach-omap1/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap1/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap1/leds-h2p2-debug.c b/arch/arm/mach-omap1/leds-h2p2-debug.c index f6b14a1..7f4bba9 100644 --- a/arch/arm/mach-omap1/leds-h2p2-debug.c +++ b/arch/arm/mach-omap1/leds-h2p2-debug.c @@ -14,6 +14,7 @@ #include <linux/kernel_stat.h> #include <linux/sched.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <mach/hardware.h> #include <asm/leds.h> diff --git a/arch/arm/mach-omap1/leds.c b/arch/arm/mach-omap1/leds.c index ae6dd93..7b1a383 100644 --- a/arch/arm/mach-omap1/leds.c +++ b/arch/arm/mach-omap1/leds.c @@ -6,6 +6,7 @@ #include <linux/gpio.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/leds.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap2/board-am3517evm.c b/arch/arm/mach-omap2/board-am3517evm.c index 00abda1..029ebdf 100644 --- a/arch/arm/mach-omap2/board-am3517evm.c +++ b/arch/arm/mach-omap2/board-am3517evm.c @@ -25,6 +25,7 @@ #include <linux/can/platform/ti_hecc.h> #include <linux/davinci_emac.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> #include <mach/hardware.h> #include <mach/am35xx.h> diff --git a/arch/arm/mach-omap2/board-cm-t35.c b/arch/arm/mach-omap2/board-cm-t35.c index ea34109..34cb904 100644 --- a/arch/arm/mach-omap2/board-cm-t35.c +++ b/arch/arm/mach-omap2/board-cm-t35.c @@ -23,6 +23,7 @@ #include <linux/input/matrix_keypad.h> #include <linux/delay.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/i2c/at24.h> #include <linux/i2c/twl.h> diff --git a/arch/arm/mach-omap2/board-zoom-display.c b/arch/arm/mach-omap2/board-zoom-display.c index 28187f1..d64f35f 100644 --- a/arch/arm/mach-omap2/board-zoom-display.c +++ b/arch/arm/mach-omap2/board-zoom-display.c @@ -18,6 +18,8 @@ #include <video/omapdss.h> #include <mach/board-zoom.h> +#include <plat/cpu.h> + #define LCD_PANEL_RESET_GPIO_PROD 96 #define LCD_PANEL_RESET_GPIO_PILOT 55 #define LCD_PANEL_QVGA_GPIO 56 diff --git a/arch/arm/mach-omap2/board-zoom-peripherals.c b/arch/arm/mach-omap2/board-zoom-peripherals.c index 00f73b7..ae5d36b 100644 --- a/arch/arm/mach-omap2/board-zoom-peripherals.c +++ b/arch/arm/mach-omap2/board-zoom-peripherals.c @@ -19,6 +19,7 @@ #include <linux/regulator/fixed.h> #include <linux/wl12xx.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> diff --git a/arch/arm/mach-omap2/gpio.c b/arch/arm/mach-omap2/gpio.c index 9ad7d48..7b52bdf 100644 --- a/arch/arm/mach-omap2/gpio.c +++ b/arch/arm/mach-omap2/gpio.c @@ -21,6 +21,7 @@ #include <linux/slab.h> #include <linux/interrupt.h> #include <linux/of.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/hsmmc.c b/arch/arm/mach-omap2/hsmmc.c index a9675d8..80399d7 100644 --- a/arch/arm/mach-omap2/hsmmc.c +++ b/arch/arm/mach-omap2/hsmmc.c @@ -15,6 +15,8 @@ #include <linux/delay.h> #include <linux/gpio.h> #include <mach/hardware.h> +#include <linux/platform_data/gpio-omap.h> + #include <plat/mmc.h> #include <plat/omap-pm.h> #include <plat/mux.h> diff --git a/arch/arm/mach-omap2/include/mach/gpio.h b/arch/arm/mach-omap2/include/mach/gpio.h index be4d290..5621cc5 100644 --- a/arch/arm/mach-omap2/include/mach/gpio.h +++ b/arch/arm/mach-omap2/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap2/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap2/msdi.c b/arch/arm/mach-omap2/msdi.c index fb5bc6c..9e57b4a 100644 --- a/arch/arm/mach-omap2/msdi.c +++ b/arch/arm/mach-omap2/msdi.c @@ -23,6 +23,7 @@ #include <linux/kernel.h> #include <linux/err.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2420_data.c b/arch/arm/mach-omap2/omap_hwmod_2420_data.c index 50cfab6..5b49ebf 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2420_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2420_data.c @@ -18,7 +18,6 @@ #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mcspi.h> #include <plat/dmtimer.h> #include <plat/l3_2xxx.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2430_data.c b/arch/arm/mach-omap2/omap_hwmod_2430_data.c index 58b5bc1..1fd66f0 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2430_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2430_data.c @@ -18,7 +18,6 @@ #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mcbsp.h> #include <plat/mcspi.h> #include <plat/dmtimer.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c index afad69c..c6416c9 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c @@ -10,7 +10,7 @@ */ #include <plat/omap_hwmod.h> #include <plat/serial.h> -#include <plat/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/dma.h> #include <plat/dmtimer.h> #include <plat/mcspi.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c index c9e3820..5820c73 100644 --- a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c @@ -15,6 +15,7 @@ * XXX these should be marked initdata for multi-OMAP kernels */ #include <linux/power/smartreflex.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <mach/irqs.h> @@ -24,7 +25,6 @@ #include <plat/l3_3xxx.h> #include <plat/l4_3xxx.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mmc.h> #include <plat/mcbsp.h> #include <plat/mcspi.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c index 242aee4..fe987f8 100644 --- a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c @@ -19,12 +19,12 @@ */ #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/power/smartreflex.h> #include <plat/omap_hwmod.h> #include <plat/cpu.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/dma.h> #include <plat/mcspi.h> #include <plat/mcbsp.h> diff --git a/arch/arm/mach-omap2/pm24xx.c b/arch/arm/mach-omap2/pm24xx.c index de1caf1..2708a8b 100644 --- a/arch/arm/mach-omap2/pm24xx.c +++ b/arch/arm/mach-omap2/pm24xx.c @@ -29,6 +29,7 @@ #include <linux/irq.h> #include <linux/time.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach/time.h> #include <asm/mach/irq.h> diff --git a/arch/arm/mach-omap2/pm34xx.c b/arch/arm/mach-omap2/pm34xx.c index 05bd8f0..90480f7 100644 --- a/arch/arm/mach-omap2/pm34xx.c +++ b/arch/arm/mach-omap2/pm34xx.c @@ -28,6 +28,8 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/platform_data/gpio-omap.h> + #include <trace/events/power.h> #include <asm/suspend.h> diff --git a/arch/arm/plat-omap/debug-leds.c b/arch/arm/plat-omap/debug-leds.c index 39407cb..195aaae 100644 --- a/arch/arm/plat-omap/debug-leds.c +++ b/arch/arm/plat-omap/debug-leds.c @@ -12,6 +12,7 @@ #include <linux/platform_device.h> #include <linux/leds.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> #include <mach/hardware.h> #include <asm/leds.h> diff --git a/drivers/gpio/gpio-omap.c b/drivers/gpio/gpio-omap.c index e6efd77..64fbce3 100644 --- a/drivers/gpio/gpio-omap.c +++ b/drivers/gpio/gpio-omap.c @@ -25,11 +25,9 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/irqdomain.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> -#include <asm/irq.h> -#include <mach/irqs.h> -#include <asm/gpio.h> #include <asm/mach/irq.h> #define OFF_MODE 1 @@ -385,13 +383,16 @@ static int _set_gpio_triggering(struct gpio_bank *bank, int gpio, static int gpio_irq_type(struct irq_data *d, unsigned type) { struct gpio_bank *bank = irq_data_get_irq_chip_data(d); - unsigned gpio; + unsigned gpio = 0; int retval; unsigned long flags; - if (!cpu_class_is_omap2() && d->irq > IH_MPUIO_BASE) +#ifdef CONFIG_ARCH_OMAP1 + if (d->irq > IH_MPUIO_BASE) gpio = OMAP_MPUIO(d->irq - IH_MPUIO_BASE); - else +#endif + + if (!gpio) gpio = irq_to_gpio(bank, d->irq); if (type & ~IRQ_TYPE_SENSE_MASK) diff --git a/drivers/mtd/nand/ams-delta.c b/drivers/mtd/nand/ams-delta.c index 861ca8f..78a524b 100644 --- a/drivers/mtd/nand/ams-delta.c +++ b/drivers/mtd/nand/ams-delta.c @@ -23,12 +23,16 @@ #include <linux/mtd/mtd.h> #include <linux/mtd/nand.h> #include <linux/mtd/partitions.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> + #include <asm/io.h> -#include <mach/hardware.h> #include <asm/sizes.h> -#include <linux/gpio.h> + #include <plat/board-ams-delta.h> +#include <mach/hardware.h> + /* * MTD structure for E3 (Delta) */ diff --git a/arch/arm/plat-omap/include/plat/gpio.h b/include/linux/platform_data/gpio-omap.h similarity index 94% rename from arch/arm/plat-omap/include/plat/gpio.h rename to include/linux/platform_data/gpio-omap.h index 50fb7cc..e8741c2 100644 --- a/arch/arm/plat-omap/include/plat/gpio.h +++ b/include/linux/platform_data/gpio-omap.h @@ -1,6 +1,4 @@ /* - * arch/arm/plat-omap/include/mach/gpio.h - * * OMAP GPIO handling defines and functions * * Copyright (C) 2003-2005 Nokia Corporation @@ -155,6 +153,8 @@ #define OMAP4_GPIO_CLEARDATAOUT 0x0190 #define OMAP4_GPIO_SETDATAOUT 0x0194 +#define OMAP_MAX_GPIO_LINES 192 + #define OMAP_MPUIO(nr) (OMAP_MAX_GPIO_LINES + (nr)) #define OMAP_GPIO_IS_MPUIO(nr) ((nr) >= OMAP_MAX_GPIO_LINES) @@ -213,16 +213,5 @@ extern void omap2_gpio_prepare_for_idle(int off_mode); extern void omap2_gpio_resume_after_idle(void); extern void omap_set_gpio_debounce(int gpio, int enable); extern void omap_set_gpio_debounce_time(int gpio, int enable); -/*-------------------------------------------------------------------------*/ - -/* - * Wrappers for "new style" GPIO calls, using the new infrastructure - * which lets us plug in FPGA, I2C, and other implementations. - * - * The original OMAP-specific calls should eventually be removed. - */ - -#include <linux/errno.h> -#include <asm-generic/gpio.h> #endif diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 0e28322..78e1419 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -34,6 +34,7 @@ #include <mach/hardware.h> #include <mach/gpio.h> #include <plat/mcbsp.h> +#include <linux/platform_data/gpio-omap.h> /* Register descriptions for twl4030 codec part */ #include <linux/mfd/twl4030-audio.h>

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[alsa-devel] Need help with failed driver initialization
by Timur Tabi 11 Sep '12

11 Sep '12

Every 6-12 months it seems, my audio drivers break because of some update to the ASoC. It's that time again. It appears that support for deferred probing isn't working for me. I turned on debug printks (and added a few of my own to my drivers), and here's the log: It looks like everything is registering, but the audio device still doesn't exist. Can anyone give me a clue what's wrong? bus: 'i2c': driver_probe_device: matched device 0-004f with driver cs4270 bus: 'i2c': really_probe: probing driver cs4270 with device 0-004f cs4270 0-004f: found device at i2c address 4F cs4270 0-004f: hardware revision 3 cs4270 0-004f: codec register 0-004f cs4270 0-004f: dai register 0-004f #1 cs4270 0-004f: Mapped DAI cs4270-hifi to CODEC cs4270.0-004f Registered DAI 'cs4270-hifi' Registered codec 'cs4270.0-004f' driver: '0-004f': driver_bound: bound to device 'cs4270' bus: 'i2c': really_probe: bound device 0-004f to driver cs4270 Freescale MPC8610 HPCD ALSA SoC machine driver bus: 'platform': driver_probe_device: matched device e0016000.ssi with driver fsl-ssi-dai bus: 'platform': really_probe: probing driver fsl-ssi-dai with device e0016000.ssi fsl-ssi-dai e0016000.ssi: probing fsl-ssi-dai e0016000.ssi: registering with ASoC fsl-ssi-dai e0016000.ssi: dai register e0016000.ssi Registered DAI 'e0016000.ssi' fsl-ssi-dai e0016000.ssi: AsoC registration complete fsl-ssi-dai e0016000.ssi: platform registration 'snd-soc-mpc8610hpcd' bus: 'platform': driver_probe_device: matched device snd-soc-mpc8610hpcd.0 with driver snd-soc-mpc8610hpcd bus: 'platform': really_probe: probing driver snd-soc-mpc8610hpcd with device snd-soc-mpc8610hpcd.0 bus: 'platform': driver_probe_device: matched device soc-audio with driver soc-audio bus: 'platform': really_probe: probing driver soc-audio with device soc-audio soc-audio soc-audio: ASoC machine snd-soc-mpc8610hpcd should use snd_soc_register_card() soc-audio soc-audio: binding playback at idx 0 soc-audio soc-audio: platform e0021100.dma-channel not registered platform soc-audio: Driver soc-audio requests probe deferral platform soc-audio: Added to deferred list driver: 'snd-soc-mpc8610hpcd.0': driver_bound: bound to device 'snd-soc-mpc8610hpcd' bus: 'platform': really_probe: bound device snd-soc-mpc8610hpcd.0 to driver snd-soc-mpc8610hpcd fsl-ssi-dai e0016000.ssi: probe successful driver: 'e0016000.ssi': driver_bound: bound to device 'fsl-ssi-dai' bus: 'platform': really_probe: bound device e0016000.ssi to driver fsl-ssi-dai bus: 'platform': driver_probe_device: matched device e0021100.dma-channel with driver fsl-pcm-audio bus: 'platform': really_probe: probing driver fsl-pcm-audio with device e0021100.dma-channel fsl-pcm-audio e0021100.dma-channel: probing fsl-pcm-audio e0021100.dma-channel: platform register e0021100.dma-channel Registered platform 'e0021100.dma-channel' fsl-pcm-audio e0021100.dma-channel: probe successful driver: 'e0021100.dma-channel': driver_bound: bound to device 'fsl-pcm-audio' bus: 'platform': really_probe: bound device e0021100.dma-channel to driver fsl-pcm-audio bus: 'platform': driver_probe_device: matched device e0021180.dma-channel with driver fsl-pcm-audio bus: 'platform': really_probe: probing driver fsl-pcm-audio with device e0021180.dma-channel fsl-pcm-audio e0021180.dma-channel: probing fsl-pcm-audio e0021180.dma-channel: platform register e0021180.dma-channel Registered platform 'e0021180.dma-channel' fsl-pcm-audio e0021180.dma-channel: probe successful driver: 'e0021180.dma-channel': driver_bound: bound to device 'fsl-pcm-audio' bus: 'platform': really_probe: bound device e0021180.dma-channel to driver fsl-pcm-audio ... ALSA device list: No soundcards found. platform soc-audio: Retrying from deferred list bus: 'platform': driver_probe_device: matched device soc-audio with driver soc-audio bus: 'platform': really_probe: probing driver soc-audio with device soc-audio soc-audio: probe of soc-audio failed with error -22 When I use speaker-test, I get this: # speaker-test -t sine -c 2 -r 48000 speaker-test 1.0.11rc2 Playback device is default Stream parameters are 48000Hz, S16_BE, 2 channels Sine wave rate is 440.0000Hz ALSA lib confmisc.c:768:(parse_card) cannot find card '0' ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2202:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [asoc:for-next 110/115] sound/soc/soc-core.c:901 soc_bind_dai_link() error: we previously assumed 'dai_link->platform_of_node' could be null (see line 887)
by Fengguang Wu 11 Sep '12

11 Sep '12

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: bffb28a4233d25d37a444785f766e98a041b4413 commit: a6179d2741640a1ec88a1c5b9c2c28bdf45666db [110/115] ASoC: core: display the platform node name if there is no platform name All smatch warnings: + sound/soc/soc-core.c:901 soc_bind_dai_link() error: we previously assumed 'dai_link->platform_of_node' could be null (see line 887) vim +901 sound/soc/soc-core.c 884 885 /* find one from the set of registered platforms */ 886 list_for_each_entry(platform, &platform_list, list) { > 887 if (dai_link->platform_of_node) { 888 if (platform->dev->of_node != 889 dai_link->platform_of_node) 890 continue; 891 } else { 892 if (strcmp(platform->name, platform_name)) 893 continue; 894 } 895 896 rtd->platform = platform; 897 } 898 if (!rtd->platform) { 899 dev_err(card->dev, "platform %s not registered\n", 900 dai_link->platform_name ? : > 901 dai_link->platform_of_node->full_name); 902 return -EPROBE_DEFER; 903 } 904 905 card->num_rtd++; 906 907 return 0; 908 } --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm8900: Convert to devm_kzalloc()
by Mark Brown 11 Sep '12

11 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8900.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 077c962..5f7a78e 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1226,7 +1226,8 @@ static int __devinit wm8900_spi_probe(struct spi_device *spi) struct wm8900_priv *wm8900; int ret; - wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + wm8900 = devm_kzalloc(&spi->dev, sizeof(struct wm8900_priv), + GFP_KERNEL); if (wm8900 == NULL) return -ENOMEM; @@ -1235,15 +1236,13 @@ static int __devinit wm8900_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8900, &wm8900_dai, 1); - if (ret < 0) - kfree(wm8900); + return ret; } static int __devexit wm8900_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1264,7 +1263,8 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, struct wm8900_priv *wm8900; int ret; - wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + wm8900 = devm_kzalloc(&i2c->dev, sizeof(struct wm8900_priv), + GFP_KERNEL); if (wm8900 == NULL) return -ENOMEM; @@ -1273,15 +1273,13 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8900, &wm8900_dai, 1); - if (ret < 0) - kfree(wm8900); + return ret; } static __devexit int wm8900_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: core: display the platform node name if there is no platform name
by Timur Tabi 11 Sep '12

11 Sep '12

A platform can be matched by either a string name or a device tree node pointer, so we can't assume that dai_link->platform_name is valid. Fix an error message in soc_bind_dai_link() to use the device tree node name if the platform name is null. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad65459..d81ef5b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -897,7 +897,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->platform) { dev_err(card->dev, "platform %s not registered\n", - dai_link->platform_name); + dai_link->platform_name ? : + dai_link->platform_of_node->full_name); return -EPROBE_DEFER; } -- 1.7.3.4

3 6

[alsa-devel] [PATCH 1/3] ASoC: Revert "ASoC: mc13783: Provide codec->control_data"
by Fabio Estevam 11 Sep '12

11 Sep '12

Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap) , it is not necessary to provide codec->control_data anymore. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/mc13783.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8def29a..bc95599 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -585,8 +585,6 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - codec->control_data = priv->mc13xxx; - mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: am3517evm: Remove unused cpu_dai from hw_params
by Peter Ujfalusi 11 Sep '12

11 Sep '12

cpu_dai is not in use in this function and just generates warning at compile time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Commit: fca04ae ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing left this line in the code which generates now warning. This patch is on top of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap Regards, Peter sound/soc/omap/am3517evm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index a997988..77a8701 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -41,7 +41,6 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; /* Set the codec system clock for DAC and ADC */ -- 1.7.12

2 1

[alsa-devel] [PATCH] ASoC: tegra: fix maxburst settings in dmaengine code
by Stephen Warren 11 Sep '12

11 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> The I2S controllers are programmed with an "attention" level of 4 DWORDs. This must match the configuration passed to the DMA driver, so that when they burst in data, they don't overflow the available FIFO space. Also, the burst size is relevant to the destination for playback, and source for capture, not vice-versa as originally written. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, I'd like to take this patch through the Tegra tree, since it's a dependency for the series that converts Tegra to dmaengine. Thanks. sound/soc/tegra/tegra_pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 5658bce..8d6900c 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.dst_addr = dmap->addr; - slave_config.src_maxburst = 0; + slave_config.dst_maxburst = 4; } else { slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.src_addr = dmap->addr; - slave_config.dst_maxburst = 0; + slave_config.src_maxburst = 4; } slave_config.slave_id = dmap->req_sel; -- 1.7.0.4

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[alsa-devel] snd-intel-hda: screaming IRQ with 3.5.3 kernel (HDMI)
by Chris Rankin 10 Sep '12

10 Sep '12

Hi, I have recently added a Radeon HD4670 to one of my older PCs, and have noticed that it has IRQ trouble when I try to enable the HDMI audio: irq 17: nobody cared (try booting with the "irqpoll" option) Pid: 0, comm: swapper/0 Not tainted 3.5.3 #2 Call Trace: [<c1063806>] ? __report_bad_irq+0x11/0x94 [<c1063a2d>] ? note_interrupt+0x120/0x19b [<c1062325>] ? handle_irq_event_percpu+0xe8/0xfb [<c1062361>] ? handle_irq_event+0x29/0x40 [<c10640fc>] ? handle_level_irq+0x93/0x93 [<c1064163>] ? handle_fasteoi_irq+0x67/0x90 <IRQ> [<c1003625>] ? do_IRQ+0x2e/0x83 [<c1229829>] ? common_interrupt+0x29/0x30 [<c106007b>] ? audit_update_watch+0x59/0x285 [<c1007479>] ? default_idle+0x23/0x3b [<c1007b38>] ? cpu_idle+0x4b/0x7e [<c1314877>] ? start_kernel+0x2a4/0x2a7 handlers: [<f8c97b75>] azx_interrupt [snd_hda_intel] Disabling IRQ #17 This is frequently followed by KMS trouble too: irq 16: nobody cared (try booting with the "irqpoll" option) Pid: 0, comm: swapper/0 Not tainted 3.5.3 #2 Call Trace: [<c1063806>] ? __report_bad_irq+0x11/0x94 [<c1063a2d>] ? note_interrupt+0x120/0x19b [<c1062325>] ? handle_irq_event_percpu+0xe8/0xfb [<c1062361>] ? handle_irq_event+0x29/0x40 [<c10640fc>] ? handle_level_irq+0x93/0x93 [<c1064163>] ? handle_fasteoi_irq+0x67/0x90 <IRQ> [<c1003625>] ? do_IRQ+0x2e/0x83 [<c1229829>] ? common_interrupt+0x29/0x30 [<c106007b>] ? audit_update_watch+0x59/0x285 [<c1007479>] ? default_idle+0x23/0x3b [<c1007b38>] ? cpu_idle+0x4b/0x7e [<c1314877>] ? start_kernel+0x2a4/0x2a7 handlers: [<f818bfb3>] radeon_driver_irq_handler_kms [radeon] [<f81fe55f>] usb_hcd_irq [usbcore] Disabling IRQ #16 The PCI entries are: 01:00.1 Audio device: ATI Technologies Inc RV710/730 Subsystem: ATI Technologies Inc RV710/730 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (2000ns min), Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 17 Region 0: Memory at ff8ec000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: snd_hda_intel and: 01:00.0 VGA compatible controller: ATI Technologies Inc RV730 Pro AGP [Radeon HD 4600 Series] (prog-if 00 [VGA controller]) Subsystem: ATI Technologies Inc Device 0028 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx+ Latency: 64 (2000ns min), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 16 Region 0: Memory at e0000000 (32-bit, prefetchable) [size=256M] Region 1: I/O ports at ec00 [size=256] Region 2: Memory at ff8f0000 (32-bit, non-prefetchable) [size=64K] Expansion ROM at ff800000 [disabled] [size=128K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] AGP version 3.0 Status: RQ=256 Iso- ArqSz=0 Cal=0 SBA+ ITACoh- GART64- HTrans- 64bit- FW+ AGP3+ Rate=x4,x8 Command: RQ=32 ArqSz=2 Cal=0 SBA+ AGP+ GART64- 64bit- FW- Rate=x8 Kernel driver in use: radeon Does anyone have any suggestions, please? Thanks, Chris

2 5

[alsa-devel] [PATCH 1/3] ASoC: cs4270: Conver to data based control init
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 44a176f..c6e5a73 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -521,14 +521,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) return ret; } - /* Add the non-DAPM controls */ - ret = snd_soc_add_codec_controls(codec, cs4270_snd_controls, - ARRAY_SIZE(cs4270_snd_controls)); - if (ret < 0) { - dev_err(codec->dev, "failed to add controls\n"); - return ret; - } - /* get the power supply regulators */ for (i = 0; i < ARRAY_SIZE(supply_names); i++) cs4270->supplies[i].supply = supply_names[i]; @@ -634,6 +626,9 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .remove = cs4270_remove, .suspend = cs4270_soc_suspend, .resume = cs4270_soc_resume, + + .controls = cs4270_snd_controls, + .num_controls = ARRAY_SIZE(cs4270_snd_controls), .volatile_register = cs4270_reg_is_volatile, .readable_register = cs4270_reg_is_readable, .reg_cache_size = CS4270_LASTREG + 1, -- 1.7.10.4

2 4

Re: [alsa-devel] 3.6-rc Cirrus HDA reconfig oops...
by Takashi Iwai 10 Sep '12

10 Sep '12

At Mon, 10 Sep 2012 23:00:39 +0800, Daniel J Blueman wrote: > > On 10 September 2012 16:29, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 10 Sep 2012 09:46:43 +0200, > > Takashi Iwai wrote: > >> > >> At Sun, 9 Sep 2012 13:02:16 +0800, > >> Daniel J Blueman wrote: > >> > > >> > With a non-specialised Cirrus HDA codec, it's readily possible to > >> > cause a kernel oops as root with 3.6-rc kernels [1]. > >> > > >> > patch_cirrus.c: > >> > > >> > static int cs_init(struct hda_codec *codec) > >> > { > >> > struct cs_spec *spec = codec->spec; > >> > > >> > /* init_verb sequence for C0/C1/C2 errata*/ > >> > snd_hda_sequence_write(codec, cs_errata_init_verbs); > >> > > >> > snd_hda_sequence_write(codec, cs_coef_init_verbs); > >> > > >> > if (spec->gpio_mask) { > >> > snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, > >> > spec->gpio_mask); > >> > > >> > Here, spec is NULL, so loading spec->gpio_mask causes a fatal > >> > pagefault at address 0x180 (ie the offset of gpio_mask in the struct). > >> > I was going to prepare a patch to guard for this, but since spec is > >> > expected in so many places, there is a likely a behavioural issue > >> > preventing a spec struct being generated. > >> > > >> > Let me know for further testing and debug. > >> > >> It's because of calling a leftover callback. > >> The patch below should fix the issue. I'll queue it up to for-linus > >> branch. > > > > Oops, a wrong patch. The corrected one below. > > [...] > > Superb; this fixes the issue nicely. > > Thanks Takashi! OK, thanks for confirmation! Takashi

1 0

Re: [alsa-devel] UAC2 clock selector for AudioBox 1818VSL
by Daniel Mack 10 Sep '12

10 Sep '12

On 07.09.2012 19:57, louis(a)museresearch.com wrote: > Hi Daniel, > Thanks for the quick reply. I am able to set the clock on my Mac, though > I do need to do further tests and make sure that the sync is actually > reliable there. The setting is available from the built-in, standard OS > X audio setup application "Audio MIDI Setup". > > There should be a S/PDIF clock signal going to the digital inputs on the > device. Some of the other controls such as per-channel mute also appear to > work. I've also attached the lsusb output. > > There are some other details that may be relevant, but I didn't want to > make my initial post too long/confusing because these extra details may > end up just being a red herring: Each mixer element name (as seen in > amixer) is prefixed with "AudioBox 1818 VSL Clock Selector" even if it is > not a clock selector. That strange but I believe not related. We should still fix that :) > I am confident though that the control I've been > playing with is the clock selector because it lists the different clock > types as options. I also see this when I plug the device in or load the > ALSA driver: > > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x201, wIndex = 0xa00, type = 4 > ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 > (id 10) > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x200, wIndex = 0xa00, type = 4 > ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 > (id 10) > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x201, wIndex = 0xb00, type = 4 > ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 > (id 11) > ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = > 0x200, wIndex = 0xb00, type = 4 > ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 > (id 11) Do you see any messages when you try to set the clock selector with amixer? If you're running a recent kernel, try the following patch. That will produce log messages for each mixer related control packet that is sent out to the device. Thanks, Daniel

2 1

[alsa-devel] Invitation to connect on LinkedIn
by harsha priya gupta via LinkedIn 10 Sep '12

10 Sep '12

LinkedIn ------------ harsha priya gupta requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - harsha priya Accept invitation from harsha priya gupta http://www.linkedin.com/e/jpxkk3-h6xtnnj8-6b/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of harsha priya gupta http://www.linkedin.com/e/jpxkk3-h6xtnnj8-6b/rso/147246195/LJfd/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-h6xtnnj8-6b/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH 05/22] ASoC: Ux500: Enable MOP500 driver for Device Tree
by Lee Jones 10 Sep '12

10 Sep '12

Here we ensure that the MOP500 audio driver will be probed during a Device Tree boot. We also parse the sound node to link together the codec, dma and the CPU-side Digital Audio Interface. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/mop500.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 31c4d26..6840df7 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -16,6 +16,7 @@ #include <linux/module.h> #include <linux/io.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <sound/soc.h> #include <sound/initval.h> @@ -56,8 +57,35 @@ static struct snd_soc_card mop500_card = { .num_links = ARRAY_SIZE(mop500_dai_links), }; +static int __devinit mop500_of_probe(struct platform_device *pdev, + struct device_node *np) +{ + struct device_node *codec_np, *msp_np[2]; + int i; + + msp_np[0] = of_parse_phandle(np, "stericsson,cpu-dai", 0); + msp_np[1] = of_parse_phandle(np, "stericsson,cpu-dai", 1); + codec_np = of_parse_phandle(np, "stericsson,audio-codec", 0); + + if (!(msp_np[0] && msp_np[1] && codec_np)) { + dev_err(&pdev->dev, "Phandle missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < 2; i++) { + mop500_dai_links[i].cpu_of_node = msp_np[i]; + mop500_dai_links[i].cpu_dai_name = NULL; + mop500_dai_links[i].codec_of_node = codec_np; + mop500_dai_links[i].codec_name = NULL; + } + + snd_soc_of_parse_card_name(&mop500_card, "stericsson,card-name"); + + return 0; +} static int __devinit mop500_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; int ret; pr_debug("%s: Enter.\n", __func__); @@ -66,6 +94,12 @@ static int __devinit mop500_probe(struct platform_device *pdev) mop500_card.dev = &pdev->dev; + if (np) { + ret = mop500_of_probe(pdev, np); + if (ret) + return ret; + } + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", __func__, mop500_card.name); platform_set_drvdata(pdev, &mop500_card); @@ -101,10 +135,16 @@ static int __devexit mop500_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id snd_soc_mop500_match[] = { + { .compatible = "stericsson,snd-soc-mop500", }, + {}, +}; + static struct platform_driver snd_soc_mop500_driver = { .driver = { .owner = THIS_MODULE, .name = "snd-soc-mop500", + .of_match_table = snd_soc_mop500_match, }, .probe = mop500_probe, .remove = __devexit_p(mop500_remove), -- 1.7.9.5

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[alsa-devel] [PATCH 06/22] ASoC: Ux500: Enable ux500 MSP driver for Device Tree
by Lee Jones 10 Sep '12

10 Sep '12

Register both parts of the MSP driver from Device Tree so that they are probed when Device Tree is enabled. Also, as there is platform data involved, we ensure that there is allocated memory to place the configuration into and that the correct information is extracted from the DT binary. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_msp_dai.c | 6 ++++++ sound/soc/ux500/ux500_msp_i2s.c | 22 +++++++++++++++++++--- 2 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 772cb19..0f7dd49 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -833,10 +833,16 @@ static int __devexit ux500_msp_drv_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id ux500_msp_i2c_match[] = { + { .compatible = "stericsson,ux500-msp-i2s", }, + {}, +}; + static struct platform_driver msp_i2s_driver = { .driver = { .name = "ux500-msp-i2s", .owner = THIS_MODULE, + .of_match_table = ux500_msp_i2c_match, }, .probe = ux500_msp_drv_probe, .remove = ux500_msp_drv_remove, diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 2cbfc54..5e0bf8c 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -18,6 +18,7 @@ #include <linux/pinctrl/consumer.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/of.h> #include <mach/hardware.h> #include <mach/msp.h> @@ -692,16 +693,31 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, int ret = 0; struct resource *res = NULL; struct i2s_controller *i2s_cont; + struct device_node *np = pdev->dev.of_node; struct ux500_msp *msp; - dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, - pdev->name, platform_data->id); - *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); msp = *msp_p; if (!msp) return -ENOMEM; + if (np) { + if (!platform_data) { + platform_data = devm_kzalloc(&pdev->dev, + sizeof(struct msp_i2s_platform_data), GFP_KERNEL); + if (!platform_data) + ret = -ENOMEM; + } + } else + if (!platform_data) + ret = -EINVAL; + + if (ret) + goto err_res; + + dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, + pdev->name, platform_data->id); + msp->id = platform_data->id; msp->dev = &pdev->dev; msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: wm8960: Convert to direct regmap API usage
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8960.c | 107 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 84 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 804f411..7cb0d07c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -52,25 +52,72 @@ * We can't read the WM8960 register space when we are * using 2 wire for device control, so we cache them instead. */ -static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0000, 0x0000, - 0x0000, 0x0008, 0x0000, 0x000a, - 0x01c0, 0x0000, 0x00ff, 0x00ff, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x007b, 0x0100, 0x0032, - 0x0000, 0x00c3, 0x00c3, 0x01c0, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0100, 0x0100, 0x0050, 0x0050, - 0x0050, 0x0050, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0040, 0x0000, - 0x0000, 0x0050, 0x0050, 0x0000, - 0x0002, 0x0037, 0x004d, 0x0080, - 0x0008, 0x0031, 0x0026, 0x00e9, +static const struct reg_default wm8960_reg_defaults[] = { + { 0x0, 0x0097 }, + { 0x1, 0x0097 }, + { 0x2, 0x0000 }, + { 0x3, 0x0000 }, + { 0x4, 0x0000 }, + { 0x5, 0x0008 }, + { 0x6, 0x0000 }, + { 0x7, 0x000a }, + { 0x8, 0x01c0 }, + { 0x9, 0x0000 }, + { 0xa, 0x00ff }, + { 0xb, 0x00ff }, + + { 0x10, 0x0000 }, + { 0x11, 0x007b }, + { 0x12, 0x0100 }, + { 0x13, 0x0032 }, + { 0x14, 0x0000 }, + { 0x15, 0x00c3 }, + { 0x16, 0x00c3 }, + { 0x17, 0x01c0 }, + { 0x18, 0x0000 }, + { 0x19, 0x0000 }, + { 0x1a, 0x0000 }, + { 0x1b, 0x0000 }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + + { 0x20, 0x0100 }, + { 0x21, 0x0100 }, + { 0x22, 0x0050 }, + + { 0x25, 0x0050 }, + { 0x26, 0x0000 }, + { 0x27, 0x0000 }, + { 0x28, 0x0000 }, + { 0x29, 0x0000 }, + { 0x2a, 0x0040 }, + { 0x2b, 0x0000 }, + { 0x2c, 0x0000 }, + { 0x2d, 0x0050 }, + { 0x2e, 0x0050 }, + { 0x2f, 0x0000 }, + { 0x30, 0x0002 }, + { 0x31, 0x0037 }, + + { 0x33, 0x0080 }, + { 0x34, 0x0008 }, + { 0x35, 0x0031 }, + { 0x36, 0x0026 }, + { 0x37, 0x00e9 }, }; +static bool wm8960_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8960_RESET: + return true; + default: + return false; + } +} + struct wm8960_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); struct snd_soc_dapm_widget *lout1; @@ -555,6 +602,8 @@ static int wm8960_mute(struct snd_soc_dai *dai, int mute) static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -566,7 +615,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8960->regmap); /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, @@ -667,7 +716,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - snd_soc_cache_sync(codec); + regcache_sync(wm8960->regmap); break; default: break; @@ -915,7 +964,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level = wm8960_set_bias_level_capless; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8960->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -963,9 +1012,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8960 = { .suspend = wm8960_suspend, .resume = wm8960_resume, .set_bias_level = wm8960_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8960_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8960_reg, +}; + +static const struct regmap_config wm8960_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8960_PLL4, + + .reg_defaults = wm8960_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8960_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8960_volatile, }; static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, @@ -979,8 +1037,11 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->regmap = regmap_init_i2c(i2c, &wm8960_regmap); + if (IS_ERR(wm8960->regmap)) + return PTR_ERR(wm8960->regmap); + i2c_set_clientdata(i2c, wm8960); - wm8960->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8904: remove redundant code
by Bo Shen 10 Sep '12

10 Sep '12

The core_intercon is added two times, remove the redundant one Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 0013afe..2b9766d 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1185,8 +1185,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets, ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, core_intercon, - ARRAY_SIZE(core_intercon)); snd_soc_dapm_add_routes(dapm, adc_intercon, ARRAY_SIZE(adc_intercon)); snd_soc_dapm_add_routes(dapm, dac_intercon, -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8741: Move regulator acquisition to I2C/SPI probe()
by Mark Brown 10 Sep '12

10 Sep '12

Better style as we acquire resources before trying the ASoC card probe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8741.c | 37 ++++++++++++++++++++++--------------- 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 35f3d23..742744b 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -403,17 +403,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); int ret = 0; - int i; - - for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) - wm8741->supplies[i].supply = wm8741_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; - } ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); @@ -450,7 +439,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err: return ret; } @@ -460,7 +448,6 @@ static int wm8741_remove(struct snd_soc_codec *codec) struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); return 0; } @@ -492,13 +479,23 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8741_priv *wm8741; - int ret; + int ret, i; wm8741 = devm_kzalloc(&i2c->dev, sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + i2c_set_clientdata(i2c, wm8741); wm8741->control_type = SND_SOC_I2C; @@ -536,13 +533,23 @@ static struct i2c_driver wm8741_i2c_driver = { static int __devinit wm8741_spi_probe(struct spi_device *spi) { struct wm8741_priv *wm8741; - int ret; + int ret, i; wm8741 = devm_kzalloc(&spi->dev, sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + wm8741->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8741); -- 1.7.10.4

1 1

[alsa-devel] ASoC: Multiplexing CODECs together
by Mike Looijmans 10 Sep '12

10 Sep '12

I'm using multiple TI codecs and an OMAP-L138. So far, I've been able to use a single codec. I've changed the davinci-mcasp.c code to respond to "set_tdm_slot", which I now use to tell it which serializers must be activated for capture and playback. This works fine now, any codec on any serializer port can be used independently now. I want to take the next step and have two codecs "mixed" together, so that two codecs form a 4-channel ADC device. They are all running on the same clocks. My idea so far is to create a "multiplex" codec driver, which exports mixer controls to tell it which real codecs to activate. No problem so far. Now when an application starts capture from the multiplex, it should call the "real" codecs (e.g. set_hw_params). This poses several problems, because the structs passed to the calls contain information that is irrelevant for the codec(s) that must actually do the work. Creating a multiplex codec, as opposed to say a dai, made sense to me because an application can "select" it. So my question is, Am I on the right track here? Or should I forget about the codec multiplex driver, because I have to write something completely? Or is this not the correct list to ask this kind of questions? In which case I'd be happy to learn where I might get some tips? Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) - (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. Ieder gebruik van de inhoud van het e-mail bericht, waaronder de daarbij behorende bijlagen, door een ander dan de geadresseerde is onrechtmatig jegens ons dan wel de eventueel in het e-mail bericht of de bijlagen voorkomende andere personen. TOPIC Embedded Systems is niet aansprakelijk voor enigerlei schade voortvloeiend uit het gebruik en/of acceptatie van dit e-mail bericht of de daarbij behorende bijlagen. The contents of this message, as well as any enclosures, are addressed personally to, and thus solely intended for the addressee. They may contain information regarding a third party. A recipient who is neither the addressee, nor empowered to receive this message on behalf of the addressee, is kindly requested to immediately inform the sender of receipt, and to destroy the message and the enclosures. Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

1 0

[alsa-devel] ASoC: fsi: tidyup patches
by Kuninori Morimoto 10 Sep '12

10 Sep '12

Hi Mark These are tidyup patches for fsi.c Kuninori Morimoto (3): ASoC: fsi: fixup pm_runtime_disable() timing on fsi_probe() ASoC: fsi: tidyup: remove un-necessary operation from fsi_probe() ASoC: fsi: convert to devm_xxx() sound/soc/sh/fsi.c | 28 +++++++++------------------- 1 file changed, 9 insertions(+), 19 deletions(-) Best regards --- Kuninori Morimoto

2 4

[alsa-devel] [PATCH 1/2] ASoC: sta32x: Move regulator acquisition to I2C probe
by Mark Brown 10 Sep '12

10 Sep '12

This is better style as it ensures we don't try to do the ASoC probe without required resources. Also convert to devm_ while we're at it, saving a bit of code, and fix a leak of enable on error. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sta32x.c | 32 +++++++++++++++----------------- 1 file changed, 15 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 51b7313..039597e 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -825,22 +825,11 @@ static int sta32x_probe(struct snd_soc_codec *codec) sta32x->codec = codec; sta32x->pdata = dev_get_platdata(codec->dev); - /* regulators */ - for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) - sta32x->supplies[i].supply = sta32x_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), - sta32x->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; - } - ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; + return ret; } /* Tell ASoC what kind of I/O to use to read the registers. ASoC will @@ -849,7 +838,7 @@ static int sta32x_probe(struct snd_soc_codec *codec) ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - return ret; + goto err; } /* Chip documentation explicitly requires that the reset values @@ -915,9 +904,8 @@ static int sta32x_probe(struct snd_soc_codec *codec) return 0; -err_get: - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); err: + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return ret; } @@ -928,7 +916,6 @@ static int sta32x_remove(struct snd_soc_codec *codec) sta32x_watchdog_stop(sta32x); sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; } @@ -966,13 +953,24 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct sta32x_priv *sta32x; - int ret; + int ret, i; sta32x = devm_kzalloc(&i2c->dev, sizeof(struct sta32x_priv), GFP_KERNEL); if (!sta32x) return -ENOMEM; + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, sta32x); ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); -- 1.7.10.4

2 4

[alsa-devel] [PATCH 1/4] ASoC: wm8523: Convert to devm_kzalloc()
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8523.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 1c3ffb2..af2289e 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -516,7 +516,8 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, struct wm8523_priv *wm8523; int ret; - wm8523 = kzalloc(sizeof(struct wm8523_priv), GFP_KERNEL); + wm8523 = devm_kzalloc(&i2c->dev, sizeof(struct wm8523_priv), + GFP_KERNEL); if (wm8523 == NULL) return -ENOMEM; @@ -525,8 +526,7 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8523, &wm8523_dai, 1); - if (ret < 0) - kfree(wm8523); + return ret; } @@ -534,7 +534,6 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, static __devexit int wm8523_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.10.4

1 3

Re: [alsa-devel] 3.6-rc Cirrus HDA reconfig oops...
by Takashi Iwai 10 Sep '12

10 Sep '12

At Sun, 9 Sep 2012 13:02:16 +0800, Daniel J Blueman wrote: > > With a non-specialised Cirrus HDA codec, it's readily possible to > cause a kernel oops as root with 3.6-rc kernels [1]. > > patch_cirrus.c: > > static int cs_init(struct hda_codec *codec) > { > struct cs_spec *spec = codec->spec; > > /* init_verb sequence for C0/C1/C2 errata*/ > snd_hda_sequence_write(codec, cs_errata_init_verbs); > > snd_hda_sequence_write(codec, cs_coef_init_verbs); > > if (spec->gpio_mask) { > snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, > spec->gpio_mask); > > Here, spec is NULL, so loading spec->gpio_mask causes a fatal > pagefault at address 0x180 (ie the offset of gpio_mask in the struct). > I was going to prepare a patch to guard for this, but since spec is > expected in so many places, there is a likely a behavioural issue > preventing a spec struct being generated. > > Let me know for further testing and debug. It's because of calling a leftover callback. The patch below should fix the issue. I'll queue it up to for-linus branch. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix Oops at codec reset/reconfig snd_hda_codec_reset() calls restore_pincfgs() where the codec is powered up again, which eventually tries to resume and initialize via the callbacks of the codec. However, it's the place just after codec free callback, thus no codec callbacks should be called after that. On a codec like CS4206, it results in Oops due to the access in init callback. This patch fixes the issue by clearing the codec callbacks properly after freeing codec. Reported-by: Daniel J Blueman <daniel(a)quora.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f25c24c..aa4cbca 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2353,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) } if (codec->patch_ops.free) codec->patch_ops.free(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; codec->spec = NULL; -- 1.7.11.5

1 1

[alsa-devel] [RFC update 0/2] dmaengine/ASoC: omap: Enable element mode in cyclic DMA
by Peter Ujfalusi 10 Sep '12

10 Sep '12

Hi Russell, Enable the element mode (thus allowing mono playback and probably unblocking OMAP1, OMAP2420) in OMAP dmaengine and omap-pcm. Janusz: would it be possible for you to test Russell's series plus this on OMAP1 to make sure that we do not broke it? Russell: we should wait for Janusz to test this on OMAP1. Based on the feedback we can plan on how to proceed with the dmaengine for OMAP audio. Regards, Peter --- Peter Ujfalusi (2): dmaengine: omap: Support for element mode in cyclic DMA ASoC: omap-pcm: Do not check DMA sync_mode drivers/dma/omap-dma.c | 5 ++++- sound/soc/omap/omap-pcm.c | 10 ---------- 2 files changed, 4 insertions(+), 11 deletions(-) -- 1.7.12

2 5

[alsa-devel] [PATCH 1/2] ASoC: wm8983: Convert to devm_kzalloc()
by Mark Brown 10 Sep '12

10 Sep '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8983.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 367388f..57c33fc 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1085,7 +1085,7 @@ static int __devinit wm8983_spi_probe(struct spi_device *spi) struct wm8983_priv *wm8983; int ret; - wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL); + wm8983 = devm_kzalloc(&spi->dev, sizeof *wm8983, GFP_KERNEL); if (!wm8983) return -ENOMEM; @@ -1094,15 +1094,12 @@ static int __devinit wm8983_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8983, &wm8983_dai, 1); - if (ret < 0) - kfree(wm8983); return ret; } static int __devexit wm8983_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1123,7 +1120,7 @@ static __devinit int wm8983_i2c_probe(struct i2c_client *i2c, struct wm8983_priv *wm8983; int ret; - wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL); + wm8983 = devm_kzalloc(&i2c->dev, sizeof *wm8983, GFP_KERNEL); if (!wm8983) return -ENOMEM; @@ -1132,15 +1129,13 @@ static __devinit int wm8983_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8983, &wm8983_dai, 1); - if (ret < 0) - kfree(wm8983); + return ret; } static __devexit int wm8983_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.10.4

1 1

[alsa-devel] [PATCH] HDA-Intel: Fix wrong CTLINDEX entries for HBR Passthrough
by Wang Xingchao 10 Sep '12

10 Sep '12

>From Anssi Hannula: "The CTLINDEX=x values in /usr/share/alsa/cards/HDA-Intel.conf are off when you have S/PDIF output in the same card as HDMI, which is often the case on Intel. The workaround is to bump all the CTLINDEX=x values one step up in the HDA-Intel.pcm.hdmi.foo entries, i.e. CTLINDEX=0 => CTLINDEX=1 CTLINDEX=1 => CTLINDEX=2 etc, in the /usr/share/alsa/cards/HDA-Intel.conf." Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- src/conf/cards/HDA-Intel.conf | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/conf/cards/HDA-Intel.conf b/src/conf/cards/HDA-Intel.conf index d4f2667..7d6de8f 100644 --- a/src/conf/cards/HDA-Intel.conf +++ b/src/conf/cards/HDA-Intel.conf @@ -211,7 +211,7 @@ HDA-Intel.pcm.hdmi.0 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=3," - "CTLINDEX=0," + "CTLINDEX=1," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," @@ -234,7 +234,7 @@ HDA-Intel.pcm.hdmi.1 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=7," - "CTLINDEX=1," + "CTLINDEX=2," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," @@ -257,7 +257,7 @@ HDA-Intel.pcm.hdmi.2 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=8," - "CTLINDEX=2," + "CTLINDEX=3," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," @@ -280,7 +280,7 @@ HDA-Intel.pcm.hdmi.3 { "cards.HDA-Intel.pcm.hdmi.common:" "CARD=" $CARD "," "DEVICE=9," - "CTLINDEX=3," + "CTLINDEX=4," "AES0=" $AES0 "," "AES1=" $AES1 "," "AES2=" $AES2 "," -- 1.7.9.5

3 2

[alsa-devel] ALSA release cycle
by David Henningsson 08 Sep '12

08 Sep '12

Hi, At Plumber's we discussed the ALSA release cycle. Our releases recently have been irregular, and the reasoning behind why a release was done at that time, has not been very obvious. IIRC, we kind of leaned towards releasing every six months. I don't remember if there was any consensus about whether to try to align this cycle to something else (e g Gnome, KDE, Fedora, Ubuntu, etc), or not. We also said that we should discuss this on the mailinglist as the ALSA release manager (Jaroslav Kysela) was not present during Plumber's. So this effectively is a mail to kick off that discussion. Any opinions? Also, as a side note (or perhaps proof of the problem!), it seems ALSA 1.0.26 was just released without even a notification on this mailinglist...? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

4 5

Re: [alsa-devel] How to define card specific pcm devices?
by Liam Girdwood 08 Sep '12

08 Sep '12

On Wed, 2012-09-05 at 12:24 +0100, Girdwood, Liam wrote: > On 5 September 2012 06:41, Tanu Kaskinen <tanuk(a)iki.fi> wrote: > On Wed, 2012-08-29 at 09:44 +0800, Raymond Yau wrote: > > strictly speaking your mic is not a true stereo mic and the > channel > > map is [left, null] instead of the normal map [left,right] > of line-in > > or stereo mic > > > I think specifying the channel map like you suggest would be a > good > solution. Supporting it would require some extra work in > PulseAudio, but > I think PulseAudio will anyway have to be extended to query > the channel > map information from UCM. > > Now the problem is that UCM doesn't provide the channel map > information > at all. Should the channel map be provided in the device > "Value" > section? Like this: > > SectionDevice."my-mono-mic" { > > ... > > Value { > # Does this value become redundant if > # the "CaptureChannelMap" value is added? > CaptureChannels "2" > I would expect to see a "1" here unless we cannot open the pcm in mono mode. > # Channel names match the constants in > # Takashi's proposed channel map API > CaptureChannelMap "FL, NA" > } > > } This seems fine and should be a simple patch for UCM too. Pulsaudio could use the map information/preference from UCM (like above) to configure the driver using Takashi's new channel map API. > > Opinions? (Liam added to CC - am I right in that you should be > CC'd in > all UCM related discussions?) Yes, please and Mark too (now on CC). Thanks Liam >

3 3

[alsa-devel] sound/soc/soc-dapm.c:1023:4: error: implicit declaration of function 'regulator_allow_bypass'
by Fengguang Wu 08 Sep '12

08 Sep '12

[add the missing subject line..] On Sat, Sep 08, 2012 at 09:44:50AM +0800, Fengguang Wu wrote: > Hi Mark, > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > head: e8bb382aa74747f23e2903c15e48e9f06de5c60a > commit: 9ad4ba77112a3700129cdb142796a54680baa266 [100/103] ASoC: dapm: Allow regulators to bypass as well as disable when idle > config: x86_64-allmodconfig (attached as .config) > > All related error/warning messages: > > sound/soc/soc-dapm.c: In function 'dapm_regulator_event': > sound/soc/soc-dapm.c:1023:4: error: implicit declaration of function 'regulator_allow_bypass' [-Werror=implicit-function-declaration] > cc1: some warnings being treated as errors > > vim +1023 sound/soc/soc-dapm.c > 1018 { > 1019 int ret; > 1020 > 1021 if (SND_SOC_DAPM_EVENT_ON(event)) { > 1022 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { > > 1023 ret = regulator_allow_bypass(w->regulator, true); > 1024 if (ret != 0) > 1025 dev_warn(w->dapm->dev, > 1026 "Failed to bypass %s: %d\n", > 1027 w->name, ret); > 1028 } > > --- > 0-DAY kernel build testing backend Open Source Technology Centre > Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

1 0

Re: [alsa-devel] UAC2 clock selector for AudioBox 1818VSL
by louis＠museresearch.com 07 Sep '12

07 Sep '12

Hi Daniel, Thanks for the quick reply. I am able to set the clock on my Mac, though I do need to do further tests and make sure that the sync is actually reliable there. The setting is available from the built-in, standard OS X audio setup application "Audio MIDI Setup". There should be a S/PDIF clock signal going to the digital inputs on the device. Some of the other controls such as per-channel mute also appear to work. I've also attached the lsusb output (attachment stripped for the mailing list). There are some other details that may be relevant, but I didn't want to make my initial post too long/confusing because these extra details may end up just being a red herring: Each mixer element name (as seen in amixer) is prefixed with "AudioBox 1818 VSL Clock Selector" even if it is not a clock selector. I am confident though that the control I've been playing with is the clock selector because it lists the different clock types as options. I also see this when I plug the device in or load the ALSA driver: ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 (id 10) ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 ALSA sound/usb/mixer.c:846 10:0: cannot get min/max values for control 2 (id 10) ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xb00, type = 4 ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 (id 11) ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xb00, type = 4 ALSA sound/usb/mixer.c:846 11:0: cannot get min/max values for control 2 (id 11) Amixer reports the control at numid=2 to be a boolean, but because of the naming bug I can't see what the control really is. I'm assuming this isn't relevant, but I'm including it here anyway. Louis > On 06.09.2012 22:20, louis(a)museresearch.com wrote: >> Hi! >> I'm attempting to get the "Clock Selector" mixer control to work with >> the PreSonus AudioBox 1818VSL (a UAC2 device). >> >> The control shows up in amixer as an enumerated value, but I don't seem to >> be able to set it (I've tried amixer, alsactl, and setting it >> programmatically). The possible values are correct (listing Internal Clock, S/PDIF Clock, and ADAT Clock as possible options); It just doesn't >> appear to take my setting, by which I mean amixer doesn't update the settings, I still have the occasional click when the sampling rates match >> over S/PDIF and I attempt to use external sync. No error is returned. >> >> After digging around USB specs and ALSA's usb/mixer.c and being unfamiliar >> with both the USB audio spec and the inner workings of ALSA, I thought it'd be best to ask the list about this: How would I check to see if ALSA >> is asking the device to change clock source at all? > > Hmm. A clock selector is handled like any other control entity on the hardware. Some thoughts: > > - Are you able to control other settings like the volume? > > - It's fairly common for devices to not accept a new clock source setting if they can't lock to the external source within some time. Did you connect anything to the digital inputs? > > - Are you able to control that setting with the native Mac OS X driver? (IOW, without installing a 3rd-party driver) > > Also, please send the output of "lsusb -v" for that device. > > > Daniel >

1 0

[alsa-devel] [PATCH 1/4] ASoC: dapm: Add flags to regulator supplies
by Mark Brown 07 Sep '12

07 Sep '12

This will be used to enable additional control of the regulators. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++-- sound/soc/codecs/ab8500-codec.c | 8 ++++---- sound/soc/codecs/wm2200.c | 4 ++-- sound/soc/codecs/wm5100.c | 6 +++--- sound/soc/codecs/wm5102.c | 12 ++++++------ sound/soc/codecs/wm5110.c | 12 ++++++------ sound/soc/codecs/wm8996.c | 2 +- 7 files changed, 25 insertions(+), 24 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 07e2510..c96bf5a 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -244,10 +244,11 @@ struct device; { .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ .shift = wshift, .invert = winvert, .event = wevent, \ .event_flags = wflags} -#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \ +#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags) \ { .id = snd_soc_dapm_regulator_supply, .name = wname, \ .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ - .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .invert = wflags} /* dapm kcontrol types */ diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b783650..3f46bff 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -390,10 +390,10 @@ static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), /* Regulators */ - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0, 0), /* Power */ SND_SOC_DAPM_SUPPLY("Audio Power", diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 71debd0..efa93dbb 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1117,8 +1117,8 @@ SND_SOC_DAPM_SUPPLY("MICBIAS1", WM2200_MIC_BIAS_CTRL_1, WM2200_MICB1_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM2200_MIC_BIAS_CTRL_2, WM2200_MICB2_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20, 0), SND_SOC_DAPM_INPUT("IN1L"), SND_SOC_DAPM_INPUT("IN1R"), diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 33aa363..f4b327e 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -893,9 +893,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM5100_CLOCKING_3, WM5100_SYSCLK_ENA_SHIFT, 0, SND_SOC_DAPM_SUPPLY("ASYNCCLK", WM5100_CLOCKING_6, WM5100_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), SND_SOC_DAPM_SUPPLY("CP1", WM5100_HP_CHARGE_PUMP_1, WM5100_CP1_ENA_SHIFT, 0, NULL, 0), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 97223e4..3f5da8c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -312,12 +312,12 @@ SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0), SND_SOC_DAPM_SIGGEN("TONE"), SND_SOC_DAPM_SIGGEN("NOISE"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 94cd4f1..e7f8d17 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -359,12 +359,12 @@ SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0), SND_SOC_DAPM_SIGGEN("TONE"), SND_SOC_DAPM_SIGGEN("NOISE"), diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 00f183d..6dcb02c 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -931,7 +931,7 @@ SND_SOC_DAPM_INPUT("IN2RP"), SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), -- 1.7.10.4

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[alsa-devel] [PATCH v3 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 07 Sep '12

07 Sep '12

Hello, Generated on top of: git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/omap Changed since v2: - Added commit message to patch 2 (as Tero requested it) - Fixed patch 6 (dt: Add empty...) since the previous patch caused x86_64 build to fail due to missing struct before device_node. Changes since v1: - Get the MCLK frequencey from twl-core driver (via new API) - hs_extmute_disable_level parameter has been removed - empty of_find_node_by_name() in of.h for !CONFIG_OF builds Mark: the extmute GPIO handling (when it is used) remained in the codec driver for now. I can think more on how to add support for this type of mute control. If you can point me to other codec needing this it would help on designing something which is common enough for other users. As for now I have a separate series to add GPIO controlled output amps (hp/spk). I will send that to review soon. Currently this piece of code does not work when booted with device tree since I have done it on n900 which has the GPIO numbers as defines. I need to revisit the aproach on how to do it in a dynamic way. Introl mail from v1: The following series adds DT support for the twl4030 audio submodule which provides audio codec and vibra functionality. The MFD core driver is probed via DT, it will create the needed child devices based on the provided information in the DT blob. Child drivers (vibra, ASoC codec) will parse the core's node if needed to get the needed parameters for their configuration. In the ASoC codec driver the hs_extmute callback (which was used to toggle a GPIO line) has been removed. The codec driver will receive the GPIO number (if it is needed on the platform). If the series is OK (and no objections from the maintainers), it would be good if this can go via audio. Changed files are well contained within the twl4030-audio stack so I do not expect merge issues later. The series has been tested on BeagleBoard (with the McBSP DT series, and with the upcoming DT audio support for BeagleBoard). Regards, Peter --- Peter Ujfalusi (14): MFD: twl4030-audio: Clean up MODULE_* and platform_driver part MFD: twl4030-audio: Convert to use devm_kzalloc MFD: twl4030-audio: Rearange and clean-up the probe function MFD: twl-core: Add API to query the HFCLK rate MFD: twl4030-audio: Get audio MCLK via twl-core API instead of pdata dt: Add empty of_find_node_by_name() function MFD: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/MFD: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel .../devicetree/bindings/mfd/twl4030-audio.txt | 46 +++++++++ arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - drivers/input/misc/twl4030-vibra.c | 20 +++- drivers/mfd/twl-core.c | 32 ++++++ drivers/mfd/twl4030-audio.c | 107 ++++++++++++++------- include/linux/i2c/twl.h | 3 +- include/linux/of.h | 6 ++ sound/soc/codecs/twl4030.c | 103 ++++++++++++++++---- sound/soc/omap/zoom2.c | 4 - 10 files changed, 261 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt -- 1.7.12

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[alsa-devel] ALSA package release v1.0.26 (user space)
by Jaroslav Kysela 07 Sep '12

07 Sep '12

Hello all, I would like to notify you, that ALSA packages version 1.0.26 are available for download: ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/utils/alsa-utils-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/tools/alsa-tools-1.0.26.1.tar.bz2 ftp://ftp.alsa-project.org/pub/plugins/alsa-plugins-1.0.26.tar.bz2 ftp://ftp.alsa-project.org/pub/pyalsa/pyalsa-1.0.26.tar.bz2 The alsa-firmware and alsa-oss packages have no changes and they were not released this time. The changelog is available at this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.25_v1.0.26 The alsa-driver package will be released in next weeks once the driver repository with the build code and the mirrored kernel code will be reorganized to be more "GIT pull" friendly for those using the whole kernel tree as the development platform. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [alsa-jack-plugin] Feedback to modification proposal
by Merx, Thomas 07 Sep '12

07 Sep '12

Hello all, I would like to get some feedback here from you experts on the alsa-jack-plugin. In my project there is a jack-plugin that provides several jack input ports were different streams from different source applications shall be connectable. The source applications would provide the audio samples to jack via the alsa-jack-plugin. On top of that there is a control layer that has to connect the jack output ports of the alsa-jack-plugin with the input ports of my jack-plugin. The API of the control layer simply uses an ID for the source and an ID for the sink to know which jack output ports have to be connected to which input ports. The challenge I'm now facing is that the full name of the jack ports registered by the alsa-jack-plugin are changing from startup to startup because they include the process id of the playback application. Considering the case that there could be two instances of the same application and each of them providing a playback stream via the alsa-jack-plugin, it is challenging for the control layer to choose the correct stream for connecting it to my jack-plugin. So my main problem is that I cannot associate the IDs used for the control layer with the jack port names automatically created by the alsa-jack-plugin. A possible solution to my problem could be to add an additional entry in the asound.conf section where the jack plugin is defined to provide a name which shall be chosen as a prefix for the full jack port name, instead of the default naming scheme currently chosen. To have it backward compatible it could be implemented in a way that if the entry in the asound.conf is missing the current naming scheme is chosen and only if the new entry is set it will be used for the name to be registered at the jack server. What do you think about that? Does it make sense to have that functionality? Best regards Thomas Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Peter Gleissner, Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

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[alsa-devel] [RFC PATCH 1/11] expose sound device topology information
by Clemens Ladisch 07 Sep '12

07 Sep '12

Hi, these are some tentative patches to allow sound drivers to provide routing information to userspace. (So far, snd-usb-audio has a partial implementation; I haven't yet had time for snd-hda-intel. I haven't even looked at ASoC.) The media controller API <linux/media.h> allows prividing routing information, but its implementation does not quite fit the sound drivers: * it allows reconfiguration of links, but not in a way that would be useful for ALSA selector controls; * it allows drivers to access the entity graph and pipelines, which is not needed; * it is rather heavyweight. Therefore, these patches create their own implementation of the media controller API. In this implementation, all entity and link information is completely static, and more space-efficient. For simplicity, the ioctls hook into the ALSA control device instead of creating a new /dev/media* device. The separate implementation does not allow sharing of one media controller device in the case of combined audio/video devices. However, separate drivers would already be a problem for HDMI outputs (GPU and sound); it appears we might need some mechanism to connect the topologies of multiple media devices. TLVs (for jack entities, and for associating controls with entities) are not yet defined. Example output for my simple UA-1A device (not all information is decoded correctly): $ media-ctl -p -d /dev/snd/controlC5 Opening media device /dev/snd/controlC5 Enumerating entities Found 4 entities Enumerating pads and links Media controller API version 0.0.0 Media device information ------------------------ driver snd-usb-audio model EDIROL UA-1A serial bus info usb-0000:00:12.2-4.3 hw revision 0x101 driver version 0.0.0 Device topology - entity 1: USB Audio (1 pad, 1 link) type Node subtype ALSA device node name /dev/tty (should be hw:5,0,0) pad0: Source -> "OT3":0 [ENABLED,IMMUTABLE] - entity 3: OT3 (1 pad, 1 link) type Unknown subtype Unknown (should be Jack) pad0: Sink <- "USB Audio":0 [ENABLED,IMMUTABLE] - entity 4: IT4 (1 pad, 1 link) type Unknown subtype Unknown pad0: Source -> "USB Audio":0 [ENABLED,IMMUTABLE] - entity 7: USB Audio (1 pad, 1 link) type Node subtype ALSA device node name /dev/tty pad0: Sink <- "IT4":0 [ENABLED,IMMUTABLE] include/linux/media.h | 9 include/sound/core.h | 13 + include/sound/media.h | 116 ++++++++++ include/sound/pcm.h | 6 sound/core/Kconfig | 7 sound/core/Makefile | 1 sound/core/init.c | 4 sound/core/media.c | 425 ++++++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 46 ++++ sound/core/sound.c | 3 sound/pci/hda/hda_intel.c | 2 sound/usb/card.c | 2 sound/usb/mixer.c | 278 ++++++++++++++++++++++++ sound/usb/quirks.c | 4 sound/usb/stream.c | 16 - sound/usb/stream.h | 1 16 files changed, 925 insertions(+), 8 deletions(-) Regards, Clemens

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[alsa-devel] [RFC] Channel mapping API (take 2)
by Takashi Iwai 07 Sep '12

07 Sep '12

Hi, my proposal for channel mapping API seems accepted fairly well through discussions at Plumbers, so I continued to work on it and uploaded the updated version. The kernel part is found in sound-unstable tree topic/tlv-chmap branch git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git and the alsa-lib part is found in github tree topic/chmap branch git://git.github.com/tiwai/alsa-lib.git The updated parts are: - SNDRV_CHMAP_NA was newly added, indicating the channel is not available or silent. - The channel position value is masked in lower 16bit. The upper bits are used for more channel attributes as bit flags. #define SNDRV_CHMAP_POSITION_MASK 0xffff The phase inverted channel has this bit: #define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) And the driver-specific non-standard channel position has this: #define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) - The alsa-lib API functions use snd_pcm_chmap_query_t and snd_pcm_chmap_t instead of ambiguous integer arrays. ================================================================ /** the channel map header */ typedef struct snd_pcm_chmap { unsigned int channels; unsigned int pos[0]; } snd_pcm_chmap_t; /** the header of array items returned from snd_pcm_query_chmaps() */ typedef struct snd_pcm_chmap_query { enum snd_pcm_chmap_type type; snd_pcm_chmap_t map; } snd_pcm_chmap_query_t; snd_pcm_chmap_query_t **snd_pcm_query_chmaps(snd_pcm_t *pcm); void snd_pcm_free_chmaps(snd_pcm_chmap_query_t **maps); snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm); int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); ================================================================ The rest aren't so much changed. Only slight bug fixes. A big remaining question is whether the current kernel-side implementation is OK. We may add some PCM ops pointer instead of the current style using the direct control tlv/read/write override. But the kernel-side API can be changed even later, so I don't worry about it so much for now, as long as we can keep the ABI definition. thanks, Takashi

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[alsa-devel] [PATCH 0/3] HDMI HBR(High Bit Rate) Feature bug fix for Intel's Chips
by Wang Xingchao 07 Sep '12

07 Sep '12

As i have no A/V receiver handy and could not test HBR playback directly, i removed the "RFC" type until the patchset was tested and proved working well. The patchset was tested on Intel chips and get positive feedback. Quote from "Øyvind Kvålsvoll <oyvind(a)kvalsvoll.com>": "TrueHD and DTS-MA work fine from XBMC, 2-ch playback also works fine from XBMC. Channels are all mapped to the right speaker, all 7.1." Quote from alanwww1: "I tested the patches. They work perfectly. I tested it with both DTS Master Audio and Dolby Tru HD streams. Also tested with speaker-test. Channel mapping was right." From: http://forum.xbmc.org/showthread.php?tid=128298&pid=1184574#pid1184574 Maybe there's still potential bug and i will keep on track that. The idea of this patch comes from Anssi, Big credit to him at first! Thanks the guys from XBMC forum which help test the patches. For people interested to test the patch, please remember to apply another change in alsa-lib side, you can refer to the patch detail from XBMC forum: http://forum.xbmc.org/showthread.php?tid=128298&pid=1178776#pid1178776 I will send that patch to alsa-lib mail-list later. Changelog: * do not touch channel_allocation priority order, use new api to get correct channels number * use spdif->status to check audio type, pcm or non-pcm * create new non_pcm_mapping[] for non_pcm audio in seperate block * add non_pcm hdmi_spec_per_cvt to remember previous audio type, used to determine whether need a channel mapping update during the pcm/non-pcm stream switch with same channel number. Wang Xingchao (3): ALSA: HDMI - Fix channel_allocation array wrong order ALSA: HDMI - Enable HBR feature on Intel chips ALSA: HDMI - Setup channel mapping for non_pcm audio sound/pci/hda/patch_hdmi.c | 56 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 51 insertions(+), 5 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] Should codec suspend when no paths are marked to be active across a suspend action?
by Mark Brown 07 Sep '12

07 Sep '12

On Thu, Sep 06, 2012 at 12:55:00PM -0700, Taylor Hutt wrote: > Shouldn't the codec be suspended if there are no paths set to remain > active across the suspend? Yes. If you're using an analogue bypass path there's a very recent commit in -next from me which should help, previously we weren't ever really suspending bypass paths at all (even before the changes to keep some paths active) though we might end up doing so by accident. Nobody noticed since essentially nobody uses analogue bypass any more.

1 0

[alsa-devel] UAC2 clock selector for AudioBox 1818VSL
by louis＠museresearch.com 06 Sep '12

06 Sep '12

Hi! I'm attempting to get the "Clock Selector" mixer control to work with the PreSonus AudioBox 1818VSL (a UAC2 device). The control shows up in amixer as an enumerated value, but I don't seem to be able to set it (I've tried amixer, alsactl, and setting it programmatically). The possible values are correct (listing Internal Clock, S/PDIF Clock, and ADAT Clock as possible options); It just doesn't appear to take my setting, by which I mean amixer doesn't update the settings, I still have the occasional click when the sampling rates match over S/PDIF and I attempt to use external sync. No error is returned. After digging around USB specs and ALSA's usb/mixer.c and being unfamiliar with both the USB audio spec and the inner workings of ALSA, I thought it'd be best to ask the list about this: How would I check to see if ALSA is asking the device to change clock source at all? -Louis

2 1

Re: [alsa-devel] [PATCH 6/9] ARM: OMAP: Move gpio.h to include/linux/platform_data
by Tony Lindgren 06 Sep '12

06 Sep '12

* Mohammed, Afzal <afzal(a)ti.com> [120903 03:15]: > Hi, > > On Fri, Aug 31, 2012 at 06:22:34, Tony Lindgren wrote: > > This way we can remove includes of plat/gpio.h which won't work > > with the single zImage support. > > > > Note that we also remove the cpu_class_is_omap2() check > > in gpio-omap.c as the drivers should not call it as we need to > > make it local to arch/arm/mach-omap2 for single zImage support. > > > > While at it, arrange the related includes in the standard way. > > > > Cc: Grant Likely <grant.likely(a)secretlab.ca> > > Cc: Linus Walleij <linus.walleij(a)linaro.org> > > Cc: linux-mtd(a)lists.infradead.org > > Cc: alsa-devel(a)alsa-project.org > > Acked-by: Tony Lindgren <tony(a)atomide.com> > > I think there is only one Tony ;) Heh thanks I'll tell myself to replace it with Signed-off-by :) Tony

1 0

Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 06 Sep '12

06 Sep '12

At Thu, 6 Sep 2012 11:43:48 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 10:21 +0200, Takashi Iwai wrote: > > At Thu, 06 Sep 2012 09:35:26 +0200, > > Takashi Iwai wrote: > > > > In short, a patch like below may fix the issue (note: completely > > untested!) > > No it doesn't, unfortunately... OK, I start tracking down the problem a bit more deeply now. The issue happens when the first two URBs are passed to retire_playback_urb(). These are URBs filled before start_endpoints() are set, so they contain actually zero size. Even though these are a sort of dummy packets, the driver still tries to check with the queued delay account, and gives bogus errors. So, essentially the messages are harmless and nothing to worry too much, but surely it doesn't look sexy. The patch below should fix the problem. Please give it a try. thanks, Takashi === From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: usb-audio: Fix bogus error messages for delay accounting The recent fix for the missing fine delayed time adjustment gives strange error messages at each start of the playback stream, such as delay: estimated 0, actual 352 delay: estimated 353, actual 705 These come from the sanity check in retire_playback_urb(). Before the stream is activated via start_endpoints(), a few silent packets have been already sent. And at this point the delay account is still in the state as if the new packets are just queued, so the driver gets confused and spews the bogus error messages. For fixing the issue, we just need to check whether the received packet is valid, whether it's zero sized or not. Reported-by: Markus Trippelsdorf <markus(a)trippelsdorf.de> Cc: <stable(a)vger.kernel.org> [v3.5+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index fd5e982..f782ce1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1140,6 +1140,12 @@ static void retire_playback_urb(struct snd_usb_substream *subs, int processed = urb->transfer_buffer_length / stride; int est_delay; + /* ignore the delay accounting when procssed=0 is given, i.e. + * silent payloads are procssed before handling the actual data + */ + if (!processed) + return; + spin_lock_irqsave(&subs->lock, flags); est_delay = snd_usb_pcm_delay(subs, runtime->rate); /* update delay with exact number of samples played */ -- 1.7.11.5

1 0

[alsa-devel] [PATCH v2] ASoC: samsung dma - Don't indicate support for pause/resume.
by Dylan Reid 06 Sep '12

06 Sep '12

The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..b70964e 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | @@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: prtd->state &= ~ST_RUNNING; prtd->params->ops->stop(prtd->params->ch); break; -- 1.7.12.146.g16d26b1

2 1

[alsa-devel] [PATCH Resend2] ASoC: imx-mc13783: use defines instead of numerical address of register
by Gaëtan Carlier 06 Sep '12

06 Sep '12

This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++-------------------- 1 files changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c0..bbfa553 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- 1.7.7.4

2 1

Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 06 Sep '12

06 Sep '12

At Thu, 6 Sep 2012 09:17:57 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 09:08 +0200, Daniel Mack wrote: > > On 06.09.2012 08:53, Markus Trippelsdorf wrote: > > > On 2012.09.06 at 08:48 +0200, Takashi Iwai wrote: > > >> At Thu, 06 Sep 2012 08:33:30 +0200, > > >> Daniel Mack wrote: > > >>> > > >>> On 06.09.2012 08:02, Markus Trippelsdorf wrote: > > >>>> On 2012.09.04 at 16:40 +0200, Takashi Iwai wrote: > > >>>>> ---------------------------------------------------------------- > > >>>>> Sound fixes for 3.6-rc5 > > >>>>> > > >>>>> There are nothing scaring, contains only small fixes for HD-audio and > > >>>>> USB-audio: > > >>>>> - EPSS regression fix and GPIO fix for HD-audio IDT codecs > > >>>>> - A series of USB-audio regression fixes that are found since 3.5 kernel > > >>>>> > > >>>>> ---------------------------------------------------------------- > > >>>>> Daniel Mack (4): > > >>>>> ALSA: snd-usb: Fix URB cancellation at stream start > > >>>>> ALSA: snd-usb: restore delay information > > >>>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > >>>> The commit fbcfbf5f above causes the following lines to be printed > > >>>> whenever I start a new song: > > >>> > > >>> Copied Pierre-Louis Bossart - he wrote the code in 294c4fb8 which this > > >>> patch (fbcfbf5f) brings back now. > > >>> > > >>>> delay: estimated 0, actual 352 > > >>>> delay: estimated 353, actual 705 > > >>>> > > >>>> (44.1 * 8 = 352.8) > > >>>> > > >>>> This happens with an USB-DAC that identifies itself as "C-Media USB > > >>>> Headphone Set". > > >>> > > >>> And you didn't you see these lines with 3.4? > > >> > > >> Maybe the difference of start condition? > > >> > > >> Markus, does the patch below fix anything? > > > > > > Unfortunately no. > > > However reverting the following fixes the problem: > > > > > > commit 245baf983cc39524cce39c24d01b276e6e653c9e > > > Author: Daniel Mack <zonque(a)gmail.com> > > > Date: Thu Aug 30 18:52:30 2012 +0200 > > > > > > ALSA: snd-usb: fix calls to next_packet_size > > > > > > > No, this one certainly fixes a problem and does the right thing by > > restoring the original code. > > > > If you wouldn't state that you didn't see the same effect with 3.4(!), > > before the refactoring done in 3.5, I would believe the device is simply > > slightly off in its feedback rate and the tighter delay code complains > > about it while compensating, just as it did before. > > > > Are there any more than these two lines? And is audio working at all? Is > > it distorted in any way? > > There are only these two lines (printed whenever sound starts). Audio is > working just fine with no distortions. > > I did see similar lines before when the system load was very high > (happend during "make check" when building glibc). > > Here is what Pierre-Louis wrote in November 2011: > > Â»This was supposed to be an informational message, I thought it was only > enabled for debug. Regular users don't really need to know.Â« I guess the problem is that the new endpoint scheme doesn't count the last_delay update unless the stream is triggered. In the old code, retire_playback_urb is always called even before the trigger(START) is set. And, there retire_playback_urb() does nothing but updating the delay information. In the new code, retire_playback_urb is set only at snd_usb_substream_playback_trigger(). Thus at the very first shot, the delay account got confused. Takashi

2 2

[alsa-devel] Multiple speaker jacks (on an IDT codec)
by David Henningsson 06 Sep '12

06 Sep '12

Hi, I'm having a problem with a machine where the parser creates indexed speaker controls. For reference, codecs/idt92hd206-ecs-gf7100pvt shows the same problem. I could use some input on how to fix it in the best way. My first thought was that "[Jack] Speaker" is stupid in itself, and could be fixed like this: diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4f7d2df..8e32ee5 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -171,6 +171,10 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, if (conn == AC_JACK_PORT_FIXED) dev = AC_JACK_SPEAKER; } + else if (dev == AC_JACK_SPEAKER) { + if (conn == AC_JACK_PORT_COMPLEX) + dev = AC_JACK_LINE_OUT; + } switch (dev) { case AC_JACK_LINE_OUT: ...but changing generic code is always a bit scary and regression prone (even though I could not think of a good counterexample where that would happen), so not sure if that's the best idea. Next attempt is to modify the control creating function for IDT: diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9db3056..c16d0f8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3226,9 +3226,12 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, idx = i; break; case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - idx = i; - break; + if (num_outs <= 1) { + name = "Speaker"; + idx = i; + break; + } + /* Fall through in case of multi speaker outs */ default: name = chname[i]; idx = 0; ...since the current behaviour is inconsistent anyway - you get indices for 1 and 3, but "Center"/"LFE" for what should otherwise have been index 2. Another way would to just write a pin fix to turn these speakers into line outs, which is what they really are anyway. What do you think? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 3

[alsa-devel] [PATCH] ASoC: SAMSUNG: Add SND_SOC_DAIFMT_CONT option for snd_soc_set_fmt()
by 박상수 06 Sep '12

06 Sep '12

SND_SOC_DAIFMT_CONT option is now needed because of below commit: ASoC: Allow DAI formats to be specified in the dai_link Otherwise, s3c_pcm_set_fmt() returns -EINVAL. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/smdk_wm8994pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 77ecba9..cb88cbc 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -77,6 +77,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, /* Set the cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_CONT | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) -- 1.7.1

3 6

[alsa-devel] radeon multi-channel lpcm
by Pierre Ossman 06 Sep '12

06 Sep '12

Hi, I figured I'd have a go at finishing up support for multi-channel lcpm over HDMI on radeon cards. I read a post from the AMD guys stating that the GPU portions were already in place, and it was the audio pieces that were missing. Unfortunately I was unable to figure out exactly what it is that remains to be done. I poked around in patch_hdmi.c, and concluded that the generic code is what's being used for modern radeon cards. The card exposes two widgets (and output widget and a pin complex). But both widgets indicate through their capability registers that they only support stereo. I don't know enough about HDA to have any clue where to go from here. Should the capability field be ignored for HDMI? Or does it need some coaxing to allow more than stereo? The ELD stuff isn't properly updated from the radeon driver, but the code seems to suggest it is just for hints, and not critical. Rgds -- -- Pierre Ossman WARNING: This correspondence is being monitored by FRA, a Swedish intelligence agency. Make sure your server uses encryption for SMTP traffic and consider using PGP for end-to-end encryption.

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Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Daniel Mack 06 Sep '12

06 Sep '12

On 06.09.2012 08:53, Markus Trippelsdorf wrote: > On 2012.09.06 at 08:48 +0200, Takashi Iwai wrote: >> At Thu, 06 Sep 2012 08:33:30 +0200, >> Daniel Mack wrote: >>> >>> On 06.09.2012 08:02, Markus Trippelsdorf wrote: >>>> On 2012.09.04 at 16:40 +0200, Takashi Iwai wrote: >>>>> ---------------------------------------------------------------- >>>>> Sound fixes for 3.6-rc5 >>>>> >>>>> There are nothing scaring, contains only small fixes for HD-audio and >>>>> USB-audio: >>>>> - EPSS regression fix and GPIO fix for HD-audio IDT codecs >>>>> - A series of USB-audio regression fixes that are found since 3.5 kernel >>>>> >>>>> ---------------------------------------------------------------- >>>>> Daniel Mack (4): >>>>> ALSA: snd-usb: Fix URB cancellation at stream start >>>>> ALSA: snd-usb: restore delay information >>>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ >>>> The commit fbcfbf5f above causes the following lines to be printed >>>> whenever I start a new song: >>> >>> Copied Pierre-Louis Bossart - he wrote the code in 294c4fb8 which this >>> patch (fbcfbf5f) brings back now. >>> >>>> delay: estimated 0, actual 352 >>>> delay: estimated 353, actual 705 >>>> >>>> (44.1 * 8 = 352.8) >>>> >>>> This happens with an USB-DAC that identifies itself as "C-Media USB >>>> Headphone Set". >>> >>> And you didn't you see these lines with 3.4? >> >> Maybe the difference of start condition? >> >> Markus, does the patch below fix anything? > > Unfortunately no. > However reverting the following fixes the problem: > > commit 245baf983cc39524cce39c24d01b276e6e653c9e > Author: Daniel Mack <zonque(a)gmail.com> > Date: Thu Aug 30 18:52:30 2012 +0200 > > ALSA: snd-usb: fix calls to next_packet_size > No, this one certainly fixes a problem and does the right thing by restoring the original code. If you wouldn't state that you didn't see the same effect with 3.4(!), before the refactoring done in 3.5, I would believe the device is simply slightly off in its feedback rate and the tighter delay code complains about it while compensating, just as it did before. Are there any more than these two lines? And is audio working at all? Is it distorted in any way? Pierre-Louis, could you check whether I did the right thing when I ported over your delay bits to the new endpoint logic? Maybe I'm missing something here, but I currently don't see it. Thanks, Daniel

1 0

Re: [alsa-devel] [PATCH 2/4] ARM: OMAP: Split plat/hardware.h, introduce local hardware.h and soc.h for omap2+
by Mark Brown 06 Sep '12

06 Sep '12

On Fri, Aug 31, 2012 at 05:26:22PM -0700, Tony Lindgren wrote: > As the plat and mach includes need to disappear for single zImage work, > we need to remove plat/hardware.h. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> There's a topic/omap branch in ASoC which you can pull in to help reduce conflicts, or I could merge this there if that's sensible.

2 2

[alsa-devel] [PATCH Resend] ASoC: imx-mc13783: use defines instead of numerical address of register
by Gaëtan Carlier 06 Sep '12

06 Sep '12

Forgot to cc Mark Brown, sorry. This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++-------------------- 1 files changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c0..bbfa553 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- 1.7.7.4

2 1

Re: [alsa-devel] (no subject)
by Daniel Mack 06 Sep '12

06 Sep '12

On 06.09.2012 08:02, Markus Trippelsdorf wrote: > On 2012.09.04 at 16:40 +0200, Takashi Iwai wrote: >> ---------------------------------------------------------------- >> Sound fixes for 3.6-rc5 >> >> There are nothing scaring, contains only small fixes for HD-audio and >> USB-audio: >> - EPSS regression fix and GPIO fix for HD-audio IDT codecs >> - A series of USB-audio regression fixes that are found since 3.5 kernel >> >> ---------------------------------------------------------------- >> Daniel Mack (4): >> ALSA: snd-usb: Fix URB cancellation at stream start >> ALSA: snd-usb: restore delay information > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > The commit fbcfbf5f above causes the following lines to be printed > whenever I start a new song: Copied Pierre-Louis Bossart - he wrote the code in 294c4fb8 which this patch (fbcfbf5f) brings back now. > delay: estimated 0, actual 352 > delay: estimated 353, actual 705 > > (44.1 * 8 = 352.8) > > This happens with an USB-DAC that identifies itself as "C-Media USB > Headphone Set". And you didn't you see these lines with 3.4? Daniel

2 1

[alsa-devel] About joining codecs together
by Mike Looijmans 06 Sep '12

06 Sep '12

I have multiple codecs (TLV320AIC3256) connected to the McASP of an OMAP L138. I've adapted the drivers a bit so they work with my 2.6.37 kernel and the custom board (similar to a DA850). Currently, I can only use one codec at a time. The mcasp is set up in the board config, and there it determines which AXR lines are active. Each codec is connected to another line. They run on the same clock lines. Question: - How can I forward information from the codec to the mcasp driver, to inform which AXR channel to enable? There seems to be no information exchange between them. When codec 1 is running, the mcasp driver must record from AXR1, and when codec2 is running, AXR2 and so on. What I ultimately want to do is use two (or more) codecs for multichannel recording, so that I can use two codecs for a 4-channel recording. For the McASP side, this would simply mean enabling both AXR inputs (DAVINCI_MCASP_PDIR_REG) and then starting both codecs. Questions: - How does this work in user space? There are two codecs, which are to be configured independently, but there will be only one audio stream coming out. I was thinking to create a sort of "multiswitch", a codec that by itself doesn't really do anything, but presents itself as a single codec with lots of channels. The actual commands will be delivered to the appropriate "real" codecs, so that you can record from the "switch", which then in turn enables two codecs and sets up the DAI to record both streams. Would that be possible at all? What are the alternatives? Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) - (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. Ieder gebruik van de inhoud van het e-mail bericht, waaronder de daarbij behorende bijlagen, door een ander dan de geadresseerde is onrechtmatig jegens ons dan wel de eventueel in het e-mail bericht of de bijlagen voorkomende andere personen. TOPIC Embedded Systems is niet aansprakelijk voor enigerlei schade voortvloeiend uit het gebruik en/of acceptatie van dit e-mail bericht of de daarbij behorende bijlagen. The contents of this message, as well as any enclosures, are addressed personally to, and thus solely intended for the addressee. They may contain information regarding a third party. A recipient who is neither the addressee, nor empowered to receive this message on behalf of the addressee, is kindly requested to immediately inform the sender of receipt, and to destroy the message and the enclosures. Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

1 0

[alsa-devel] [PATCH 1/1] [ASoC] Fix v2: Revert 'ASoC: imx-ssi: Remove mono support'.
by Gaëtan Carlier 06 Sep '12

06 Sep '12

When patch of Javier Martin is applied on linux-next-20120824 the patch failed because comment line is missing. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/fsl/imx-ssi.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 7074ae6..3c520c4 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,14 +380,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- 1.7.7.4

1 1

[alsa-devel] [PATCH] ASoC: Revert 'ASoC: imx-ssi: Remove mono support'
by Fabio Estevam 06 Sep '12

06 Sep '12

From: Javier Martin <javier.martin(a)vista-silicon.com> Revert 0865a75(ASoC: imx-ssi: Remove mono support). The bug this patch is meant to solve doesn't occur in Visstrim_M10 boards. Furthermore, after applying this patch sound in Visstrim_M10 is played at slower rates. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/fsl/imx-ssi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 7074ae6..3c520c4 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,14 +380,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- 1.7.9.5

2 1

[alsa-devel] [PATCH -next] soc: fix wm0010.c printk format warning
by Randy Dunlap 06 Sep '12

06 Sep '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix printk format warning: sound/soc/codecs/wm0010.c:411:2: warning: format '%d' expects type 'int', but argument 4 has type 'size_t' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20120905.orig/sound/soc/codecs/wm0010.c +++ linux-next-20120905/sound/soc/codecs/wm0010.c @@ -408,7 +408,7 @@ static int wm0010_boot(struct snd_soc_co wm0010->state = WM0010_BOOTROM; spin_unlock_irqrestore(&wm0010->irq_lock, flags); - dev_dbg(codec->dev, "Downloading %d byte stage 2 loader\n", fw->size); + dev_dbg(codec->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);

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[alsa-devel] [PATCH V2] ASoC: Davinci: McASP: add support new McASP IP Variant
by Hebbar, Gururaja 06 Sep '12

06 Sep '12

The OMAP2+ variant of McASP is different from Davinci variant w.r.to some register offset. Changes - Add new MCASP_VERSION_3 to identify new variant. New DT compatible "ti,omap2-mcasp-audio" to identify version 3 controller. - The register offsets are handled depending on the version. Note: DMA parameters (dma fifo offset) are not updated and will be done later. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- Changes in V2: Use "switch" instead of "if" statement to allow future revisions :100644 100644 e6148ec... 374e145... M Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt :100644 100644 79c26aa... d0c5825... M include/linux/platform_data/davinci_asp.h :100644 100644 c3eae1d... 714e51e... M sound/soc/davinci/davinci-mcasp.c .../bindings/sound/davinci-mcasp-audio.txt | 1 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 86 +++++++++++++++++--- 3 files changed, 75 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index e6148ec..374e145 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,6 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) - reg : Should contain McASP registers offset and length - interrupts : Interrupt number for McASP diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 79c26aa..d0c5825 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -87,6 +87,7 @@ struct snd_platform_data { enum { MCASP_VERSION_1 = 0, /* DM646x */ MCASP_VERSION_2, /* DA8xx/OMAPL1x */ + MCASP_VERSION_3, /* TI81xx/AM33xx */ }; enum mcbsp_clk_input_pin { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c3eae1d..714e51e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -111,6 +111,10 @@ #define DAVINCI_MCASP_WFIFOSTS (0x1014) #define DAVINCI_MCASP_RFIFOCTL (0x1018) #define DAVINCI_MCASP_RFIFOSTS (0x101C) +#define MCASP_VER3_WFIFOCTL (0x1000) +#define MCASP_VER3_WFIFOSTS (0x1004) +#define MCASP_VER3_RFIFOCTL (0x1008) +#define MCASP_VER3_RFIFOSTS (0x100C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -384,18 +388,36 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } } mcasp_start_tx(dev); } else { if (dev->rxnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } mcasp_start_rx(dev); } @@ -416,14 +438,31 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + if (dev->txnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_tx(dev); } else { - if (dev->rxnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + if (dev->rxnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_rx(dev); } } @@ -622,20 +661,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + tx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + } } if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { if (dev->rxnumevt * rx_ser > 64) dev->rxnumevt = 1; - - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + rx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + } } } @@ -874,6 +930,10 @@ static const struct of_device_id mcasp_dt_ids[] = { .compatible = "ti,da830-mcasp-audio", .data = (void *)MCASP_VERSION_2, }, + { + .compatible = "ti,omap2-mcasp-audio", + .data = (void *)MCASP_VERSION_3, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); -- 1.7.1

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Re: [alsa-devel] [ASoC] Fix: Revert 'ASoC: imx-ssi: Remove mono support'.
by Mark Brown 06 Sep '12

06 Sep '12

On Tue, Sep 04, 2012 at 11:09:20AM +0200, javier Martin wrote: > could you pick up the fixed patch sent by Gaëtan? It's mangled, can someone resend please (ideally with an appropriate subject line and so on).

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Re: [alsa-devel] [PATCH] ASoC: mc13783: Remove mono support
by Mark Brown 06 Sep '12

06 Sep '12

On Mon, Sep 03, 2012 at 01:04:13PM -0300, Fabio Estevam wrote: > Playing a mono track on a mc13783 codec results in incorrect playback rate. > > Remove mono support so that a mono track can be played correctly. Applied, thanks.

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[alsa-devel] [PATCH] ASoC: Remove unused 'saved_value' field from snd_soc_dapm_widget struct
by Lars-Peter Clausen 06 Sep '12

06 Sep '12

The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove current PGA control handling"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index abe373d..0a15537 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,7 +510,6 @@ struct snd_soc_dapm_widget { /* dapm control */ int reg; /* negative reg = no direct dapm */ unsigned char shift; /* bits to shift */ - unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: imx-mc13783: use defines instead of numerical address of register
by Gaëtan Carlier 06 Sep '12

06 Sep '12

This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++-------------------- 1 files changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c0..bbfa553 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- 1.7.7.4

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Fix compilation error due to leftover code
by Peter Ujfalusi 06 Sep '12

06 Sep '12

Part of commit (which patches sound/soc/omap/mcbsp.c file): 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code since the tree where it has been applied did not had the earlier patch: d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration which changed code around omap_mcbsp_6pin_src_mux(). Because of the missing part from 8fef626 the sound/soc/omap/mcbsp.c does not compile in linux-next. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, This patch has been generated against linux-next as of today. The reason for this is that that is the only place where we have the upstream patch (in 3.6-rc3): d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration and the 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code commit in the same branch. When you applied 8fef626 you mentioned that it did not applied cleanly. The reason was that the branch did not had commit d0db84e applied prior to the series: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-August/054668.html Now I can not find the d0db84e commit in the asoc tree (not in for-3.6, for-3.7 and not in for-next branch) anymore. I don't really know what has happened, but as of now this patch is needed to fix the OMAP audio in linux-next. I have also prepared a branch with the OMAP patches [1]: topic/omap rebased on top of today's mainline and fixed up the 8fef626 to it's original content. But I think it is already late to revert the OMAP McBSP patches from for-next and from for-3.7 branches... [1] git://gitorious.org/omap-audio/linux-audio.git peter/topic/for-mark/omap-3.7 CC-ing linux-next as well. Regards, Peter sound/soc/omap/mcbsp.c | 31 ------------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 935ccf6..bc06175 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -762,37 +762,6 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) } -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) -{ - const char *signal, *src; - - if (!mcbsp->pdata->mux_signal) - return -EINVAL; - - switch (mux) { - case CLKR_SRC_CLKR: - signal = "clkr"; - src = "clkr"; - break; - case CLKR_SRC_CLKX: - signal = "clkr"; - src = "clkx"; - break; - case FSR_SRC_FSR: - signal = "fsr"; - src = "fsr"; - break; - case FSR_SRC_FSX: - signal = "fsr"; - src = "fsx"; - break; - default: - return -EINVAL; - } - - return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); -} - #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) #define THRESHOLD_PROP_BUILDER(prop) \ -- 1.7.12

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[alsa-devel] [PATCH] ASoC: arizona: Fix typo in 44.1kHz rates
by Mark Brown 06 Sep '12

06 Sep '12

From: Heather Lomond <hlomond(a)opensource.wolfsonmicro.com> Signed-off-by: Heather Lomond <hlomond(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9caca..1cf7a32 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = { 940800, 1411200, 1881600, - 2882400, + 2822400, 3763200, 5644800, 7526400, -- 1.7.10.4

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[alsa-devel] [asoc:for-3.7 95/95] sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT
by Fengguang Wu 06 Sep '12

06 Sep '12

[add more CC on request of Dimitris] Hi Dimitris, Mark, FYI, there are new coccinelle warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 head: e3523e01869da20fdd12ffd19ae1df7bf492650e commit: e3523e01869da20fdd12ffd19ae1df7bf492650e [95/95] ASoC: wm0010: Add initial wm0010 DSP driver All coccinelle warnings: + sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT -- + sound/soc/codecs/wm0010.c:660:1-7: preceding lock on line 359 vim +850 sound/soc/codecs/wm0010.c 847 trigger = IRQF_TRIGGER_FALLING; 848 trigger |= IRQF_ONESHOT; 849 > 850 ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, 851 "wm0010", wm0010); 852 if (ret) 853 dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", Please consider folding the attached diff :-) --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

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Re: [alsa-devel] [PATCH] ASoC: wm0010: Fix warning, use format %zu for type size_t
by Mark Brown 06 Sep '12

06 Sep '12

On Wed, Sep 05, 2012 at 10:22:24PM +0200, Emil Goode wrote: > Fix warning by using format specifier %zu for type size_t > > Sparse warning: > sound/soc/codecs/wm0010.c:411:2: warning: > format ‘%d’ expects argument of type ‘int’, > but argument 4 has type ‘size_t’ [-Wformat] Applied, thanks.

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[alsa-devel] [PATCH V2] ASoC: tegra: move platform data header
by Stephen Warren 06 Sep '12

06 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: * Remove stale filename from comment. * Add include guards --- .../sound/tegra_wm8903.h | 7 +++++-- sound/soc/tegra/tegra_wm8903.c | 3 +-- 2 files changed, 6 insertions(+), 4 deletions(-) rename arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h => include/sound/tegra_wm8903.h (89%) diff --git a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h b/include/sound/tegra_wm8903.h similarity index 89% rename from arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h rename to include/sound/tegra_wm8903.h index 9d29334..57b202e 100644 --- a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h +++ b/include/sound/tegra_wm8903.h @@ -1,6 +1,4 @@ /* - * arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h - * * Copyright 2011 NVIDIA, Inc. * * This software is licensed under the terms of the GNU General Public @@ -14,6 +12,9 @@ * */ +#ifndef __SOUND_TEGRA_WM38903_H +#define __SOUND_TEGRA_WM38903_H + struct tegra_wm8903_platform_data { int gpio_spkr_en; int gpio_hp_det; @@ -21,3 +22,5 @@ struct tegra_wm8903_platform_data { int gpio_int_mic_en; int gpio_ext_mic_en; }; + +#endif diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index d4f14e4..cee13b7 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,13 +34,12 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <mach/tegra_wm8903_pdata.h> - #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tegra_wm8903.h> #include "../codecs/wm8903.h" -- 1.7.0.4

2 1

[alsa-devel] [PATCH] hda-emu: Add support for new pm_notify callback format
by David Henningsson 05 Sep '12

05 Sep '12

The format of the pm_notify callback has recently changed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 24 ++++++++++++++++++++++++ hda-emu.c | 12 ++++++++++++ 2 files changed, 36 insertions(+) diff --git a/configure.ac b/configure.ac index 18118f3..43db88f 100644 --- a/configure.ac +++ b/configure.ac @@ -222,6 +222,30 @@ if test "$OLD_HDA_CMD" = "1"; then AC_DEFINE(OLD_HDA_CMD) fi +AC_MSG_CHECKING(for new pm_notify callback) +ac_save_CFLAGS="$CFLAGS" +CFLAGS="$CFLAGS -Iinclude" +HAVE_NEW_PM_NOTIFY= +AC_TRY_COMPILE([ +#include "hda-types.h" +#include "hda-log.h" +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include "kernel/hda_codec.h" +],[ + extern struct hda_bus_ops *ops; + ops->pm_notify(NULL, 0); +], + [AC_MSG_RESULT(yes); HAVE_NEW_PM_NOTIFY="1"], + [AC_MSG_RESULT(no)], + [AC_MSG_RESULT(unknown)] +) +CFLAGS=$ac_save_CFLAGS +if test "$HAVE_NEW_PM_NOTIFY" = "1"; then + AC_DEFINE(HAVE_NEW_PM_NOTIFY) +fi + AC_MSG_CHECKING(for attach_pcm bus ops) ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS -Iinclude" diff --git a/hda-emu.c b/hda-emu.c index b4bd8a9..2a2b3d1 100644 --- a/hda-emu.c +++ b/hda-emu.c @@ -776,6 +776,13 @@ static int azx_pcm_create(struct hda_codec *codec) * power management */ +#ifdef HAVE_NEW_PM_NOTIFY +static void new_pm_notify(struct hda_bus *bus, bool power_up) +{ + hda_log(HDA_LOG_INFO, "PM-Notified, power_up = %d\n", power_up); +} +#else + static void pm_notify(struct hda_bus *bus) { hda_log(HDA_LOG_INFO, "PM-Notified\n"); @@ -787,6 +794,7 @@ static void old_pm_notify(struct hda_codec *codec) pm_notify(codec->bus); } #endif +#endif /* HAVE_NEW_PM_NOTIFY */ /* @@ -1112,7 +1120,11 @@ int main(int argc, char **argv) #ifdef HAVE_HDA_ATTACH_PCM temp.ops.attach_pcm = attach_pcm; #endif +#ifdef HAVE_NEW_PM_NOTIFY + temp.ops.pm_notify = new_pm_notify; +#else temp.ops.pm_notify = pm_notify; +#endif #endif /* OLD_HDA_CMD */ gather_codec_hooks(); -- 1.7.9.5

2 1

Re: [alsa-devel] ALSA: core: add support for compress_offload
by Dan Carpenter 05 Sep '12

05 Sep '12

Hello Vinod Koul, The patch b21c60a4edd2: "ALSA: core: add support for compress_offload" from Dec 23, 2011, leads to the following warning: sound/core/compress_offload.c:85 snd_compr_open() warn: bitwise AND condition is false here 83 if (f->f_flags & O_WRONLY) 84 dirn = SND_COMPRESS_PLAYBACK; 85 else if (f->f_flags & O_RDONLY) 86 dirn = SND_COMPRESS_CAPTURE; 87 else { 88 pr_err("invalid direction\n"); 89 return -EINVAL; 90 } O_RDONLY is 0 so we never do compressed capture. Perhaps the tests should be: if ((file->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; else if (file->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; else return -EINVAL; Also it seems like a bad thing to print an error here, because can the user flood dmesg? regards, dan carpenter

2 1

[alsa-devel] [patch] ASoC: wm0010: unlock on error path
by Dan Carpenter 05 Sep '12

05 Sep '12

We're holding the wm0010->lock mutex when we goto err_core. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This driver is only in linux-next. diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 5f99148..1c052d6 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -663,7 +663,9 @@ abort: wm0010_halt(codec); mutex_unlock(&wm0010->lock); return ret; + err_core: + mutex_unlock(&wm0010->lock); regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), wm0010->core_supplies); err:

2 1

Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Takashi Iwai 05 Sep '12

05 Sep '12

At Wed, 5 Sep 2012 15:00:15 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Applied this one, too. Thanks. Takashi > --- > sound/pci/emu10k1/memory.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c > index 0a43662..1dce0a39 100644 > --- a/sound/pci/emu10k1/memory.c > +++ b/sound/pci/emu10k1/memory.c > @@ -263,8 +263,8 @@ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *b > spin_lock_irqsave(&emu->memblk_lock, flags); > if (blk->mapped_page >= 0) { > /* update order link */ > - list_del(&blk->mapped_order_link); > - list_add_tail(&blk->mapped_order_link, &emu->mapped_order_link_head); > + list_move_tail(&blk->mapped_order_link, > + &emu->mapped_order_link_head); > spin_unlock_irqrestore(&emu->memblk_lock, flags); > return 0; > } > >

1 0

Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Takashi Iwai 05 Sep '12

05 Sep '12

At Wed, 5 Sep 2012 14:33:21 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied. Takashi > --- > sound/drivers/opl4/opl4_synth.c | 9 +++------ > 1 file changed, 3 insertions(+), 6 deletions(-) > > diff --git a/sound/drivers/opl4/opl4_synth.c b/sound/drivers/opl4/opl4_synth.c > index 49b9e24..4b91adc 100644 > --- a/sound/drivers/opl4/opl4_synth.c > +++ b/sound/drivers/opl4/opl4_synth.c > @@ -504,8 +504,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha > spin_lock_irqsave(&opl4->reg_lock, flags); > for (i = 0; i < voices; i++) { > voice[i] = snd_opl4_get_voice(opl4); > - list_del(&voice[i]->list); > - list_add_tail(&voice[i]->list, &opl4->on_voices); > + list_move_tail(&voice[i]->list, &opl4->on_voices); > voice[i]->chan = chan; > voice[i]->note = note; > voice[i]->velocity = vel & 0x7f; > @@ -555,8 +554,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha > > static void snd_opl4_voice_off(struct snd_opl4 *opl4, struct opl4_voice *voice) > { > - list_del(&voice->list); > - list_add_tail(&voice->list, &opl4->off_voices); > + list_move_tail(&voice->list, &opl4->off_voices); > > voice->reg_misc &= ~OPL4_KEY_ON_BIT; > snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); > @@ -571,8 +569,7 @@ void snd_opl4_note_off(void *private_data, int note, int vel, struct snd_midi_ch > > static void snd_opl4_terminate_voice(struct snd_opl4 *opl4, struct opl4_voice *voice) > { > - list_del(&voice->list); > - list_add_tail(&voice->list, &opl4->off_voices); > + list_move_tail(&voice->list, &opl4->off_voices); > > voice->reg_misc = (voice->reg_misc & ~OPL4_KEY_ON_BIT) | OPL4_DAMP_BIT; > snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); >

1 0

Re: [alsa-devel] [PATCH] ALSA: use list_move_tail instead of list_del/list_add_tail
by Clemens Ladisch 05 Sep '12

05 Sep '12

Wei Yongjun wrote: > Using list_move_tail() instead of list_del() + list_add_tail(). > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Acked-by: Clemens Ladisch <clemens(a)ladisch.de>

1 0

[alsa-devel] [PATCH 0/3] HDMI HBR(High Bit Rate) Feature bug fix for Intel's Chips
by Wang Xingchao 05 Sep '12

05 Sep '12

Please Note This patchset has not been tested yet on any Hardware platform. i could only confirm there's no compile error. :) I test previous single patch on the Yamaha A/V receiver which works well, the test results was also confirmed by some guys from XBMC forum. The idea of this patch comes from Anssi, Big credit to him at first! Any comments are appreciated! Changelog: * do not touch channel_allocation priority order, use new api to get correct channels number * use spdif->status to check audio type, pcm or non-pcm * create new non_pcm_mapping[] for non_pcm audio in seperate block * add non_pcm hdmi_spec_per_cvt to remember previous audio type, used to determine whether need a channel mapping update during the pcm/non-pcm stream switch with same channel number. Wang Xingchao (3): ALSA: hdmi - fix channel_allocation array wrong order ALSA: HDMI - Enable HBR feature on Intel chips ALSA: HDMI - Setup channel mapping for non_pcm audio sound/pci/hda/patch_hdmi.c | 56 ++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 51 insertions(+), 5 deletions(-) -- 1.7.9.5

3 5

[alsa-devel] [PATCH] ALSA: HDMI - Enable HBR feature on Intel chips
by Wang Xingchao 05 Sep '12

05 Sep '12

HDMI channel remapping apparently effects HBR packets on Intel's chips. For compressed non-PCM audio, use "straight-through" channel mapping. For uncompressed multi-channel pcm audio, use normal mapping. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/patch_hdmi.c | 50 +++++++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d9439c5..fda89e3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -231,6 +231,11 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, /* Dolby Surround */ { .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, /* surround40 */ { .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, /* surround41 */ @@ -239,22 +244,15 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, /* surround51 */ { .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, { .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, { .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, { .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, { .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, { .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, { .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, @@ -537,22 +535,36 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, + hda_nid_t cvt_nid, int ca) { int i; int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); + unsigned short val; + int tmp_mapping[8]; + bool non_pcm = false; + + val = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + non_pcm = !!(val & AC_DIG1_NONAUDIO); + + if (hdmi_channel_mapping[ca][1] == 0 || non_pcm) { + for (i = 0; i < channel_allocations[ca].channels; i++) { + if (non_pcm) + tmp_mapping[i] = i | (i << 4); + else + hdmi_channel_mapping[ca][i] = i | (i << 4); + } for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + if (non_pcm) + tmp_mapping[i] = 0xf | (i << 4); + else + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); } for (i = 0; i < 8; i++) { err = snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); + non_pcm ? tmp_mapping[i] : hdmi_channel_mapping[ca][i]); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -686,7 +698,7 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, } static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - struct snd_pcm_substream *substream) + hda_nid_t cvt_nid, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; @@ -731,13 +743,13 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * sizeof(*dp_ai) to avoid partial match/update problems when * the user switches between HDMI/DP monitors. */ + hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, ca); if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, sizeof(ai))) { snd_printdd("hdmi_setup_audio_infoframe: " "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -1151,7 +1163,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, substream); + hdmi_setup_audio_infoframe(codec, pin_idx, cvt_nid, substream); pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- 1.7.9.5

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Re: [alsa-devel] How to define card specific pcm devices?
by Raymond Yau 05 Sep '12

05 Sep '12

2012-8-17 下午10:51 於 "Tanu Kaskinen" <tanu.kaskinen(a)digia.com> 寫道： > > Ping? > > If card specific pcm definitions are not possible, would it be > acceptable to have a generic "front_left:CARD" or "first_channel:CARD" > pcm, which would transform "front:CARD" or "hw:CARD" into a mono device? As pulseaudio server use SND_PCM_NO_AUTO_CHANNEL which diable channel conversion, this left channel device only work with other alsa applications except pulseaudio when your sound card does not support mono. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ff4af902cf4ac0… the solution is to change the mixing in pulseaudio by using left channel for this mic and those stereo mic with inverted phase , and average for mono playback device . stereo mic and line-in how about the 4 channel mic? > > -- Tanu > > > On Fri, 2012-08-10 at 14:25 +0300, Tanu Kaskinen wrote: > > Hi, > > > > I'm writing UCM configuration for RX-51 (Nokia N900), and I have a > > problem: "hw:RX51" can't be opened in mono mode, but there's only one > > integrated mono microphone, and its audio is routed to the left input > > channel. Applications will get only silence in the right channel when > > they use "hw:RX51". > > > > The best solution that I have come up is to define a pcm that converts > > "hw:RX51" into a mono device. The problem is that I don't know how > > exactly to do that. I guess the route plugin can be used for that: > > > > pcm.rx51_mono_from_left { > > type route > > slave.pcm hw:RX51 > > slave.channels 2 > > ttable.0.0 1 > > } > > > > I tried putting that definition in /usr/share/alsa/cards/RX-51.conf (and > > also RX51.conf, since it's not entirely clear to me when the dash should > > be omitted). I have this in the UCM configuration: > > > > Value { > > CapturePCM "rx51_mono_from_left" > > CaptureChannels "1" > > } > > > > This is printed to the PulseAudio log: > > > > (alsa-lib)pcm.c: Unknown PCM rx51_mono_from_left > > > > So the rx51_mono_from_left definition is not loaded. I'm not terribly > > surprised: I guess the card configuration files are loaded only when > > necessary, and when PulseAudio tries to open "rx51_mono_from_left", > > alsa-lib can't really know that it's somehow related to the RX-51 card. > > > > How should I solve this? > > > >

2 4

[alsa-devel] [PATCH] ASoC: tegra: move platform data header
by Stephen Warren 05 Sep '12

05 Sep '12

From: Stephen Warren <swarren(a)nvidia.com> Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- .../sound/tegra_wm8903.h | 0 sound/soc/tegra/tegra_wm8903.c | 3 +-- 2 files changed, 1 insertions(+), 2 deletions(-) rename arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h => include/sound/tegra_wm8903.h (100%) diff --git a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h b/include/sound/tegra_wm8903.h similarity index 100% rename from arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h rename to include/sound/tegra_wm8903.h diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index d4f14e4..cee13b7 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,13 +34,12 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <mach/tegra_wm8903_pdata.h> - #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tegra_wm8903.h> #include "../codecs/wm8903.h" -- 1.7.0.4

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[alsa-devel] [RFC 0/3] Initial stab at converting OMAP ASoC support to DMA engine
by Russell King - ARM Linux 04 Sep '12

04 Sep '12

The following series of three patches is an attempt to convert the OMAP ASoC backend to use the DMA engine support. I'll bring your attention to the comments in patch 3 which highlight some of the features lost in this process. Some questions need answering there (in particular the one concerning pause/resume) especially as it seems the present driver could well be buggy wrt comments recently on the mailing list about ALSA pause/resume requirements. This works for me, given the wind is in the right direction and if it's sunny outside. I find the audio support on OMAP to be rather flakey at best, requiring reboots if it doesn't work first time.

5 7

[alsa-devel] [PATCH 0/3] ALSA: snd-usb: Regression fixes for 3.5
by Daniel Mack 04 Sep '12

04 Sep '12

Here are 3 more regression fixes for snd-usb in 3.5. It's annoying they haven't been found earlier, but at least we're now back to the streaming quality we had before the streaming refactoring. Anyone with a USB device, please give it a quick shot at least. Thanks, Daniel Daniel Mack (3): ALSA: snd-usb: restore delay information ALSA: snd-usb: fix calls to next_packet_size ALSA: snd-usb: fix cross-interface streaming devices sound/usb/endpoint.c | 13 +------------ sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++---- 3 files changed, 49 insertions(+), 16 deletions(-) -- 1.7.11.4

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[alsa-devel] [PATCH v3 0/3] introduce device_buffer
by Vinod Koul 04 Sep '12

04 Sep '12

Old wine in a new bottle with added flavour now. Based on our discussions in LPC, now we will report the combined delay value as sum of the buffering done in driver and hardware. This is used to report delay Additionally we tell pcm what is buffer in device/driver, this is used to check for error calulcations Vinod Koul (3): ALSA: pcm - introduce device_buffer ASoC: add device_buffer in asoc ASoC: mid-x86 - implement buffer query in sst_platform driver include/sound/pcm.h | 1 + include/sound/soc-dai.h | 6 ++++++ include/sound/soc.h | 6 ++++++ sound/core/pcm_lib.c | 10 +++++++++- sound/soc/mid-x86/sst_platform.c | 23 +++++++++++++++++++++++ sound/soc/mid-x86/sst_platform.h | 4 +++- sound/soc/soc-pcm.c | 13 ++++++++++++- 7 files changed, 60 insertions(+), 3 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] ALSA: Add DSP firmware loader
by Ian Minett 04 Sep '12

04 Sep '12

From: Ian Minett <ian_minett(a)creativelabs.com> Hi Takashi, Thanks for the recent code samples, they were very useful. We've built on them to produce this latest patchset, incorporating the updates you provided to add the loader functions, and doing a minimal update to the CA0132 codec to use the new ops to perform the firmware transfer. Please let us know if any tweaks are needed to the update. Thanks, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> --- 1: - memalloc.h - pcm.h - pcm_memory.c - sgbuf.c Include Takashi's updates: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases. Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. 2: - hda_intel.c - hda_codec.h Include Takashi's updates: Pass DMA buffer pointers in calls to setup_bdle(). Add new callback routines to controller to handle loading firmware binaries. Add new switch to Kconfig to enable DSP firmware loader. 3: - patch_ca0132.c - ca0132_regs.h Add support for DSP firmware transfer to CA0132 codec: Add transfer handler functions. Call the new snd_hda_codec_load_dsp_xxx loader ops to control download. Add chip access routines.

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[alsa-devel] [PATCH] ALSA: snd-usb: Add quirks for Playback Designs devices
by Daniel Mack 04 Sep '12

04 Sep '12

Playback Designs' USB devices have some hardware limitations on their USB interface. In particular: - They need a 20ms delay after each class compliant request as the hardware ACKs the USB packets before the device is actually ready for the next command. Sending data immediately will result in buffer overflows in the hardware. - The devices send bogus feedback data at the start of each stream which confuse the feedback format auto-detection. This patch introduces a new quirks hook that is called after each control packet and which adds a delay for all devices that match Playback Designs' USB VID for now. In addition, it adds a counter to snd_usb_endpoint to drop received packets on the floor. Another new quirks function that is called once an endpoint is started initializes that counter for these devices on their sync endpoint. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-and-tested-by: Andreas Koch <andreas(a)akdesigninc.com> Supported-by: Demian Martin <demianm_1(a)yahoo.com> --- sound/usb/card.h | 2 ++ sound/usb/endpoint.c | 8 ++++++++ sound/usb/helper.c | 5 +++++ sound/usb/quirks.c | 24 ++++++++++++++++++++++++ sound/usb/quirks.h | 10 ++++++++++ 5 files changed, 49 insertions(+) diff --git a/sound/usb/card.h b/sound/usb/card.h index 2b9ffff..d9d2b5a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -92,6 +92,8 @@ struct snd_usb_endpoint { unsigned char silence_value; unsigned int stride; int iface, alt_idx; + int skip_packets; /* quirks for devices to ignore the first n packets + in a stream */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d6e2bb4..a83a18d 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -31,6 +31,7 @@ #include "card.h" #include "endpoint.h" #include "pcm.h" +#include "quirks.h" #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 @@ -170,6 +171,11 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep, { struct urb *urb = urb_ctx->urb; + if (unlikely(ep->skip_packets > 0)) { + ep->skip_packets--; + return; + } + if (ep->sync_slave) snd_usb_handle_sync_urb(ep->sync_slave, ep, urb); @@ -822,6 +828,8 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep) ep->unlink_mask = 0; ep->phase = 0; + snd_usb_endpoint_start_quirk(ep); + /* * If this endpoint has a data endpoint as implicit feedback source, * don't start the urbs here. Instead, mark them all as available, diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 9eed8f4..c1db28f 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -21,6 +21,7 @@ #include "usbaudio.h" #include "helper.h" +#include "quirks.h" /* * combine bytes and get an integer value @@ -97,6 +98,10 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, memcpy(data, buf, size); kfree(buf); } + + snd_usb_ctl_msg_quirk(dev, pipe, request, requesttype, + value, index, data, size); + return err; } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2781726..1eaacf1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -761,3 +761,27 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, } } +void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) +{ + /* + * "Playback Design" products send bogus feedback data at the start + * of the stream. Ignore them. + */ + if ((le16_to_cpu(ep->chip->dev->descriptor.idVendor) == 0x23ba) && + ep->type == SND_USB_ENDPOINT_TYPE_SYNC) + ep->skip_packets = 4; +} + +void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, + __u8 request, __u8 requesttype, __u16 value, + __u16 index, void *data, __u16 size) +{ + /* + * "Playback Design" products need a 20ms delay after each + * class compliant request + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(20); +} + diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 03e5e94..0ca9e91 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -1,6 +1,10 @@ #ifndef __USBAUDIO_QUIRKS_H #define __USBAUDIO_QUIRKS_H +struct audioformat; +struct snd_usb_endpoint; +struct snd_usb_substream; + int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, struct usb_driver *driver, @@ -20,4 +24,10 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp); +void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); + +void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, + __u8 request, __u8 requesttype, __u16 value, + __u16 index, void *data, __u16 size); + #endif /* __USBAUDIO_QUIRKS_H */ -- 1.7.11.4

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Re: [alsa-devel] [PATCH] ASoC: mc13783: Remove mono support
by Gaëtan Carlier 04 Sep '12

04 Sep '12

Hi, On 09/03/2012 06:04 PM, Fabio Estevam wrote: > Playing a mono track on a mc13783 codec results in incorrect playback rate. > > Remove mono support so that a mono track can be played correctly. > > Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> > --- > sound/soc/codecs/mc13783.c | 8 ++++---- > 1 file changed, 4 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c > index 8f726c0..115a403 100644 > --- a/sound/soc/codecs/mc13783.c > +++ b/sound/soc/codecs/mc13783.c > @@ -659,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { > .id = MC13783_ID_STEREO_DAC, > .playback = { > .stream_name = "Playback", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = MC13783_FORMATS, > @@ -670,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { > .id = MC13783_ID_STEREO_CODEC, > .capture = { > .stream_name = "Capture", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = MC13783_RATES_RECORD, > .formats = MC13783_FORMATS, > @@ -692,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { > .id = MC13783_ID_SYNC, > .playback = { > .stream_name = "Playback", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = MC13783_FORMATS, > }, > .capture = { > .stream_name = "Capture", > - .channels_min = 1, > + .channels_min = 2, > .channels_max = 2, > .rates = MC13783_RATES_RECORD, > .formats = MC13783_FORMATS, > On imx27_3ds board: Tested-by: Gaëtan Carlier <gcembed(a)gmail.com> Regards, Gaëtan Carlier

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Re: [alsa-devel] [ASoC] Fix: Revert 'ASoC: imx-ssi: Remove mono support'.
by Gaëtan Carlier 04 Sep '12

04 Sep '12

Hi Javier, On 09/03/2012 10:27 AM, Javier Martin wrote: > The following commit should be reverted: 0865a75d4166bddc533fd50831829ceefb94f9b0 > > The bug this patch is meant to solve doesn't occur in Visstrim_M10 boards. > Furthermore, after applying this patch sound in Visstrim_M10 is played > at slower rates. > > Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> > --- > diff --git b/sound/soc/fsl/imx-ssi.c a/sound/soc/fsl/imx-ssi.c > index ac337ac..28dd76c 100644 > --- b/sound/soc/fsl/imx-ssi.c > +++ a/sound/soc/fsl/imx-ssi.c > @@ -380,13 +380,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) > static struct snd_soc_dai_driver imx_ssi_dai = { > .probe = imx_ssi_dai_probe, > .playback = { > - .channels_min = 2, > + .channels_min = 1, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = SNDRV_PCM_FMTBIT_S16_LE, > }, > .capture = { > - .channels_min = 2, > + .channels_min = 1, > .channels_max = 2, > .rates = SNDRV_PCM_RATE_8000_96000, > .formats = SNDRV_PCM_FMTBIT_S16_LE, > When applied on linux-next-20120824, the patch failed because original imx-ssi.c file looks like : static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &imx_ssi_pcm_dai_ops, }; The comment line is missing in your patch. Here is the right patch : @@ -378,18 +378,17 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) } static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &imx_ssi_pcm_dai_ops, Regards, Gaëtan Carlier.

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[alsa-devel] [Patch] Support for (original) Xbox Communicator
by Marko Friedemann 03 Sep '12

03 Sep '12

Hey guys, attached is a patch for the USB quirks to support the original Xbox Communicator (aka XboxLive Headset). I had actually submitted this some time ago against an earlier version but wasn't subscribed to the list and it was lost in moderation. I present it again now, against alsa 1.0.25. Tested and working under ubuntu 12.04.1 with kernel 3.2.0-29-generic. Regards, Marko -- "Somit werden Zitate zu zählbaren Einheiten wissenschaftlicher Reputation." -- NeMa-News - Trends in der Wissenschaft

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Re: [alsa-devel] [PATCH] sound: Remove the last mention of SNDRV_MAIN_OBJECT_FILE
by Takashi Iwai 03 Sep '12

03 Sep '12

At Sun, 2 Sep 2012 20:04:17 -0700, Josh Triplett wrote: > > SNDRV_MAIN_OBJECT_FILE hasn't done anything since the pre-git days, and > the only remaining reference occurs as a #define in sound/last.c. Drop > that last mention of it. > > Signed-off-by: Josh Triplett <josh(a)joshtriplett.org> Applied now. Thanks. Takashi > --- > sound/last.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/sound/last.c b/sound/last.c > index 7ffc182..43f2228 100644 > --- a/sound/last.c > +++ b/sound/last.c > @@ -19,7 +19,6 @@ > * > */ > > -#define SNDRV_MAIN_OBJECT_FILE > #include <linux/init.h> > #include <sound/core.h> > > -- > 1.7.10.4 >

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Re: [alsa-devel] [PATCH] ALSA: fix possible memory leak in snd_mixer_oss_build_input()
by Takashi Iwai 03 Sep '12

03 Sep '12

At Sun, 2 Sep 2012 22:10:27 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > uinfo has been allocated in this function and should be > freed before leaving from the error handling cases. > > spatch with a semantic match is used to found this problem. > (http://coccinelle.lip6.fr/) > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Applied now. Thanks. Takashi > --- > sound/core/oss/mixer_oss.c | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c > index 18297f7..29f6ded 100644 > --- a/sound/core/oss/mixer_oss.c > +++ b/sound/core/oss/mixer_oss.c > @@ -1046,6 +1046,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix > > if (kctl->info(kctl, uinfo)) { > up_read(&mixer->card->controls_rwsem); > + kfree(uinfo); > return 0; > } > strcpy(str, ptr->name); > @@ -1061,6 +1062,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix > uinfo->value.enumerated.item = slot.capture_item; > if (kctl->info(kctl, uinfo)) { > up_read(&mixer->card->controls_rwsem); > + kfree(uinfo); > return 0; > } > if (!strcmp(uinfo->value.enumerated.name, str)) { > >

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[alsa-devel] [PATCH] ALSA: hda - Fix runtime PM accounting
by Takashi Iwai 02 Sep '12

02 Sep '12

The flag codec->d3_clk_stop_ok isn't always set when the codec is powered off. If it's not set, the controller driver doesn't change the runtime PM count. This causes an unblanced accounting when multiple codecs are present. Usually, the first codec is powered down with CLK_STOP_OK=0, then the next codec with CLK_STOP_OK=1. At this moment, the first codec is changed also to CLK_STOP_OK=1, but this isn't informed to the controller. Thus it still thinks as if one codec is active. For fixing this issue, move the check of CLK_STOP_OK bit at the runtime suspend time, and simple up/down by the codec power save in azx_power_notifier(). For CLK_STOP_OK bit checks of all codecs, a new helper function is introduced. With this change, d3_clk_stop_ok field is no longer referred, thus removed. Also, d3_clk_stop bit field is moved to a place where more efficiently packed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I faced this issue while I'm checking the pm issue. Mengdong, could you verify whether this change is OK? sound/pci/hda/hda_codec.c | 30 ++++++++++++++++++++---------- sound/pci/hda/hda_codec.h | 6 +++--- sound/pci/hda/hda_intel.c | 4 ++-- 3 files changed, 25 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4a2f35c..b7ac1ff 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3546,10 +3546,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; -#ifdef CONFIG_SND_HDA_POWER_SAVE - codec->d3_stop_clk_ok = 0; -#endif - if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3572,12 +3568,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } - -#ifdef CONFIG_SND_HDA_POWER_SAVE - if ((power_state == AC_PWRST_D3) - && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) - codec->d3_stop_clk_ok = 1; -#endif } #ifdef CONFIG_SND_HDA_HWDEP @@ -4532,6 +4522,26 @@ void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) } EXPORT_SYMBOL_HDA(snd_hda_power_save); +/* check whether all codecs are ready to stop clock */ +bool snd_hda_bus_ready_for_d3(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + unsigned int state; + int fg = codec->afg ? codec->afg : codec->mfg; + + if (!codec->d3_stop_clk) + return false; + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (!(state & AC_PWRST_CLK_STOP_OK)) + return false; + } + return true; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_ready_for_d3); + /** * snd_hda_check_amp_list_power - Check the amp list and update the power * @codec: HD-audio codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 13c834f..abd85e5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -868,6 +868,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ @@ -875,9 +876,6 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; - - unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ @@ -1068,9 +1066,11 @@ const char *snd_hda_get_jack_location(u32 cfg); #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); +bool snd_hda_bus_ready_for_d3(struct hda_bus *bus); #else static inline void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) {} +static inline bool snd_hda_bus_ready_for_d3(struct hda_bus *bus) { return false; } #endif /** diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c9c779..9b02174 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2410,7 +2410,7 @@ static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + if (bus->power_keep_link_on) return; if (codec->power_on) @@ -2529,7 +2529,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; #ifdef CONFIG_SND_HDA_POWER_SAVE - if (!power_save_controller) + if (!power_save_controller || !snd_hda_bus_ready_for_d3(chip->bus)) return -EAGAIN; #endif -- 1.7.11.4

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[alsa-devel] [PATCH] ASoC: dapm: Ensure bypass paths are suspended and resumed
by Mark Brown 01 Sep '12

01 Sep '12

Since bypass paths aren't part of DAPM streams and we may not have any DAPM streams there may not be anything that triggers a DAPM sync for them. Mark all input and output widgets as dirty and then sync to do so at the end of suspend and resume. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 8 ++++++++ sound/soc/soc-dapm.c | 22 ++++++++++++++++++++++ 3 files changed, 31 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index abe373d..5b584cb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -412,6 +412,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 01ec2d6..3be2e40 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -609,6 +609,10 @@ int snd_soc_suspend(struct device *dev) SND_SOC_DAPM_STREAM_SUSPEND); } + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); + /* suspend all CODECs */ list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with @@ -756,6 +760,10 @@ static void soc_resume_deferred(struct work_struct *work) /* userspace can access us now we are back as we were before */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); + + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); } /* powers up audio subsystem after a suspend */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 53f69b5..fd97dd7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -141,6 +141,28 @@ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) } EXPORT_SYMBOL_GPL(dapm_mark_dirty); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_card *card = dapm->card; + struct snd_soc_dapm_widget *w; + + mutex_lock(&card->dapm_mutex); + + list_for_each_entry(w, &card->widgets, list) { + switch (w->id) { + case snd_soc_dapm_input: + case snd_soc_dapm_output: + dapm_mark_dirty(w, "Rechecking inputs and outputs"); + break; + default: + break; + } + } + + mutex_unlock(&card->dapm_mutex); +} +EXPORT_SYMBOL_GPL(dapm_mark_io_dirty); + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) -- 1.7.10.4

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Regression fixes for 3.5
by Daniel Mack 01 Sep '12

01 Sep '12

(please don't drop mailing lists from Cc:) On 31.08.2012 18:18, ☈king wrote: > On 08/31/2012 12:15 PM, Daniel Mack wrote: >> What kernel do you work on? Did you try the other patch that I sent >> some days ago and that already went mainline? >> https://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commit;h=01561… > > Hrm. > > I am using Gentoo's kernel from gentoo-sources. When I wrote the other > mail I was on 3.5.1, but now I'm using 3.5.3, and when I apply that > patch, I get: > > 1 out of 5 hunks FAILED -- saving rejects to file sound/usb/pcm.c.rej > % cat sound/usb/pcm.c.rej > --- sound/usb/pcm.c > +++ sound/usb/pcm.c > @@ -544,13 +544,10 @@ > subs->last_frame_number = 0; > runtime->delay = 0; > - /* clear the pending deactivation on the target EPs */ > - deactivate_endpoints(subs); > - > /* for playback, submit the URBs now; otherwise, the first hwptr_done > * updates for all URBs would happen at the same time when starting */ > if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) > - return start_endpoints(subs); > + return start_endpoints(subs, 1); > return 0; > } > > > It already doesn't have that deactivate_endpoints() line. > > So, I manually adding the 1 arg to start_endpoints() for "can_sleep". The important thing is not to call deactivate_endpoints() from the prepare callback. > Rebooting to this kernel gives the exact same results, though. Please try a kernel built from Takashi's branch: https://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git That alone should fix your problem. The three patches of this series fix some occasional stream dropouts. Thanks for testing, Daniel

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[alsa-devel] [PATCH] ASoC: samsung dma - Don't indicate support for pause/resume.
by Dylan Reid 01 Sep '12

01 Sep '12

The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..56a507a 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | -- 1.7.12.146.g16d26b1

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Re: [alsa-devel] no 5.1 surround sound on 3stack-6ch motherboard
by Raymond Yau 01 Sep '12

01 Sep '12

> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1044150 > > you need to switch channel mode from 2ch to 6ch and restore this > setting by alsactl before pulseaudio start > > Simple mixer control 'Channel Mode',0 > Capabilities: enum > Items: '2ch' '4ch' '6ch' > Item0: '2ch > > but how can i do this?? amixer -c0 set "Channel Mode" "6ch" alsactl store restart pulseaudio server http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=Do… Realtek codecs -------------- * Channel Mode This is an enum control to change the surround-channel setup, appears only when the surround channels are available. It gives the number of channels to be used, "2ch", "4ch", "6ch", and "8ch". According to the configuration, this also controls the jack-retasking of multi-I/O jacks. can you post the pulseaudio log ? I don't understand how can pulseaudio server switch to active profile surround51 when blue and pink jacks had not been retasked as output ( Pin-ctls are still IN ) and channel mode is still 2ch. analog-input rear mic and line-in ports should not be available too. http://www.intel.com/support/motherboards/desktop/sb/cs-020642.htm#6ch Jack retasking is a feature that allows an audio jack to support more than one function. For example, the pink audio jack can support microphone input when the system configuration is set to Stereo Mode, and as Center/LFE or Rear Surround (depending on the audio solution used) when set as 6 channel. Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19c50: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181345f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=05, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 index: 1 name: <alsa_card.pci-0000_00_1b.0> driver: <module-alsa-card.c> owner module: 5 properties: alsa.card = "0" alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0xf7d00000 irq 44" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" device.form_factor = "internal" device.string = "0" device.description = "Built-in Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" profiles: output:analog-stereo: Analogue Stereo Output (priority 1000) output:analog-stereo+input:analog-stereo: Analogue Stereo Duplex (priority 1010) output:analog-surround-40: Analogue Surround 4.0 Output (priority 700) output:analog-surround-40+input:analog-stereo: Analogue Surround 4.0 Output + Analogue Stereo Input (priority 710) output:analog-surround-41: Analogue Surround 4.1 Output (priority 800) output:analog-surround-41+input:analog-stereo: Analogue Surround 4.1 Output + Analogue Stereo Input (priority 810) output:analog-surround-50: Analogue Surround 5.0 Output (priority 700) output:analog-surround-50+input:analog-stereo: Analogue Surround 5.0 Output + Analogue Stereo Input (priority 710) output:analog-surround-51: Analogue Surround 5.1 Output (priority 3800) output:analog-surround-51+input:analog-stereo: Analogue Surround 5.1 Output + Analogue Stereo Input (priority 3810) input:analog-stereo: Analogue Stereo Input (priority 10) off: Off (priority 0) active profile: <output:analog-surround-51+input:analog-stereo> sinks: alsa_output.pci-0000_00_1b.0.analog-surround-51/#0: Built-in Audio Analogue Surround 5.1 sources: alsa_output.pci-0000_00_1b.0.analog-surround-51.monitor/#1: Monitor of Built-in Audio Analogue Surround 5.1 alsa_input.pci-0000_00_1b.0.analog-stereo/#2: Built-in Audio Analogue Stereo ports: analog-output: Analogue Output (priority 9900, available: unknown) properties: analog-output-headphones: Headphones (priority 9000, available: unknown) properties: analog-input-microphone-front: Front Microphone (priority 8500, available: no) properties: analog-input-microphone-rear: Rear Microphone (priority 8200, available: yes) properties: analog-input-linein: Line In (priority 8100, available: yes) properties:

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[alsa-devel] [PATCH 6/9] ARM: OMAP: Move gpio.h to include/linux/platform_data
by Tony Lindgren 01 Sep '12

01 Sep '12

This way we can remove includes of plat/gpio.h which won't work with the single zImage support. Note that we also remove the cpu_class_is_omap2() check in gpio-omap.c as the drivers should not call it as we need to make it local to arch/arm/mach-omap2 for single zImage support. While at it, arrange the related includes in the standard way. Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: linux-mtd(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Acked-by: Tony Lindgren <tony(a)atomide.com> --- arch/arm/mach-omap1/board-ams-delta.c | 3 ++- arch/arm/mach-omap1/board-osk.c | 1 + arch/arm/mach-omap1/gpio15xx.c | 1 + arch/arm/mach-omap1/gpio16xx.c | 1 + arch/arm/mach-omap1/gpio7xx.c | 1 + arch/arm/mach-omap1/include/mach/gpio.h | 2 -- arch/arm/mach-omap1/leds-h2p2-debug.c | 4 +++- arch/arm/mach-omap1/leds.c | 1 + arch/arm/mach-omap2/board-am3517evm.c | 9 ++++++--- arch/arm/mach-omap2/board-cm-t35.c | 12 +++++++----- arch/arm/mach-omap2/board-zoom-display.c | 6 +++++- arch/arm/mach-omap2/board-zoom-peripherals.c | 1 + arch/arm/mach-omap2/gpio.c | 1 + arch/arm/mach-omap2/hsmmc.c | 5 ++++- arch/arm/mach-omap2/include/mach/gpio.h | 2 -- arch/arm/mach-omap2/msdi.c | 1 + arch/arm/mach-omap2/omap_hwmod_2420_data.c | 1 - arch/arm/mach-omap2/omap_hwmod_2430_data.c | 6 +++--- arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c | 2 +- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 6 +++--- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 6 +++--- arch/arm/mach-omap2/pm24xx.c | 1 + arch/arm/mach-omap2/pm34xx.c | 2 ++ drivers/gpio/gpio-omap.c | 15 ++++++++------- drivers/mtd/nand/ams-delta.c | 8 ++++++-- include/linux/platform_data/gpio-omap.h | 15 ++------------- sound/soc/omap/sdp3430.c | 1 + 27 files changed, 65 insertions(+), 49 deletions(-) rename arch/arm/plat-omap/include/plat/gpio.h => include/linux/platform_data/gpio-omap.h (94%) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 147d665..4b5578b 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -26,6 +26,8 @@ #include <linux/export.h> #include <linux/omapfb.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/keypad-omap.h> #include <media/soc_camera.h> @@ -35,7 +37,6 @@ #include <asm/mach/map.h> #include <plat/board-ams-delta.h> -#include <linux/platform_data/keypad-omap.h> #include <plat/mux.h> #include <mach/hardware.h> diff --git a/arch/arm/mach-omap1/board-osk.c b/arch/arm/mach-omap1/board-osk.c index 9413fd3..c780a1b 100644 --- a/arch/arm/mach-omap1/board-osk.c +++ b/arch/arm/mach-omap1/board-osk.c @@ -39,6 +39,7 @@ #include <linux/mtd/partitions.h> #include <linux/mtd/physmap.h> #include <linux/i2c/tps65010.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/omap1_bl.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap1/gpio15xx.c b/arch/arm/mach-omap1/gpio15xx.c index ebef15e..98e6f39 100644 --- a/arch/arm/mach-omap1/gpio15xx.c +++ b/arch/arm/mach-omap1/gpio15xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1_MPUIO_VBASE OMAP1_MPUIO_BASE #define OMAP1510_GPIO_BASE 0xFFFCE000 diff --git a/arch/arm/mach-omap1/gpio16xx.c b/arch/arm/mach-omap1/gpio16xx.c index 2a48cd2..33f4192 100644 --- a/arch/arm/mach-omap1/gpio16xx.c +++ b/arch/arm/mach-omap1/gpio16xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP1610_GPIO1_BASE 0xfffbe400 #define OMAP1610_GPIO2_BASE 0xfffbec00 diff --git a/arch/arm/mach-omap1/gpio7xx.c b/arch/arm/mach-omap1/gpio7xx.c index acf12b7..958ce9a 100644 --- a/arch/arm/mach-omap1/gpio7xx.c +++ b/arch/arm/mach-omap1/gpio7xx.c @@ -17,6 +17,7 @@ */ #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #define OMAP7XX_GPIO1_BASE 0xfffbc000 #define OMAP7XX_GPIO2_BASE 0xfffbc800 diff --git a/arch/arm/mach-omap1/include/mach/gpio.h b/arch/arm/mach-omap1/include/mach/gpio.h index e737706..ebf86c0 100644 --- a/arch/arm/mach-omap1/include/mach/gpio.h +++ b/arch/arm/mach-omap1/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap1/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap1/leds-h2p2-debug.c b/arch/arm/mach-omap1/leds-h2p2-debug.c index f6b14a1..3ea12c7 100644 --- a/arch/arm/mach-omap1/leds-h2p2-debug.c +++ b/arch/arm/mach-omap1/leds-h2p2-debug.c @@ -14,13 +14,15 @@ #include <linux/kernel_stat.h> #include <linux/sched.h> #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> #include <asm/leds.h> #include <asm/mach-types.h> #include <plat/fpga.h> +#include <mach/hardware.h> + #include "leds.h" diff --git a/arch/arm/mach-omap1/leds.c b/arch/arm/mach-omap1/leds.c index ae6dd93..7b1a383 100644 --- a/arch/arm/mach-omap1/leds.c +++ b/arch/arm/mach-omap1/leds.c @@ -6,6 +6,7 @@ #include <linux/gpio.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/leds.h> #include <asm/mach-types.h> diff --git a/arch/arm/mach-omap2/board-am3517evm.c b/arch/arm/mach-omap2/board-am3517evm.c index 00abda1..c6d63d7 100644 --- a/arch/arm/mach-omap2/board-am3517evm.c +++ b/arch/arm/mach-omap2/board-am3517evm.c @@ -25,19 +25,22 @@ #include <linux/can/platform/ti_hecc.h> #include <linux/davinci_emac.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> -#include <mach/am35xx.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> -#include "common.h" #include <plat/usb.h> + +#include <mach/hardware.h> +#include <mach/am35xx.h> + #include <video/omapdss.h> #include <video/omap-panel-generic-dpi.h> #include <video/omap-panel-tfp410.h> +#include "common.h" #include "am35xx-emac.h" #include "mux.h" #include "control.h" diff --git a/arch/arm/mach-omap2/board-cm-t35.c b/arch/arm/mach-omap2/board-cm-t35.c index ad83654..c2b2371 100644 --- a/arch/arm/mach-omap2/board-cm-t35.c +++ b/arch/arm/mach-omap2/board-cm-t35.c @@ -23,6 +23,9 @@ #include <linux/input/matrix_keypad.h> #include <linux/delay.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/mtd-nand-omap2.h> +#include <linux/platform_data/spi-omap2-mcspi.h> #include <linux/i2c/at24.h> #include <linux/i2c/twl.h> @@ -37,17 +40,16 @@ #include <asm/mach/arch.h> #include <asm/mach/map.h> -#include "common.h" -#include <linux/platform_data/mtd-nand-omap2.h> #include <plat/gpmc.h> #include <plat/usb.h> + +#include <mach/hardware.h> + #include <video/omapdss.h> #include <video/omap-panel-generic-dpi.h> #include <video/omap-panel-tfp410.h> -#include <linux/platform_data/spi-omap2-mcspi.h> - -#include <mach/hardware.h> +#include "common.h" #include "mux.h" #include "sdram-micron-mt46h32m32lf-6.h" #include "hsmmc.h" diff --git a/arch/arm/mach-omap2/board-zoom-display.c b/arch/arm/mach-omap2/board-zoom-display.c index 660a2932..0e48a51 100644 --- a/arch/arm/mach-omap2/board-zoom-display.c +++ b/arch/arm/mach-omap2/board-zoom-display.c @@ -15,9 +15,13 @@ #include <linux/i2c/twl.h> #include <linux/spi/spi.h> #include <linux/platform_data/spi-omap2-mcspi.h> -#include <video/omapdss.h> + +#include <plat/cpu.h> + #include <mach/board-zoom.h> +#include <video/omapdss.h> + #define LCD_PANEL_RESET_GPIO_PROD 96 #define LCD_PANEL_RESET_GPIO_PILOT 55 #define LCD_PANEL_QVGA_GPIO 56 diff --git a/arch/arm/mach-omap2/board-zoom-peripherals.c b/arch/arm/mach-omap2/board-zoom-peripherals.c index 00f73b7..ae5d36b 100644 --- a/arch/arm/mach-omap2/board-zoom-peripherals.c +++ b/arch/arm/mach-omap2/board-zoom-peripherals.c @@ -19,6 +19,7 @@ #include <linux/regulator/fixed.h> #include <linux/wl12xx.h> #include <linux/mmc/host.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach-types.h> #include <asm/mach/arch.h> diff --git a/arch/arm/mach-omap2/gpio.c b/arch/arm/mach-omap2/gpio.c index 9ad7d48..7b52bdf 100644 --- a/arch/arm/mach-omap2/gpio.c +++ b/arch/arm/mach-omap2/gpio.c @@ -21,6 +21,7 @@ #include <linux/slab.h> #include <linux/interrupt.h> #include <linux/of.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/hsmmc.c b/arch/arm/mach-omap2/hsmmc.c index a9675d8..4b0e166 100644 --- a/arch/arm/mach-omap2/hsmmc.c +++ b/arch/arm/mach-omap2/hsmmc.c @@ -14,12 +14,15 @@ #include <linux/string.h> #include <linux/delay.h> #include <linux/gpio.h> -#include <mach/hardware.h> +#include <linux/platform_data/gpio-omap.h> + #include <plat/mmc.h> #include <plat/omap-pm.h> #include <plat/mux.h> #include <plat/omap_device.h> +#include <mach/hardware.h> + #include "mux.h" #include "hsmmc.h" #include "control.h" diff --git a/arch/arm/mach-omap2/include/mach/gpio.h b/arch/arm/mach-omap2/include/mach/gpio.h index be4d290..5621cc5 100644 --- a/arch/arm/mach-omap2/include/mach/gpio.h +++ b/arch/arm/mach-omap2/include/mach/gpio.h @@ -1,5 +1,3 @@ /* * arch/arm/mach-omap2/include/mach/gpio.h */ - -#include <plat/gpio.h> diff --git a/arch/arm/mach-omap2/msdi.c b/arch/arm/mach-omap2/msdi.c index fb5bc6c..9e57b4a 100644 --- a/arch/arm/mach-omap2/msdi.c +++ b/arch/arm/mach-omap2/msdi.c @@ -23,6 +23,7 @@ #include <linux/kernel.h> #include <linux/err.h> +#include <linux/platform_data/gpio-omap.h> #include <plat/omap_hwmod.h> #include <plat/omap_device.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2420_data.c b/arch/arm/mach-omap2/omap_hwmod_2420_data.c index 558c300..b305d7f 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2420_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2420_data.c @@ -18,7 +18,6 @@ #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/dmtimer.h> #include <plat/l3_2xxx.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2430_data.c b/arch/arm/mach-omap2/omap_hwmod_2430_data.c index d662766..79ac236 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2430_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2430_data.c @@ -12,15 +12,15 @@ * XXX handle crossbar/shared link difference for L3? * XXX these should be marked initdata for multi-OMAP kernels */ +#include <linux/platform_data/dsp-mcbsp.h> +#include <linux/platform_data/spi-omap2-mcspi.h> + #include <plat/omap_hwmod.h> #include <mach/irqs.h> #include <plat/cpu.h> #include <plat/dma.h> #include <plat/serial.h> #include <plat/i2c.h> -#include <plat/gpio.h> -#include <linux/platform_data/dsp-mcbsp.h> -#include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/dmtimer.h> #include <plat/mmc.h> #include <plat/l3_2xxx.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c index 161ae0a..44e4764 100644 --- a/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c @@ -10,9 +10,9 @@ */ #include <plat/omap_hwmod.h> #include <plat/serial.h> -#include <plat/gpio.h> #include <plat/dma.h> #include <plat/dmtimer.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/spi-omap2-mcspi.h> #include <mach/irqs.h> diff --git a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c index 656bb7f..15f617e 100644 --- a/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_3xxx_data.c @@ -15,6 +15,9 @@ * XXX these should be marked initdata for multi-OMAP kernels */ #include <linux/power/smartreflex.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/dsp-mcbsp.h> +#include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/omap_hwmod.h> #include <mach/irqs.h> @@ -24,10 +27,7 @@ #include <plat/l3_3xxx.h> #include <plat/l4_3xxx.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/mmc.h> -#include <linux/platform_data/dsp-mcbsp.h> -#include <linux/platform_data/spi-omap2-mcspi.h> #include <plat/dmtimer.h> #include "omap_hwmod_common_data.h" diff --git a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c index 3a06e7d..c9edaf2 100644 --- a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c @@ -19,15 +19,15 @@ */ #include <linux/io.h> +#include <linux/platform_data/gpio-omap.h> +#include <linux/platform_data/spi-omap2-mcspi.h> +#include <linux/platform_data/dsp-mcbsp.h> #include <linux/power/smartreflex.h> #include <plat/omap_hwmod.h> #include <plat/cpu.h> #include <plat/i2c.h> -#include <plat/gpio.h> #include <plat/dma.h> -#include <linux/platform_data/spi-omap2-mcspi.h> -#include <linux/platform_data/dsp-mcbsp.h> #include <plat/mmc.h> #include <plat/dmtimer.h> #include <plat/common.h> diff --git a/arch/arm/mach-omap2/pm24xx.c b/arch/arm/mach-omap2/pm24xx.c index de1caf1..2708a8b 100644 --- a/arch/arm/mach-omap2/pm24xx.c +++ b/arch/arm/mach-omap2/pm24xx.c @@ -29,6 +29,7 @@ #include <linux/irq.h> #include <linux/time.h> #include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <asm/mach/time.h> #include <asm/mach/irq.h> diff --git a/arch/arm/mach-omap2/pm34xx.c b/arch/arm/mach-omap2/pm34xx.c index e4fc88c..2a08978 100644 --- a/arch/arm/mach-omap2/pm34xx.c +++ b/arch/arm/mach-omap2/pm34xx.c @@ -28,6 +28,8 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/platform_data/gpio-omap.h> + #include <trace/events/power.h> #include <asm/suspend.h> diff --git a/drivers/gpio/gpio-omap.c b/drivers/gpio/gpio-omap.c index e6efd77..64fbce3 100644 --- a/drivers/gpio/gpio-omap.c +++ b/drivers/gpio/gpio-omap.c @@ -25,11 +25,9 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/irqdomain.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> -#include <mach/hardware.h> -#include <asm/irq.h> -#include <mach/irqs.h> -#include <asm/gpio.h> #include <asm/mach/irq.h> #define OFF_MODE 1 @@ -385,13 +383,16 @@ static int _set_gpio_triggering(struct gpio_bank *bank, int gpio, static int gpio_irq_type(struct irq_data *d, unsigned type) { struct gpio_bank *bank = irq_data_get_irq_chip_data(d); - unsigned gpio; + unsigned gpio = 0; int retval; unsigned long flags; - if (!cpu_class_is_omap2() && d->irq > IH_MPUIO_BASE) +#ifdef CONFIG_ARCH_OMAP1 + if (d->irq > IH_MPUIO_BASE) gpio = OMAP_MPUIO(d->irq - IH_MPUIO_BASE); - else +#endif + + if (!gpio) gpio = irq_to_gpio(bank, d->irq); if (type & ~IRQ_TYPE_SENSE_MASK) diff --git a/drivers/mtd/nand/ams-delta.c b/drivers/mtd/nand/ams-delta.c index 861ca8f..78a524b 100644 --- a/drivers/mtd/nand/ams-delta.c +++ b/drivers/mtd/nand/ams-delta.c @@ -23,12 +23,16 @@ #include <linux/mtd/mtd.h> #include <linux/mtd/nand.h> #include <linux/mtd/partitions.h> +#include <linux/gpio.h> +#include <linux/platform_data/gpio-omap.h> + #include <asm/io.h> -#include <mach/hardware.h> #include <asm/sizes.h> -#include <linux/gpio.h> + #include <plat/board-ams-delta.h> +#include <mach/hardware.h> + /* * MTD structure for E3 (Delta) */ diff --git a/arch/arm/plat-omap/include/plat/gpio.h b/include/linux/platform_data/gpio-omap.h similarity index 94% rename from arch/arm/plat-omap/include/plat/gpio.h rename to include/linux/platform_data/gpio-omap.h index 50fb7cc..e8741c2 100644 --- a/arch/arm/plat-omap/include/plat/gpio.h +++ b/include/linux/platform_data/gpio-omap.h @@ -1,6 +1,4 @@ /* - * arch/arm/plat-omap/include/mach/gpio.h - * * OMAP GPIO handling defines and functions * * Copyright (C) 2003-2005 Nokia Corporation @@ -155,6 +153,8 @@ #define OMAP4_GPIO_CLEARDATAOUT 0x0190 #define OMAP4_GPIO_SETDATAOUT 0x0194 +#define OMAP_MAX_GPIO_LINES 192 + #define OMAP_MPUIO(nr) (OMAP_MAX_GPIO_LINES + (nr)) #define OMAP_GPIO_IS_MPUIO(nr) ((nr) >= OMAP_MAX_GPIO_LINES) @@ -213,16 +213,5 @@ extern void omap2_gpio_prepare_for_idle(int off_mode); extern void omap2_gpio_resume_after_idle(void); extern void omap_set_gpio_debounce(int gpio, int enable); extern void omap_set_gpio_debounce_time(int gpio, int enable); -/*-------------------------------------------------------------------------*/ - -/* - * Wrappers for "new style" GPIO calls, using the new infrastructure - * which lets us plug in FPGA, I2C, and other implementations. - * - * The original OMAP-specific calls should eventually be removed. - */ - -#include <linux/errno.h> -#include <asm-generic/gpio.h> #endif diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 33f4ac2..da45b51 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -33,6 +33,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> +#include <linux/platform_data/gpio-omap.h> #include <linux/platform_data/dsp-mcbsp.h> /* Register descriptions for twl4030 codec part */

2 1

Re: [alsa-devel] [PATCH] ASoC: correct the check for NULL dma_buffer pointer
by Mark Brown 31 Aug '12

31 Aug '12

On Fri, Aug 31, 2012 at 08:55:21AM +0530, Prasad Joshi wrote: > The if condition > if (!buf && !buf->area) > > checks if the buf pointer is NULL and then dereferences it again to > check if the buffer area is NULL, resulting in possible NULL > dereference. Applied, thanks.

1 0

[alsa-devel] [PATCH 0/3] introduce device_buffer
by Vinod Koul 31 Aug '12

31 Aug '12

Old wine ina new bottle, so here is the second attempt for adding the notion of how devices (in my case a DSP) can buffer data and that can be used by sound core Vinod Koul (3): ALSA: pcm - introduce device_buffer ASoC: add device_buffer in asoc ASoC: mid-x86 - implement buffer query in sst_platform driver include/sound/pcm.h | 1 + include/sound/soc-dai.h | 6 ++++++ include/sound/soc.h | 7 ++++++- sound/core/pcm_lib.c | 14 +++++++++++--- sound/core/pcm_native.c | 6 +++--- sound/soc/mid-x86/sst_platform.c | 23 +++++++++++++++++++++++ sound/soc/mid-x86/sst_platform.h | 4 +++- sound/soc/soc-pcm.c | 13 +++++++++++++ 8 files changed, 66 insertions(+), 8 deletions(-) -- 1.7.9.5

2 4

Re: [alsa-devel] [asoc:for-next 127/128] make[1]: *** No rule to make target `sound/soc/ep93xx'.
by Fengguang Wu 31 Aug '12

31 Aug '12

On Wed, Aug 29, 2012 at 08:13:38PM +0400, Alexander Shiyan wrote: > On Tue, 28 Aug 2012 22:38:56 -0700 > Fengguang Wu <fengguang.wu(a)intel.com> wrote: > > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > > head: 03a5da2b72ba426ed475795db375fc7780338ff4 > > commit: 0ed275eff31029c39355828cb48c46c0a006e2f8 [127/128] ASoC: Rename ep93xx soc directory to cirrus > > config: x86_64-allyesconfig (attached as .config) > > Build errors: > > make[1]: *** No rule to make target `sound/soc/ep93xx'. > > Cannot reproduce this bug. Possible this breakage may be resolved by run > "make distclean" before "make all". Sorry! It's actually a bug of the build system.. I just fixed it. Thanks, Fengguang

1 0

Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Regression fixes for 3.5
by ☈king 31 Aug '12

31 Aug '12

I am having the following symptoms, before and after application of this thread's patches: Given: - An onboard card - A Logitech h390 USB headset - alsa-info of: http://www.alsa-project.org/db/?f=942b5e96b53a667270f895ec5061ca14af0339ee When: - I do: aplay $(locate .wav | grep Front) it outputs: alsa ALSA lib pcm_direct.c:980:(snd1_pcm_direct_initialize_slave) unable to install hw params ALSA lib pcm_dmix.c:1030:(snd_pcm_dmix_open) unable to initialize slave aplay: main:682: audio open error: Broken pipe - I do a Skype "Make a test sound" /var/log/messages streams dozens of these: kernel: cannot submit urb 0, error -28: not enough bandwidth Note that this is a new laptop and headset, so I haven't compared how either works on other setups, such as with other kernels or alsa versions. Any hints? Any further info needed? Thanks! —☈

2 2

[alsa-devel] double dai question and analog paths
by Raffaele Recalcati 31 Aug '12

31 Aug '12

I can play succesfully each of the following substreams: static struct snd_soc_dai_link omap3baia_dai[] = { { .name = "TLV320AIC31", .stream_name = "AIC31", .cpu_dai_name = "omap-mcbsp-dai.3", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", .init = omap3baia_tlvaic31_init, .ops = &omap3baia_tlvaic31_ops, }, { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai_name = "omap-mcbsp-dai.1", .platform_name = "omap-pcm-audio", .codec_dai_name = "twl4030-hifi", .codec_name = "twl4030-codec", .init = omap3baia_twl4030_init, .ops = &omap3baia_twl4030_ops, }, }; I have also some analog connections between TWL4030 and TLV320AIC31 and I'd like to create an automatic audio path (for example from TWL4030 HFL to TLV320AIC31 LINE2L), but I obtain "Failed to add route Virtual Tps to Tlv->LINE2L" static const struct snd_soc_dapm_route audio_map_twl4030[] = { {"Virtual Tps to Tlv", NULL, "HFL"}, /* from tps to tlv */ {"LINE2L", NULL, "Virtual Tps to Tlv"}, /* from tps to tlv */ }; Because LINE2L is obviously defined inside TWL4030 codec. static const struct snd_soc_dapm_widget dapm_widgets_twl4030[] = { SND_SOC_DAPM_OUTPUT("Virtual Tps to Tlv"), }; Is there a clean way to manage this situation? I have also tried without "Virtual Tps to Tlv" but the problem is the same. Thx, Raffaele

1 0

[alsa-devel] [PATCH] ALSA: snd-usb: use list_for_each_safe for endpoint resources
by Pavel Roskin 31 Aug '12

31 Aug '12

snd_usb_endpoint_free() frees the structure that contains its argument. Signed-off-by: Pavel Roskin <proski(a)gnu.org> Cc: stable(a)vger.kernel.org --- sound/usb/card.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d5b5c33..4a469f0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, struct snd_usb_audio *chip) { struct snd_card *card; - struct list_head *p; + struct list_head *p, *n; if (chip == (void *)-1L) return; @@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_stream_disconnect(p); } /* release the endpoint resources */ - list_for_each(p, &chip->ep_list) { + list_for_each_safe(p, n, &chip->ep_list) { snd_usb_endpoint_free(p); } /* release the midi resources */

2 1

Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 30 Aug '12

30 Aug '12

On 25.08.2012 14:17, Josh Boyer wrote: > On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: >> On 25.08.2012 14:07, Bruno Wolff III wrote: >>> On Sat, Aug 25, 2012 at 14:02:51 +0200, >>> Daniel Mack <zonque(a)gmail.com> wrote: >>>> >>>> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >>>> scheduling-while-atomic bug in PCM capture stream") and see if that >>> >>> I can try that, but it takes a long time to build a new kernel on my >>> old hardware. >>> >>>> helps? If not, can you summarize again which kernels still work for you >>>> and which don't? >>> >>> The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The earliest that >>> doesn't work is 3.6.0-0.rc2.git2.1.fc18. >>> >> >> The report you sent doesn't look like it could be caused by e9ba389c5. >> It fixes a kernel Ooops. But as it is the only relevant patch in that >> area, it would be interesting if reverting it fixes anything. > > Yep, agreed. If this revert kernel doesn't work, we're likely down to a > git bisect, Bruno. > >> Btw - thanks a lot for testing -rc kernels, much appreciated! > > Indeed! Could you please try this patch on top of Takashi's? Thanks again! Daniel

2 6

[alsa-devel] [PATCH 04/22] ASoC: Ux500: Move MSP pinctrl setup into the MSP driver
by Lee Jones 30 Aug '12

30 Aug '12

In the initial submission of the MSP driver msp1 and msp3's associated pinctrl mechanism was passed back to platform code using a plat_init() call-back routine, but it has no place in platform code. The MSP driver should set this up for the appropriate ports. Instead we use a use_pinctrl identifier which is passed from platform_data/Device Tree which indicates which ports should use pinctrl. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- arch/arm/mach-ux500/board-mop500-msp.c | 75 +------------------------------- arch/arm/mach-ux500/include/mach/msp.h | 2 - sound/soc/ux500/ux500_msp_i2s.c | 75 +++++++++++++++++++++++++------- sound/soc/ux500/ux500_msp_i2s.h | 2 - 4 files changed, 60 insertions(+), 94 deletions(-) diff --git a/arch/arm/mach-ux500/board-mop500-msp.c b/arch/arm/mach-ux500/board-mop500-msp.c index df15646..fc78d5d 100644 --- a/arch/arm/mach-ux500/board-mop500-msp.c +++ b/arch/arm/mach-ux500/board-mop500-msp.c @@ -23,53 +23,6 @@ #include "devices-db8500.h" #include "pins-db8500.h" -/* MSP1/3 Tx/Rx usage protection */ -static DEFINE_SPINLOCK(msp_rxtx_lock); - -/* Reference Count */ -static int msp_rxtx_ref; - -/* Pin modes */ -struct pinctrl *msp1_p; -struct pinctrl_state *msp1_def; -struct pinctrl_state *msp1_sleep; - -int msp13_i2s_init(void) -{ - int retval = 0; - unsigned long flags; - - spin_lock_irqsave(&msp_rxtx_lock, flags); - if (msp_rxtx_ref == 0 && !(IS_ERR(msp1_p) || IS_ERR(msp1_def))) { - retval = pinctrl_select_state(msp1_p, msp1_def); - if (retval) - pr_err("could not set MSP1 defstate\n"); - } - if (!retval) - msp_rxtx_ref++; - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - - return retval; -} - -int msp13_i2s_exit(void) -{ - int retval = 0; - unsigned long flags; - - spin_lock_irqsave(&msp_rxtx_lock, flags); - WARN_ON(!msp_rxtx_ref); - msp_rxtx_ref--; - if (msp_rxtx_ref == 0 && !(IS_ERR(msp1_p) || IS_ERR(msp1_sleep))) { - retval = pinctrl_select_state(msp1_p, msp1_sleep); - if (retval) - pr_err("could not set MSP1 sleepstate\n"); - } - spin_unlock_irqrestore(&msp_rxtx_lock, flags); - - return retval; -} - static struct stedma40_chan_cfg msp0_dma_rx = { .high_priority = true, .dir = STEDMA40_PERIPH_TO_MEM, @@ -132,8 +85,6 @@ static struct msp_i2s_platform_data msp1_platform_data = { .id = MSP_I2S_1, .msp_i2s_dma_rx = NULL, .msp_i2s_dma_tx = &msp1_dma_tx, - .msp_i2s_init = msp13_i2s_init, - .msp_i2s_exit = msp13_i2s_exit, }; static struct stedma40_chan_cfg msp2_dma_rx = { @@ -219,47 +170,23 @@ static struct msp_i2s_platform_data msp3_platform_data = { .id = MSP_I2S_3, .msp_i2s_dma_rx = &msp1_dma_rx, .msp_i2s_dma_tx = NULL, - .msp_i2s_init = msp13_i2s_init, - .msp_i2s_exit = msp13_i2s_exit, }; int mop500_msp_init(struct device *parent) { - struct platform_device *msp1; - pr_info("%s: Register platform-device 'snd-soc-mop500'.\n", __func__); platform_device_register(&snd_soc_mop500); pr_info("Initialize MSP I2S-devices.\n"); db8500_add_msp_i2s(parent, 0, U8500_MSP0_BASE, IRQ_DB8500_MSP0, &msp0_platform_data); - msp1 = db8500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, + db8500_add_msp_i2s(parent, 1, U8500_MSP1_BASE, IRQ_DB8500_MSP1, &msp1_platform_data); db8500_add_msp_i2s(parent, 2, U8500_MSP2_BASE, IRQ_DB8500_MSP2, &msp2_platform_data); db8500_add_msp_i2s(parent, 3, U8500_MSP3_BASE, IRQ_DB8500_MSP1, &msp3_platform_data); - /* Get the pinctrl handle for MSP1 */ - if (msp1) { - msp1_p = pinctrl_get(&msp1->dev); - if (IS_ERR(msp1_p)) - dev_err(&msp1->dev, "could not get MSP1 pinctrl\n"); - else { - msp1_def = pinctrl_lookup_state(msp1_p, - PINCTRL_STATE_DEFAULT); - if (IS_ERR(msp1_def)) { - dev_err(&msp1->dev, - "could not get MSP1 defstate\n"); - } - msp1_sleep = pinctrl_lookup_state(msp1_p, - PINCTRL_STATE_SLEEP); - if (IS_ERR(msp1_sleep)) - dev_err(&msp1->dev, - "could not get MSP1 idlestate\n"); - } - } - pr_info("%s: Register platform-device 'ux500-pcm'\n", __func__); platform_device_register(&ux500_pcm); diff --git a/arch/arm/mach-ux500/include/mach/msp.h b/arch/arm/mach-ux500/include/mach/msp.h index 798be19..3cc7142 100644 --- a/arch/arm/mach-ux500/include/mach/msp.h +++ b/arch/arm/mach-ux500/include/mach/msp.h @@ -22,8 +22,6 @@ struct msp_i2s_platform_data { enum msp_i2s_id id; struct stedma40_chan_cfg *msp_i2s_dma_rx; struct stedma40_chan_cfg *msp_i2s_dma_tx; - int (*msp_i2s_init) (void); - int (*msp_i2s_exit) (void); }; #endif diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 36be11e..2cbfc54 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -15,6 +15,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/pinctrl/consumer.h> #include <linux/delay.h> #include <linux/slab.h> @@ -25,6 +26,17 @@ #include "ux500_msp_i2s.h" +/* MSP1/3 Tx/Rx usage protection */ +static DEFINE_SPINLOCK(msp_rxtx_lock); + +/* Pin modes */ +struct pinctrl *pinctrl_p; +struct pinctrl_state *pinctrl_def; +struct pinctrl_state *pinctrl_sleep; + +/* Reference Count */ +int pinctrl_rxtx_ref; + /* Protocol desciptors */ static const struct msp_protdesc prot_descs[] = { { /* I2S */ @@ -352,17 +364,23 @@ static int configure_multichannel(struct ux500_msp *msp, static int enable_msp(struct ux500_msp *msp, struct ux500_msp_config *config) { - int status = 0; + int status = 0, retval = 0; u32 reg_val_DMACR, reg_val_GCR; + unsigned long flags; /* Check msp state whether in RUN or CONFIGURED Mode */ - if ((msp->msp_state == MSP_STATE_IDLE) && (msp->plat_init)) { - status = msp->plat_init(); - if (status) { - dev_err(msp->dev, "%s: ERROR: Failed to init MSP (%d)!\n", - __func__, status); - return status; + if (msp->msp_state == MSP_STATE_IDLE) { + spin_lock_irqsave(&msp_rxtx_lock, flags); + if (pinctrl_rxtx_ref == 0 && + !(IS_ERR(pinctrl_p) || IS_ERR(pinctrl_def))) { + retval = pinctrl_select_state(pinctrl_p, + pinctrl_def); + if (retval) + pr_err("could not set MSP defstate\n"); } + if (!retval) + pinctrl_rxtx_ref++; + spin_unlock_irqrestore(&msp_rxtx_lock, flags); } /* Configure msp with protocol dependent settings */ @@ -620,7 +638,8 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) { - int status = 0; + int status = 0, retval = 0; + unsigned long flags; dev_dbg(msp->dev, "%s: Enter (dir = 0x%01x).\n", __func__, dir); @@ -631,12 +650,19 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) writel((readl(msp->registers + MSP_GCR) & (~(FRAME_GEN_ENABLE | SRG_ENABLE))), msp->registers + MSP_GCR); - if (msp->plat_exit) - status = msp->plat_exit(); - if (status) - dev_warn(msp->dev, - "%s: WARN: ux500_msp_i2s_exit failed (%d)!\n", - __func__, status); + + spin_lock_irqsave(&msp_rxtx_lock, flags); + WARN_ON(!pinctrl_rxtx_ref); + pinctrl_rxtx_ref--; + if (pinctrl_rxtx_ref == 0 && + !(IS_ERR(pinctrl_p) || IS_ERR(pinctrl_sleep))) { + retval = pinctrl_select_state(pinctrl_p, + pinctrl_sleep); + if (retval) + pr_err("could not set MSP sleepstate\n"); + } + spin_unlock_irqrestore(&msp_rxtx_lock, flags); + writel(0, msp->registers + MSP_GCR); writel(0, msp->registers + MSP_TCF); writel(0, msp->registers + MSP_RCF); @@ -678,8 +704,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->id = platform_data->id; msp->dev = &pdev->dev; - msp->plat_init = platform_data->msp_i2s_init; - msp->plat_exit = platform_data->msp_i2s_exit; msp->dma_cfg_rx = platform_data->msp_i2s_dma_rx; msp->dma_cfg_tx = platform_data->msp_i2s_dma_tx; @@ -717,6 +741,25 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, dev_dbg(&pdev->dev, "I2S device-name: '%s'\n", i2s_cont->name); msp->i2s_cont = i2s_cont; + pinctrl_p = pinctrl_get(msp->dev); + if (IS_ERR(pinctrl_p)) + dev_err(&pdev->dev, "could not get MSP pinctrl\n"); + else { + pinctrl_def = pinctrl_lookup_state(pinctrl_p, + PINCTRL_STATE_DEFAULT); + if (IS_ERR(pinctrl_def)) { + dev_err(&pdev->dev, + "could not get MSP defstate (%li)\n", + PTR_ERR(pinctrl_def)); + } + pinctrl_sleep = pinctrl_lookup_state(pinctrl_p, + PINCTRL_STATE_SLEEP); + if (IS_ERR(pinctrl_sleep)) + dev_err(&pdev->dev, + "could not get MSP idlestate (%li)\n", + PTR_ERR(pinctrl_def)); + } + return 0; err_i2s_cont: diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 2d9136d..1ce336f 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -524,8 +524,6 @@ struct ux500_msp { struct dma_chan *rx_pipeid; enum msp_state msp_state; int (*transfer) (struct ux500_msp *msp, struct i2s_message *message); - int (*plat_init) (void); - int (*plat_exit) (void); struct timer_list notify_timer; int def_elem_len; unsigned int dir_busy; -- 1.7.9.5

2 5

Re: [alsa-devel] [PATCH v2 02/14] MFD: twl4030-audio: Convert to use devm_kzalloc
by Peter Ujfalusi 30 Aug '12

30 Aug '12

Hi Tero, On 08/28/2012 12:50 PM, Tero Kristo wrote: > On Tue, 2012-08-14 at 17:22 +0300, Peter Ujfalusi wrote: >> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > > I think this one could use a short commit message, also about why > kfree():s are dropped (handled internally by devm_* etc.) I don't like empty commit messages either but this patch is self explaining I would think that the commit title explains it. -- Péter

1 0

Re: [alsa-devel] REGRESSION: Webcam mic for 145f:015b not longer working with 1.0.24 and above (OK with 1.0.23)
by Daniel Mack 30 Aug '12

30 Aug '12

On 27.08.2012 12:33, Kostas wrote: > On 27/08/2012 12:26 μμ, Daniel Mack wrote: >> On 27.08.2012 11:00, Kostas wrote: >>> Hi All, >>> >>> I have a 'Trust Webcam' (USB ID: 145f:015b) that comes with a microphone. >>> The microphone works with Alsa 1.0.23 but it does not work with the >>> latest Alsa 1.0.25 neither with Alsa 1.0.24. >>> Apparently this is a regression. >>> >>> What shall I do to figure out the source of the problem? >>> >>> Here are attached my alsa-infos, with Alsa 1.0.23 (working), 1.0.24 (NOT >>> working) and 1.0.25 (NOT working). >> Can you provide dmesg protocols for both a working and a non-working >> case please? >> >> > Below are attached the dmesg files. > Thanks - the logs look both good, I guess the interesting messages are only generated once you try to start the stream. Can you try this?

2 22

[alsa-devel] NRPACKS setting
by Raymond Celius 29 Aug '12

29 Aug '12

Hi, Is the NRPACKS setting equal to the number of packets per URB ? Best regards, Raymond Den 29. aug.. 2012 kl. 16.14 skrev alsa-devel-request(a)alsa-project.org: > Send Alsa-devel mailing list submissions to > alsa-devel(a)alsa-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > or, via email, send a message with subject or body 'help' to > alsa-devel-request(a)alsa-project.org > > You can reach the person managing the list at > alsa-devel-owner(a)alsa-project.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Alsa-devel digest..." > > > Today's Topics: > > 1. Re: [PATCH] ALSA: hda - Fix runtime PM accounting (Lin, Mengdong) > 2. Re: Logitech USB headset not working in 3.6-rc3 (Takashi Iwai) > 3. Re: Logitech USB headset not working in 3.6-rc3 (Daniel Mack) > 4. Re: [PATCH] ALSA: hda - Fix runtime PM accounting (Takashi Iwai) > 5. Re: Logitech USB headset not working in 3.6-rc3 (Takashi Iwai) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 29 Aug 2012 12:19:32 +0000 > From: "Lin, Mengdong" <mengdong.lin(a)intel.com> > To: Takashi Iwai <tiwai(a)suse.de> > Cc: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> > Subject: Re: [alsa-devel] [PATCH] ALSA: hda - Fix runtime PM > accounting > Message-ID: > <F46914AEC2663F4A9BB62374E5EEF8F81DAFDF(a)SHSMSX101.ccr.corp.intel.com> > Content-Type: text/plain; charset="us-ascii" > >> Now I introduced codec->epss flag that is initialized once in >> snd_hda_codec_new(), and let the codec driver overrides this flag. >> One bonus is that you can avoid the unnecessary codec parameter >> read at >> each power change. > > Yes, this is better. And the patch works well for me - [PATCH 1/2] > ALSA: hda - Avoid unnecessary parameter read for EPSS. > >> In anyway, there are a few issues: >> - The refcounting gets broken after module unload. It's because we >> leave the codec's pm_runtime usages after removal. > > Thanks for pointing out this! But I think the following codec can > introduce risk of race among multiple codecs to modify ' chip- > >power_count', > because only one codec's power on/off is serialized. > static void azx_power_notify(struct hda_bus *bus, struct hda_codec > *codec) > { > ... > if (codec->power_on) { > pm_runtime_get_sync(&chip->pci->dev); > chip->power_count++; > } else { > pm_runtime_put_sync(&chip->pci->dev); > chip->power_count--; > } > } > > How about calling pm_runtime_put_noidle() when freeing a codec in D0? > Maybe we can add a bus notify function like this: > Static void azx_codec_free_notify(struct hda_codec *codec) > { > If(codec->power_on) > pm_runtime_put_noidle(&codec->bus->pci->dev); > } > And snd_hda_codec_free() can trigger this notify. > >> - The D3-stop-clock check isn't done properly for the codecs >> providing >> own set_power_state ops. > > I cannot test the branch 'if (codec->patch_ops.set_power_state)' in > hda_set_power_state() for lack of such a codec. > But the logic seems right and the other branch works well for me :-) > >> - Ditto for the repeated power-set sequence, the error check, etc. > In hda_sync_power_state(), > if ((state & 0xff) == power_state) ... should be if ((state & > 0xf0)>> 4 == power_state), because PS-Act is bits 7:4 > break; > > And would you please explain why changing to D3 need not wait? One > of my codec does have a delay on change from D0 to D3. > > Thanks > Mengdong > > > > ------------------------------ > > Message: 2 > Date: Wed, 29 Aug 2012 15:29:03 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > To: Daniel Mack <zonque(a)gmail.com> > Cc: Bruno Wolff III <bruno(a)wolff.to>, alsa-devel(a)alsa-project.org, > linux-kernel(a)vger.kernel.org, Josh Boyer <jwboyer(a)redhat.com> > Subject: Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3 > Message-ID: <s5hsjb56d40.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=US-ASCII > > At Wed, 29 Aug 2012 13:26:25 +0200, > Daniel Mack wrote: >> >> [1 <text/plain; ISO-8859-1 (7bit)>] >> On 25.08.2012 14:17, Josh Boyer wrote: >>> On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: >>>> On 25.08.2012 14:07, Bruno Wolff III wrote: >>>>> On Sat, Aug 25, 2012 at 14:02:51 +0200, >>>>> Daniel Mack <zonque(a)gmail.com> wrote: >>>>>> >>>>>> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >>>>>> scheduling-while-atomic bug in PCM capture stream") and see if >>>>>> that >>>>> >>>>> I can try that, but it takes a long time to build a new kernel >>>>> on my >>>>> old hardware. >>>>> >>>>>> helps? If not, can you summarize again which kernels still work >>>>>> for you >>>>>> and which don't? >>>>> >>>>> The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The >>>>> earliest that >>>>> doesn't work is 3.6.0-0.rc2.git2.1.fc18. >>>>> >>>> >>>> The report you sent doesn't look like it could be caused by >>>> e9ba389c5. >>>> It fixes a kernel Ooops. But as it is the only relevant patch in >>>> that >>>> area, it would be interesting if reverting it fixes anything. >>> >>> Yep, agreed. If this revert kernel doesn't work, we're likely >>> down to a >>> git bisect, Bruno. >>> >>>> Btw - thanks a lot for testing -rc kernels, much appreciated! >>> >>> Indeed! >> >> Could you please try this patch on top of Takashi's? Thanks again! >> >> >> Daniel >> >> [2 0001-ALSA-snd-usb-Fix-URB-cancellation-at-stream-start.patch >> <text/x-patch (7bit)>] >>> From 6617bb2463ae3fad21390b59cc2a71f96b9e9ca8 Mon Sep 17 00:00:00 >>> 2001 >> From: Daniel Mack <zonque(a)gmail.com> >> Date: Wed, 29 Aug 2012 13:17:05 +0200 >> Subject: [PATCH] ALSA: snd-usb: Fix URB cancellation at stream start >> >> Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug >> in >> PCM capture stream") fixed a scheduling-while-atomic bug that >> happened >> when snd_usb_endpoint_start was called from the trigger callback, >> which >> is an atmic context. However, the patch breaks the idea of the >> endpoints >> reference counting, which is the reason why the driver has been >> refactored lately. >> >> Revert that commit and let snd_usb_endpoint_start() take care of >> the URB >> cancellation again. As this function is called from both atomic and >> non-atomic context, add a flag to denote whether the function may >> sleep. >> >> Signed-off-by: Daniel Mack <zonque(a)gmail.com> >> Cc: stable(a)kernel.org [3.5+] >> --- >> sound/usb/endpoint.c | 10 ++++++++-- >> sound/usb/endpoint.h | 2 +- >> sound/usb/pcm.c | 13 +++++-------- >> 3 files changed, 14 insertions(+), 11 deletions(-) >> >> diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c >> index c411812..678456c 100644 >> --- a/sound/usb/endpoint.c >> +++ b/sound/usb/endpoint.c >> @@ -799,7 +799,9 @@ int snd_usb_endpoint_set_params(struct >> snd_usb_endpoint *ep, >> /** >> * snd_usb_endpoint_start: start an snd_usb_endpoint >> * >> - * @ep: the endpoint to start >> + * @ep: the endpoint to start >> + * @can_sleep: flag indicating whether the operation is executed in >> + * non-atomic context >> * >> * A call to this function will increment the use count of the >> endpoint. >> * In case it is not already running, the URBs for this endpoint >> will be >> @@ -809,7 +811,7 @@ int snd_usb_endpoint_set_params(struct >> snd_usb_endpoint *ep, >> * >> * Returns an error if the URB submission failed, 0 in all other >> cases. >> */ >> -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) >> +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int >> can_sleep) >> { >> int err; >> unsigned int i; >> @@ -821,6 +823,10 @@ int snd_usb_endpoint_start(struct >> snd_usb_endpoint *ep) >> if (++ep->use_count != 1) >> return 0; >> >> + /* just to be sure */ >> + deactivate_urbs(ep, 0, can_sleep); >> + wait_clear_urbs(ep); > > It'd be safer to protect the call of wait_clear_urbs() when > can_sleep=0. > > > thanks, > > Takashi > > > ------------------------------ > > Message: 3 > Date: Wed, 29 Aug 2012 15:32:34 +0200 > From: Daniel Mack <zonque(a)gmail.com> > To: Takashi Iwai <tiwai(a)suse.de> > Cc: Bruno Wolff III <bruno(a)wolff.to>, alsa-devel(a)alsa-project.org, > linux-kernel(a)vger.kernel.org, Josh Boyer <jwboyer(a)redhat.com> > Subject: Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3 > Message-ID: <503E19F2.3080302(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 29.08.2012 15:29, Takashi Iwai wrote: >> At Wed, 29 Aug 2012 13:26:25 +0200, >> Daniel Mack wrote: >>> >>> [1 <text/plain; ISO-8859-1 (7bit)>] >>> On 25.08.2012 14:17, Josh Boyer wrote: >>>> On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: >>>>> On 25.08.2012 14:07, Bruno Wolff III wrote: >>>>>> On Sat, Aug 25, 2012 at 14:02:51 +0200, >>>>>> Daniel Mack <zonque(a)gmail.com> wrote: >>>>>>> >>>>>>> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >>>>>>> scheduling-while-atomic bug in PCM capture stream") and see if >>>>>>> that >>>>>> >>>>>> I can try that, but it takes a long time to build a new kernel >>>>>> on my >>>>>> old hardware. >>>>>> >>>>>>> helps? If not, can you summarize again which kernels still >>>>>>> work for you >>>>>>> and which don't? >>>>>> >>>>>> The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The >>>>>> earliest that >>>>>> doesn't work is 3.6.0-0.rc2.git2.1.fc18. >>>>>> >>>>> >>>>> The report you sent doesn't look like it could be caused by >>>>> e9ba389c5. >>>>> It fixes a kernel Ooops. But as it is the only relevant patch in >>>>> that >>>>> area, it would be interesting if reverting it fixes anything. >>>> >>>> Yep, agreed. If this revert kernel doesn't work, we're likely >>>> down to a >>>> git bisect, Bruno. >>>> >>>>> Btw - thanks a lot for testing -rc kernels, much appreciated! >>>> >>>> Indeed! >>> >>> Could you please try this patch on top of Takashi's? Thanks again! >>> >>> >>> Daniel >>> >>> [2 0001-ALSA-snd-usb-Fix-URB-cancellation-at-stream-start.patch >>> <text/x-patch (7bit)>] >>>> From 6617bb2463ae3fad21390b59cc2a71f96b9e9ca8 Mon Sep 17 00:00:00 >>>> 2001 >>> From: Daniel Mack <zonque(a)gmail.com> >>> Date: Wed, 29 Aug 2012 13:17:05 +0200 >>> Subject: [PATCH] ALSA: snd-usb: Fix URB cancellation at stream start >>> >>> Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic >>> bug in >>> PCM capture stream") fixed a scheduling-while-atomic bug that >>> happened >>> when snd_usb_endpoint_start was called from the trigger callback, >>> which >>> is an atmic context. However, the patch breaks the idea of the >>> endpoints >>> reference counting, which is the reason why the driver has been >>> refactored lately. >>> >>> Revert that commit and let snd_usb_endpoint_start() take care of >>> the URB >>> cancellation again. As this function is called from both atomic and >>> non-atomic context, add a flag to denote whether the function may >>> sleep. >>> >>> Signed-off-by: Daniel Mack <zonque(a)gmail.com> >>> Cc: stable(a)kernel.org [3.5+] >>> --- >>> sound/usb/endpoint.c | 10 ++++++++-- >>> sound/usb/endpoint.h | 2 +- >>> sound/usb/pcm.c | 13 +++++-------- >>> 3 files changed, 14 insertions(+), 11 deletions(-) >>> >>> diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c >>> index c411812..678456c 100644 >>> --- a/sound/usb/endpoint.c >>> +++ b/sound/usb/endpoint.c >>> @@ -799,7 +799,9 @@ int snd_usb_endpoint_set_params(struct >>> snd_usb_endpoint *ep, >>> /** >>> * snd_usb_endpoint_start: start an snd_usb_endpoint >>> * >>> - * @ep: the endpoint to start >>> + * @ep: the endpoint to start >>> + * @can_sleep: flag indicating whether the operation is executed in >>> + * non-atomic context >>> * >>> * A call to this function will increment the use count of the >>> endpoint. >>> * In case it is not already running, the URBs for this endpoint >>> will be >>> @@ -809,7 +811,7 @@ int snd_usb_endpoint_set_params(struct >>> snd_usb_endpoint *ep, >>> * >>> * Returns an error if the URB submission failed, 0 in all other >>> cases. >>> */ >>> -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) >>> +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int >>> can_sleep) >>> { >>> int err; >>> unsigned int i; >>> @@ -821,6 +823,10 @@ int snd_usb_endpoint_start(struct >>> snd_usb_endpoint *ep) >>> if (++ep->use_count != 1) >>> return 0; >>> >>> + /* just to be sure */ >>> + deactivate_urbs(ep, 0, can_sleep); >>> + wait_clear_urbs(ep); >> >> It'd be safer to protect the call of wait_clear_urbs() when >> can_sleep=0. > > Right. New patch attached. > >

1 0

[alsa-devel] [PATCH V2 0/6] ARM/ASoC: Davinci: Add Device Tree Support
by Hebbar, Gururaja 29 Aug '12

29 Aug '12

This patch-set is a cumulative of below tasks 1. Split asp header into platform and audio specific Davinci driver is shared by few OMAP platforms (like TI81xx, AM335x). Splitting asp header into Davinci platform specific and Audio specific helps to share them across platforms. 2. Add DT support for Davinci Audio Module (McASP, machine, pcm & Codec) This patch was tested on AM18x Board (Tested to check if the commits doesn't break on compile or runtime) and is based on repo at http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git; a=shortlog;h=refs/heads/for-3.7 (v3.6-rc2) Note: Matt Porter from TI is working on DMA engine conversion of DaVinci EDMA support and DT conversion of DMA parts will be taken up post that is complete. Since Audio requires DMA to work, this patch set does not yet provide complete functionality required to use sound on DT-based platforms, but is a step towards that direction. Changes from V1: - Instanciate seudo-device (davinci-pcm) directly from DAI driver rather than having a separate virtual-device registered - add more explanation to DT parameters - Change DT parameter from "ti,codec-clock" to "ti,codec-clock-rate" - use standard DT GPIO reference by phandle to populate gpio based DT parameters - Dont return err if some DT parameters are missing. They will be treated as 0. - Through Error warning if num-serializer != "serial-dir" array size - Coding Style correction Hebbar, Gururaja (6): ASoC: davinci: davinci-pcm does not need to be a plaform_driver ARM: Davinci: Remove references to davinci pcm ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific ASoC: Davinci: McASP: add device tree support for McASP ASoC: Davinci: machine: Add device tree binding ASoC: tlv320aic3x: Add device tree bindings .../bindings/sound/davinci-evm-audio.txt | 63 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 44 +++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 20 ++ arch/arm/mach-davinci/asp.h | 49 +++++ arch/arm/mach-davinci/davinci.h | 3 +- arch/arm/mach-davinci/devices-da8xx.c | 8 +- arch/arm/mach-davinci/devices.c | 11 - arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- .../linux/platform_data/davinci_asp.h | 73 ++----- sound/soc/codecs/tlv320aic3x.c | 31 +++ sound/soc/davinci/davinci-evm.c | 203 +++++++++++++++++--- sound/soc/davinci/davinci-i2s.c | 13 +- sound/soc/davinci/davinci-mcasp.c | 134 +++++++++++++- sound/soc/davinci/davinci-mcasp.h | 3 +- sound/soc/davinci/davinci-pcm.c | 24 +-- sound/soc/davinci/davinci-pcm.h | 6 +- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/davinci/davinci-vcif.c | 8 + 22 files changed, 573 insertions(+), 132 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt create mode 100644 arch/arm/mach-davinci/asp.h rename arch/arm/mach-davinci/include/mach/asp.h => include/linux/platform_data/davinci_asp.h (56%)

3 16

Re: [alsa-devel] [PATCHv2] sound: fix emu8000 DRAM sized 512 KiB too small
by Takashi Iwai 28 Aug '12

28 Aug '12

At Mon, 27 Aug 2012 22:25:21 -0400, David Flater wrote: > > v2: Fixed result still wrong in the case of 512 KiB DRAM. Oops. > > Applicable to 3.5.3 mainline. > > In emu8000.c, size_dram determines the amount of memory on the sound card by > doing write/readback tests starting at 512 KiB and incrementing by 512 KiB. > On success, detected_size is updated to the successful address and testing > continues. On failure, the loop is immediately exited. The resulting > detected_size is 512 KiB too small except in two special cases: > > 1. If there is no memory, the initial 0 value of detected_size is used, which > is correct. > 2. If the address space wraps around, detected_size is updated before the > bailout, so the result is correct. > > The patch corrects all cases and was tested with an AWE64 Gold. Before: > EMU8000 [0x620]: 3584 Kb on-board memory detected > asfxload 4GMGSMT.SF2 (4174814 B) fails. > After: > EMU8000 [0x620]: 4096 Kb on-board memory detected > asfxload 4GMGSMT.SF2 succeeds. > > I do not have a card with 512 KiB to test with, but by forcibly enabling the > added conditional I verified on the AWE64 Gold that it detects 512 KiB > (successfully reading from the first memory location) and does not hang the > card. > > C.f. Bug 46451 https://bugzilla.kernel.org/show_bug.cgi?id=46451 > > Signed-off-by: David Flater <dave(a)flaterco.com> Thanks, applied. Takashi > --- > > --- linux-3.5.3-nameless/sound/isa/sb/emu8000.c.orig 2012-08-25 22:32:13.000000000 -0400 > +++ linux-3.5.3-nameless/sound/isa/sb/emu8000.c 2012-08-27 22:06:15.729510037 -0400 > @@ -417,9 +417,6 @@ size_dram(struct snd_emu8000 *emu) > EMU8000_SMLD_READ(emu); /* discard stale data */ > if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) > break; /* no memory at this address */ > - > - detected_size = size; > - > snd_emu8000_read_wait(emu); > > /* > @@ -432,6 +429,18 @@ size_dram(struct snd_emu8000 *emu) > if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) > break; /* we must have wrapped around */ > snd_emu8000_read_wait(emu); > + > + /* Otherwise, it's valid memory. */ > + detected_size = size + 512 * 1024; > + } > + > + /* Distinguish 512 KiB from 0. */ > + if (detected_size == 0) { > + snd_emu8000_read_wait(emu); > + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); > + EMU8000_SMLD_READ(emu); /* discard stale data */ > + if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) > + detected_size = 512 * 1024; > } > > /* wait until FULL bit in SMAxW register is false */ > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ > -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] [PATCH] ASoC: Rename ep93xx soc directory to cirrus
by Mark Brown 28 Aug '12

28 Aug '12

On Tue, Aug 21, 2012 at 07:21:51PM +0400, Alexander Shiyan wrote: > This patch is to rename the directory "ep93xx" in "cirrus". > Name more accurately reflects the manufacturer and allows to add > drivers not only for architecture ep93xx in this directory. > Patch not contain any functional changes. Applied, thanks.

1 0

[alsa-devel] [RESUBMIT PATCH] ASoC: SAMSUNG: Change Kconfig to support all SAMSUNG ASoC
by Sangsu Park 28 Aug '12

28 Aug '12

All SAMSUNG ASoC needs SND_SOC_SAMSUNG configuration. This patch change Kconfig to support all SAMSUNG ASoC. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f17dd25..e7b8317 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on PLAT_SAMSUNG select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help -- 1.7.1 On 08/28/12 8:00, Kukjin Kim wrote: > On 08/27/12 11:08, Mark Brown wrote: > > On Mon, Aug 27, 2012 at 11:06:47AM -0700, Kukjin Kim wrote: > > > >> Hmm, all of Samsung stuff is available...so how about? > >> > >> + depends on PLAT_SAMSUNG > > > > Yes, that's a very good idea! > > Mark, thanks. > > Sangsu, could you please re-submit as per my suggestion? > > Thanks. > Kukjin, thanks. This is re-submit patch. In fact, I'd wanted to submit your idea :) Thanks. > Best regards, > Kgene. > -- > Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung > Electronics Co., Ltd.

3 2

[alsa-devel] [PATCH] sound: tegra_alc5632: remove HP detect GPIO inversion
by Stephen Warren 28 Aug '12

28 Aug '12

From: Stephen Warren <swarren(a)nvidia.com> Both the schematics and practical testing show that the HP detect GPIO is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio should not specify to invert the signal. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Cc: Marc Dietrich <marvin24(a)gmx.de> Cc: Leon Romanovsky <leon(a)leon.nu> Cc: <stable(a)vger.kernel.org> # v3.4 v3.5 --- sound/soc/tegra/tegra_alc5632.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e463529..76cb1b3 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { .name = "Headset detection", .report = SND_JACK_HEADSET, .debounce_time = 150, - .invert = 1, }; static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { -- 1.7.9.5

4 3

Re: [alsa-devel] [PATCH v2 02/14] MFD: twl4030-audio: Convert to use devm_kzalloc
by Mark Brown 28 Aug '12

28 Aug '12

On Tue, Aug 28, 2012 at 12:50:10PM +0300, Tero Kristo wrote: > On Tue, 2012-08-14 at 17:22 +0300, Peter Ujfalusi wrote: > > Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > I think this one could use a short commit message, also about why > kfree():s are dropped (handled internally by devm_* etc.) Honestly I think people should be able to figure this out, it's not like it's an obscure API nobody ever used before.

1 0

[alsa-devel] [PATCH] ALSA: hda: Make core jack usage unconditional
by Mark Brown 28 Aug '12

28 Aug '12

Currently the HDA drivers have a Kconfig variable SND_HDA_INPUT_JACK which disables jack reporting via the ALSA core jack API. This is bad since (modulo the current problems with the HDA-specific reporting API) it means that all reporting mechanisms will get disabled, not just input. It's also not good that this is open coded in an individual driver - if it's useful to have this control it's going to be equally useful for any driver with jack detection support so should be handled in the core. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/pci/hda/Kconfig | 9 +-------- sound/pci/hda/hda_jack.c | 8 -------- 2 files changed, 1 insertion(+), 16 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 194d625..2c7f626 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -2,6 +2,7 @@ menuconfig SND_HDA_INTEL tristate "Intel HD Audio" select SND_PCM select SND_VMASTER + select SND_JACK if INPUT=y || INPUT=SND select SND_KCTL_JACK help Say Y here to include support for Intel "High Definition @@ -62,14 +63,6 @@ config SND_HDA_INPUT_BEEP_MODE Set 1 to always enable the digital beep interface for HD-audio by default. -config SND_HDA_INPUT_JACK - bool "Support jack plugging notification via input layer" - depends on INPUT=y || INPUT=SND - select SND_JACK - help - Say Y here to enable the jack plugging notification via - input layer. - config SND_HDA_PATCH_LOADER bool "Support initialization patch loading for HD-audio" depends on EXPERIMENTAL diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index aaccc02..0760d75 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -109,7 +109,6 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_new); void snd_hda_jack_tbl_clear(struct hda_codec *codec) { -#ifdef CONFIG_SND_HDA_INPUT_JACK /* free jack instances manually when clearing/reconfiguring */ if (!codec->bus->shutdown && codec->jacktbl.list) { struct hda_jack_tbl *jack = codec->jacktbl.list; @@ -119,7 +118,6 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) snd_device_free(codec->bus->card, jack->jack); } } -#endif snd_array_free(&codec->jacktbl); } @@ -225,16 +223,13 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) continue; state = get_jack_plug_state(jack->pin_sense); snd_kctl_jack_report(codec->bus->card, jack->kctl, state); -#ifdef CONFIG_SND_HDA_INPUT_JACK if (jack->jack) snd_jack_report(jack->jack, state ? jack->type : 0); -#endif } } EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync); -#ifdef CONFIG_SND_HDA_INPUT_JACK /* guess the jack type from the pin-config */ static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { @@ -261,7 +256,6 @@ static void hda_free_jack_priv(struct snd_jack *jack) jacks->nid = 0; jacks->jack = NULL; } -#endif /** * snd_hda_jack_add_kctl - Add a kctl for the given pin @@ -292,7 +286,6 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, state = snd_hda_jack_detect(codec, nid); snd_kctl_jack_report(codec->bus->card, kctl, state); -#ifdef CONFIG_SND_HDA_INPUT_JACK if (!phantom_jack) { jack->type = get_input_jack_type(codec, nid); err = snd_jack_new(codec->bus->card, name, jack->type, @@ -303,7 +296,6 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->jack->private_free = hda_free_jack_priv; snd_jack_report(jack->jack, state ? jack->type : 0); } -#endif return 0; } -- 1.7.10.4

2 2

[alsa-devel] ASoC: SAMSUNG: Change Kconfig to support all EXYNOSs
by 박상수 28 Aug '12

28 Aug '12

All EXYNOSs use SND_SOC_SAMSUNG configuration. This patch change Kconfig to support all EXYNOSs like EXYNOS4 and EXYNOS5. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/samsung/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f17dd25..9e64ae6 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help -- 1.7.1

3 4

[alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Ricardo Neri 27 Aug '12

27 Aug '12

Hello! I have been working on prototypes for the ASoC OMAP HDMI audio driver to propagate events from the HDMI output (e.g., display getting enabled/disabled/suspended). This for the users of the driver to react to such events. For instance, if the display is disabled or disconected, audio could be stopped, rerouted or whatever other decision the user makes. This is needed because, if, for instance, the HDMI IP goes off, audio will stall and the audio users will only see a "playback write error (DMA or IRQ trouble?)" In my prototypes I have used snd_soc_jack for this purpose and I have some questions: *I see snd_soc_jack is used mostly for headsets and microphones with actual external mechanical connections. Strictly, in my case I propagate events originated by the OMAP display driver (changes in the power state), and not from external events. Some of these events are generated from an actual HDMI cable connection/disconnection, though. *Maybe the event should be propagated by omapdss/omapdrm/drm and the entity in charge of the audio policy should listen those events instead. *I do see SND_JACK_VIDEOOUT and SND_JACK_AVOUT types so maybe it is feasible for an audio driver to report events from an AV output. I was wondering about how much sense does it make to you guys use a snd_soc_jack in this case? Thanks in advance for your comments! Ricardo

6 9

[alsa-devel] Kernel module build workflow (Re: Logitech USB headset not working in 3.6-rc3)
by Takashi Iwai 27 Aug '12

27 Aug '12

At Sat, 25 Aug 2012 07:16:21 -0500, Bruno Wolff III wrote: > > On Sat, Aug 25, 2012 at 08:13:27 -0400, > Josh Boyer <jwboyer(a)redhat.com> wrote: > >On Sat, Aug 25, 2012 at 07:07:40AM -0500, Bruno Wolff III wrote: > >> On Sat, Aug 25, 2012 at 14:02:51 +0200, > >> Daniel Mack <zonque(a)gmail.com> wrote: > >> > > >> >Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix > >> >scheduling-while-atomic bug in PCM capture stream") and see if that > >> > >> I can try that, but it takes a long time to build a new kernel on my > >> old hardware. > > > >I started a scratch build of a kernel with that patch reverted (via > >patch -R). You'll find it here when it finishes: > > > >http://koji.fedoraproject.org/koji/taskinfo?taskID=4421817 > > > >josh > > I'll test it when it finishes. Thanks. It takes me a long time to build > kernels locally, and I wasn't sure how to do a scratch build on koji without > committing stuff remotely. Actually you don't have to rebuild the whole kernel at all just for debugging this usb-audio bug. The workflow below is what I do often for debugging a driver problem on a distro kernel. 1. Prepare the compile-ready kernel source tree corresponding to your running kernel. This may depend on distro; in the case of SUSE, kernel-default-devel.rpm must be installed for kernel-default.rpm, for example. 2. Copy sound/usb subdirectory of the kernel source tree locally anywhere else, say, $HOME/usb. It can be a normal user directory. 3. Apply the patch to that local directory. 4. Build modules as a normal user like % make -C /lib/modules/$(uname -r)/build M=$HOME/usb modules 5. Make an extra update directory in the module directory once, e.g. # mkdir /lib/modules/$(uname -r)/updates 6. Copy *.ko files to there # cp $HOME/usb/*.ko /lib/modules/$(uname -r)/updates 7. Run depmod once # /sbin/depmod -a 8. Reload snd-usb-audio module. If you want to take back to the original module, just remove updates module directory and run depmod again. Pretty convenient, eh? HTH, Takashi

2 2

Re: [alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Use module_spi_driver
by Mark Brown 27 Aug '12

27 Aug '12

On Mon, Aug 27, 2012 at 05:00:26PM +0530, Sachin Kamat wrote: > module_spi_driver makes the code simpler by eliminating > module_init and module_exit calls. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH - pcxhr 1/1] ALSA: pcxhr: Add 8 new sound cards
by Takashi Iwai 27 Aug '12

27 Aug '12

At Mon, 27 Aug 2012 09:03:55 +0200, Markus Bollinger wrote: > > Hi Takashi, > > sorry, I had problems to get git-send-email working, so I tried it on > another way, but it seems that was not the right one.... > Please find the patch attached. Generated from alsa.kmirror.git. Thanks, applied now. Takashi

1 0

Re: [alsa-devel] [pulseaudio-discuss] Documentation for how rewinding works
by Raymond Yau 27 Aug '12

27 Aug '12

2012-8-19 下午7:17 於 "Tanu Kaskinen" <tanuk(a)iki.fi> 寫道： > > Hi, > > While reviewing a rewind related patch, I noticed that I don't > understand the rewinding code thoroughly, even though I have studied it > a few times in the past. I decided that I should write a wiki page for > the code so that working with it would be easier in the future for me > and everybody else. > > So, here's the result: > http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer… > > While doing that, I found quite a few bugs, which are noted on that > page. I don't plan fixing them any time soon (help with that would be > very welcome), so I will just file bugs about the issues so that they > are not forgotten. > it seem that the alsa sink is more complicated than your figure. Refer to figure 14 HD Audio DMA and buffering , section 4.6 Energy Efficient HA audio mechanism most sound cards have fixed size FIFO buffer instead of dynamic FIFO and minimum transfer size(brust) similar to hda controller which snd_pcm_rewind cannot rewind any audio data in those first in first out buffer. http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa?i… 1) do the server really need to rewind when there is only one pulse client ? 2) does pulseaudio add this rewind safeguard delay to the latency ?

2 3

[alsa-devel] REGRESSION: Webcam mic for 145f:015b not longer working with 1.0.24 and above (OK with 1.0.23)
by Kostas 27 Aug '12

27 Aug '12

Hi All, I have a 'Trust Webcam' (USB ID: 145f:015b) that comes with a microphone. The microphone works with Alsa 1.0.23 but it does not work with the latest Alsa 1.0.25 neither with Alsa 1.0.24. Apparently this is a regression. What shall I do to figure out the source of the problem? Here are attached my alsa-infos, with Alsa 1.0.23 (working), 1.0.24 (NOT working) and 1.0.25 (NOT working). Thanks! 1.0.23 http://www.alsa-project.org/db/?f=a4e2c86d93ff67c5d6acd4933c76f47a60cf51ee 1.0.24 http://www.alsa-project.org/db/?f=9aa13f70ec169011b9dfc5625370ce3e6f7827a1 1.0.25 http://www.alsa-project.org/db/?f=c3f63b69655b379e3eec54ab00e63b186a677422

2 1

[alsa-devel] Driver for HT|OMEGA eCLARO (C-Media 8788 ***PCIe***)
by Aaron Fulkerson 27 Aug '12

27 Aug '12

Hello and nice to meet you all. This will be my first post if the previous one did not work. The problem is this: there is not a working driver for the HT|OMEGA eCLARO. The card is based on the C-Media 8788 chip for which there is an existing driver, however, due to the fact that the eCLARO is a PCIe-based card, the driver doesn't function properly. What are the chances of a working driver being written for this card?

1 0

[alsa-devel] [PATCH v2 0/4] ASoC: omap-mcbsp: Sidetone related changes
by Peter Ujfalusi 25 Aug '12

25 Aug '12

Hello, Changes since v1: - Macro for the ST control has been fixed - New patch to combine the st channel volume set/get macros Mark: this series depends on the McBSP DT support (to be precise patch 3 and 4 has dependency). Intro mail from v1: When we boot with DT blob the sidetone is not supported at the moment. Prepare the McBSP code for the time when we are going to be able to support ST with DT booted kernel. Meanwhile do not block the audio card if the st_data is missing from the port, but print a warning about it. This will allow us to debug the DT booted kernels with audio. Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-mcbsp: Check mcbsp->id instead of cpu_dai->id when adding ST controls ASoC: omap-mcbsp: Only print warning if the st_data is missing for the port ASoC: omap-mcbsp: Use macro to create the McBSP2/3 ST controls ASoC: omap-mcbsp: Single macro for st channel volume set/get sound/soc/omap/omap-mcbsp.c | 64 ++++++++++++++++++------------------------- 1 files changed, 27 insertions(+), 37 deletions(-) -- 1.7.8.6

3 8

[alsa-devel] [PATCH] ASoC: wm0010: Move resource acquisition to device probe
by Mark Brown 25 Aug '12

25 Aug '12

This is more idimatic for modern drivers. Also fix a couple of return codes while we're at it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 54 ++++++++++++++++++++++----------------------- 1 file changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 30ec0bd..5f99148 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -799,14 +799,35 @@ static irqreturn_t wm0010_irq(int irq, void *data) static int wm0010_probe(struct snd_soc_codec *codec) { struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); - struct spi_device *spi = to_spi_device(wm0010->dev); + + wm0010->codec = codec; + + return 0; +} + +static int __devinit wm0010_spi_probe(struct spi_device *spi) +{ unsigned long flags; unsigned long gpio_flags; int ret; int trigger; int irq; + struct wm0010_priv *wm0010; - wm0010->codec = codec; + wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), + GFP_KERNEL); + if (!wm0010) + return -ENOMEM; + + mutex_init(&wm0010->lock); + spin_lock_init(&wm0010->irq_lock); + + spi_set_drvdata(spi, wm0010); + wm0010->dev = &spi->dev; + + if (dev_get_platdata(&spi->dev)) + memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), + sizeof(wm0010->pdata)); init_completion(&wm0010->boot_completion); @@ -850,7 +871,7 @@ static int wm0010_probe(struct snd_soc_codec *codec) } } else { dev_err(wm0010->dev, "No reset GPIO configured\n"); - return ret; + return -EINVAL; } irq = spi->irq; @@ -862,9 +883,11 @@ static int wm0010_probe(struct snd_soc_codec *codec) ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, "wm0010", wm0010); - if (ret) + if (ret) { dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", irq, ret); + return ret; + } wm0010->irq = irq; if (spi->max_speed_hz) @@ -876,29 +899,6 @@ static int wm0010_probe(struct snd_soc_codec *codec) wm0010->state = WM0010_POWER_OFF; spin_unlock_irqrestore(&wm0010->irq_lock, flags); - return 0; -} - -static int __devinit wm0010_spi_probe(struct spi_device *spi) -{ - struct wm0010_priv *wm0010; - int ret; - - wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), - GFP_KERNEL); - if (!wm0010) - return -ENOMEM; - - mutex_init(&wm0010->lock); - spin_lock_init(&wm0010->irq_lock); - - spi_set_drvdata(spi, wm0010); - wm0010->dev = &spi->dev; - - if (dev_get_platdata(&spi->dev)) - memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), - sizeof(wm0010->pdata)); - ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm0010, wm0010_dai, ARRAY_SIZE(wm0010_dai)); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm0010: Add dummy widget for CLKIN
by Mark Brown 25 Aug '12

25 Aug '12

Make it easier to integrate the management of the clock supplying the WM0010 with DAPM by providing a dummy supply widget which supplies the interface widgets, this can be connected to clock outputs by the machines. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f820b57..b6f7097 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -118,9 +118,18 @@ struct wm0010_spi_msg { size_t len; }; +static const struct snd_soc_dapm_widget wm0010_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("CLKIN", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { { "SDI2 Capture", NULL, "SDI1 Playback" }, { "SDI1 Capture", NULL, "SDI2 Playback" }, + + { "SDI1 Capture", NULL, "CLKIN" }, + { "SDI2 Capture", NULL, "CLKIN" }, + { "SDI1 Playback", NULL, "CLKIN" }, + { "SDI2 Playback", NULL, "CLKIN" }, }; static const char *wm0010_state_to_str(enum wm0010_state state) @@ -718,6 +727,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .set_bias_level = wm0010_set_bias_level, .set_sysclk = wm0010_set_sysclk, + .dapm_widgets = wm0010_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm0010_dapm_widgets), .dapm_routes = wm0010_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), }; -- 1.7.10.4

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Takashi Iwai 25 Aug '12

25 Aug '12

At Sat, 25 Aug 2012 08:17:58 -0400, Josh Boyer wrote: > > On Sat, Aug 25, 2012 at 02:13:58PM +0200, Daniel Mack wrote: > > On 25.08.2012 14:07, Bruno Wolff III wrote: > > > On Sat, Aug 25, 2012 at 14:02:51 +0200, > > > Daniel Mack <zonque(a)gmail.com> wrote: > > >> > > >> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix > > >> scheduling-while-atomic bug in PCM capture stream") and see if that > > > > > > I can try that, but it takes a long time to build a new kernel on my > > > old hardware. > > > > > >> helps? If not, can you summarize again which kernels still work for you > > >> and which don't? > > > > > > The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The earliest that > > > doesn't work is 3.6.0-0.rc2.git2.1.fc18. > > > > > > > The report you sent doesn't look like it could be caused by e9ba389c5. > > It fixes a kernel Ooops. But as it is the only relevant patch in that > > area, it would be interesting if reverting it fixes anything. > > Yep, agreed. If this revert kernel doesn't work, we're likely down to a > git bisect, Bruno. Yes, please. I'm currently traveling for KS and Plumbers, so I cannot debug it deeply, unfortunately. If the culprit were really my patch, I still don't figure out why it breaks. In that case, it must be really a subtle timing issue... thanks, Takashi

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Re: [alsa-devel] [PATCH - pcxhr 1/1] ALSA: pcxhr: Add 8 new sound cards
by Takashi Iwai 25 Aug '12

25 Aug '12

At Fri, 24 Aug 2012 16:22:57 +0200, Markus Bollinger wrote: > > add new sound cards VX442HR VX442e PCX442HR PCX442e VX822HR VX822e PCX822HR > and PCX822e > > Signed-off-by: Markus Bollinger <bollinger(a)digigram.com> The changes look OK, but the patch file embedded in the mail looks broken. If it's difficult to embed the patch, could you use an attachment? thanks, Takashi > > diff --git a/pci/pcxhr/pcxhr.c b/pci/pcxhr/pcxhr.c > index e3ac1f7..be4f145 100644 > --- a/pci/pcxhr/pcxhr.c > +++ b/pci/pcxhr/pcxhr.c > @@ -91,6 +91,14 @@ enum { > PCI_ID_PCX924E, > PCI_ID_PCX924HRMIC, > PCI_ID_PCX924E_MIC, > + PCI_ID_VX442HR, > + PCI_ID_PCX442HR, > + PCI_ID_VX442E, > + PCI_ID_PCX442E, > + PCI_ID_VX822HR, > + PCI_ID_PCX822HR, > + PCI_ID_VX822E, > + PCI_ID_PCX822E, > PCI_ID_LAST > }; > > @@ -121,6 +129,14 @@ static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { > { 0x10b5, 0x9056, 0x1369, 0xbb21, 0, 0, PCI_ID_PCX924E, }, > { 0x10b5, 0x9056, 0x1369, 0xbf01, 0, 0, PCI_ID_PCX924HRMIC, }, > { 0x10b5, 0x9056, 0x1369, 0xbf21, 0, 0, PCI_ID_PCX924E_MIC, }, > + { 0x10b5, 0x9656, 0x1369, 0xd001, 0, 0, PCI_ID_VX442HR, }, > + { 0x10b5, 0x9656, 0x1369, 0xd101, 0, 0, PCI_ID_PCX442HR, }, > + { 0x10b5, 0x9056, 0x1369, 0xd021, 0, 0, PCI_ID_VX442E, }, > + { 0x10b5, 0x9056, 0x1369, 0xd121, 0, 0, PCI_ID_PCX442E, }, > + { 0x10b5, 0x9656, 0x1369, 0xd201, 0, 0, PCI_ID_VX822HR, }, > + { 0x10b5, 0x9656, 0x1369, 0xd301, 0, 0, PCI_ID_PCX822HR, }, > + { 0x10b5, 0x9056, 0x1369, 0xd221, 0, 0, PCI_ID_VX822E, }, > + { 0x10b5, 0x9056, 0x1369, 0xd321, 0, 0, PCI_ID_PCX822E, }, > { 0, } > }; > > @@ -160,6 +176,14 @@ static struct board_parameters pcxhr_board_params[] = { > [PCI_ID_PCX924E] = { "PCX924e", 1, 1, 5, 44 }, > [PCI_ID_PCX924HRMIC] = { "PCX924HR-Mic", 1, 1, 5, 44 }, > [PCI_ID_PCX924E_MIC] = { "PCX924e-Mic", 1, 1, 5, 44 }, > +[PCI_ID_VX442HR] = { "VX442HR", 2, 2, 0, 41 }, > +[PCI_ID_PCX442HR] = { "PCX442HR", 2, 2, 0, 41 }, > +[PCI_ID_VX442E] = { "VX442e", 2, 2, 1, 41 }, > +[PCI_ID_PCX442E] = { "PCX442e", 2, 2, 1, 41 }, > +[PCI_ID_VX822HR] = { "VX822HR", 4, 1, 2, 42 }, > +[PCI_ID_PCX822HR] = { "PCX822HR", 4, 1, 2, 42 }, > +[PCI_ID_VX822E] = { "VX822e", 4, 1, 3, 42 }, > +[PCI_ID_PCX822E] = { "PCX822e", 4, 1, 3, 42 }, > }; > > /* boards without hw AES1 and SRC onboard are all using fw_file_set==4 */ > diff --git a/pci/pcxhr/pcxhr_hwdep.c b/pci/pcxhr/pcxhr_hwdep.c > index ec1587c..bf207e3 100644 > --- a/pci/pcxhr/pcxhr_hwdep.c > +++ b/pci/pcxhr/pcxhr_hwdep.c > @@ -66,10 +66,10 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) > err = pcxhr_send_msg(mgr, &rmh); > if (err) > return err; > - /* test 8 or 12 phys out */ > - if ((rmh.stat[0] & MASK_FIRST_FIELD) != mgr->playback_chips * 2) > + /* test 4, 8 or 12 phys out */ > + if ((rmh.stat[0] & MASK_FIRST_FIELD) < mgr->playback_chips * 2) > return -EINVAL; > - /* test 8 or 2 phys in */ > + /* test 4, 8 or 2 phys in */ > if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) < > mgr->capture_chips * 2) > return -EINVAL; > -- > 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm0010: Enable 44.1kHz support
by Mark Brown 25 Aug '12

25 Aug '12

With appropriate clocking configuration the WM0010 driver supports 44.1kHz audio; enable that. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index e1315e0..f820b57 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -722,7 +722,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), }; -#define WM0010_RATES (SNDRV_PCM_RATE_48000) +#define WM0010_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -- 1.7.10.4

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 25 Aug '12

25 Aug '12

On 25.08.2012 14:07, Bruno Wolff III wrote: > On Sat, Aug 25, 2012 at 14:02:51 +0200, > Daniel Mack <zonque(a)gmail.com> wrote: >> >> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >> scheduling-while-atomic bug in PCM capture stream") and see if that > > I can try that, but it takes a long time to build a new kernel on my > old hardware. > >> helps? If not, can you summarize again which kernels still work for you >> and which don't? > > The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The earliest that > doesn't work is 3.6.0-0.rc2.git2.1.fc18. > The report you sent doesn't look like it could be caused by e9ba389c5. It fixes a kernel Ooops. But as it is the only relevant patch in that area, it would be interesting if reverting it fixes anything. Btw - thanks a lot for testing -rc kernels, much appreciated! Daniel

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 25 Aug '12

25 Aug '12

On 25.08.2012 13:54, Bruno Wolff III wrote: > On Sat, Aug 25, 2012 at 06:17:49 -0500, > Bruno Wolff III <bruno(a)wolff.to> wrote: >> >> I see this behavior on three different computers. Rebooting into an old >> enough kernel gets things working again. The older kernels also have >> older initramfs images. I could rebuild those to see if there is some >> issue there instead of the kernels themselves? > > I tested one of the older kernels (where the headset still worked) with a > rebuilt initramfs and the headset still worked. So it likely isn't an issue > with something in the initramfs images. > Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") and see if that helps? If not, can you summarize again which kernels still work for you and which don't? Daniel

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[alsa-devel] multistream singlechanel DACs
by Rafal Powierski 25 Aug '12

25 Aug '12

Developers, how to enable multiple playback streaming?: I am manipulating bios verb pin table, for special analog hardware, using ALC886 and kubuntu / 64bit. I need a number of separate stereo (only stereo) i/o's I have correctly configured inputs: 4. single stereo adc port-C 1Ah 5. single spdif input 1Fh I have configured 3 independent stereo outputs, but ALSA is joining them into one pcmC1D0p device. 1.port-A,15h 2.port-E,1Bh 3.port-D,14h I have tried this on win7 and I got all associations as a separate devices. in my kubuntu 12.4LTS I have only one playback device. Mixer is showing even more then needed number of volume controls, But there is only one playback device. it seams, all 3 outputs are connected to single DAC. I have tried to buildup more input devices - this is working good. I can have up to 3 (pcmC1D0c, pcmC1D1c pcmC1D2c) I can also have 2 outputs if I enable spdif-out. But there is no separate DAC assignment to separate devices resulting from my config. Is it possible to get independent DACs devices under ALSA? or else is no such configuration possible and I have to use single multiple channel device?

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 25 Aug '12

25 Aug '12

Hi, On 24.08.2012 21:08, Josh Boyer wrote: > We've had a report[1] that the Logitech USB headset 0003:046D:0A0C isn't > working with 3.6-rc3. It seems the last working kernel was based on > commit 10c63c9, and it first stopped working with a kernel based on > commit 23dcfa6. There are only a few ALSA commits between those > revisions, so hopefully this is something that is fairly easy to > identify. The only commit to USB audio in that set is: [...] > !!ALSA/HDA dmesg > !!------------------ > > [ 38.190306] SELinux: initialized (dev configfs, type configfs), uses genfs_contexts > [ 38.229616] snd_hda_intel 0000:00:1b.0: irq 66 for MSI/MSI-X > [ 38.270699] ALSA sound/usb/mixer.c:866 6:0: cannot get min/max values for control 2 (id 6) > [ 38.274097] ALSA sound/usb/mixer.c:866 1:0: cannot get min/max values for control 2 (id 1) > [ 38.276753] ALSA sound/usb/mixer.c:866 2:0: cannot get min/max values for control 2 (id 2) > [ 38.279322] ALSA sound/pci/hda/hda_auto_parser.c:322 autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) type:line > [ 38.279326] ALSA sound/pci/hda/hda_auto_parser.c:326 speaker_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 38.279329] ALSA sound/pci/hda/hda_auto_parser.c:330 hp_outs=1 (0xd/0x0/0x0/0x0/0x0) > [ 38.279331] ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0 > [ 38.279333] ALSA sound/pci/hda/hda_auto_parser.c:335 inputs: > [ 38.279336] ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x10 > [ 38.279342] ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0xf > [ 38.285983] usbcore: registered new interface driver snd-usb-audio Also, according to this dmesg, the device is probed just fine (which the commit you mentioned wouldn't change anything about though). So what does "isn't working anymore" refer to precisely? Are there any more dmesg entries generated once the stream is started? Daniel

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 24 Aug '12

24 Aug '12

On Fri, Aug 24, 2012 at 9:08 PM, Josh Boyer <jwboyer(a)redhat.com> wrote: > Hi All, > > We've had a report[1] that the Logitech USB headset 0003:046D:0A0C isn't > working with 3.6-rc3. It seems the last working kernel was based on > commit 10c63c9, and it first stopped working with a kernel based on > commit 23dcfa6. There are only a few ALSA commits between those > revisions, so hopefully this is something that is fairly easy to > identify. The only commit to USB audio in that set is: > > commit e9ba389c5ffc4dd29dfe17e00e48877302111135 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Wed Aug 15 12:32:00 2012 +0200 > > ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream > > > I've CC'd the reporter and attached the alsa-info is below. Does it work again once you revert that commit? Daniel > [1] https://bugzilla.redhat.com/show_bug.cgi?id=851619 > > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.60 > !!################################ > > !!Script ran on: Fri Aug 24 18:35:42 UTC 2012 > > > !!Linux Distribution > !!------------------ > > Fedora release 18 (Rawhide) Fedora release 18 (Rawhide) NAME=Fedora ID=fedora PRETTY_NAME="Fedora 18 (Rawhide)" CPE_NAME="cpe:/o:fedoraproject:fedora:18" Fedora release 18 (Rawhide) Fedora release 18 (Rawhide) > > > !!DMI Information > !!--------------- > > Manufacturer: Dell Inc. > Product Name: Precision WorkStation 690 > Product Version: > > > !!Kernel Information > !!------------------ > > Kernel release: 3.6.0-0.rc3.git0.1.fc18.x86_64 > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.25 > Library version: 1.0.25 > Utilities version: 1.0.25 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > snd_usb_audio > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > aRts: > Installed - Yes (/usr/bin/artsd) > Running - No > > Jack: > Installed - Yes (/usr/bin/jackd) > Running - No > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xfcffc000 irq 66 > 1 [Headset ]: USB-Audio - Logitech USB Headset > Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 631xESB/632xESB High Definition Audio Controller (rev 09) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:269a (rev 09) > Subsystem: 1028:01c0 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > align_buffer_size : -1 > bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : -1 > id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > single_cmd : N > snoop : Y > > !!Module: snd_usb_audio > async_unlink : Y > device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > ignore_ctl_error : N > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > nrpacks : 8 > pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: SigmaTel STAC9200 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x83847690 > Subsystem Id: 0x102801c0 > Revision Id: 0x102201 > No Modem Function Group found > Default PCM: > rates [0x7e0]: 44100 48000 88200 96000 176400 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 > GPIO: io=4, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0xd0401: Stereo > Device: name="STAC92xx Analog", type="Audio", device=0 > Converter: stream=8, channel=0 > Power states: > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo > Device: name="STAC92xx Analog", type="Audio", device=0 > Converter: stream=4, channel=0 > SDI-Select: 0 > Power states: > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x0a > Processing caps: benign=0, ncoeff=0 > Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > PCM: > rates [0x160]: 44100 48000 96000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Connection: 1 > 0x08 > Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x1e0]: 44100 48000 88200 96000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital > Delay: 3 samples > Node 0x07 [Audio Selector] wcaps 0x300901: Stereo R/L > Connection: 3 > 0x02* 0x08 0x0a > Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital > Pincap 0x00010024: IN EAPD Detect > EAPD 0x0: > Pin Default 0x400001f0: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: > Power: setting=D0, actual=D0 > Delay: 3 samples > Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x00000010: OUT > Pin Default 0x400001f1: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 2 > 0x05* 0x0a > Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Control: name="Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Capture Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x07 0x07] > Connection: 1 > 0x0c > Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out > Control: name="PCM Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="PCM Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: N/A > Amp-Out vals: [0x9f 0x9f] > Connection: 1 > 0x07 > Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Control: name="Mux Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 5 > 0x10* 0x0f 0x0e 0x0d 0x12 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Control: name="Front Headphone Jack", index=0, device=0 > Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense > Pin Default 0x0221401f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=01, enabled=1 > Connection: 1 > 0x0b > Node 0x0e [Pin Complex] wcaps 0x400181: Stereo > Control: name="Line Out Jack", index=0, device=0 > Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense > Pin Default 0x01014010: [Jack] Line Out at Ext Rear > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=02, enabled=1 > Connection: 1 > 0x0b > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Control: name="Line Jack", index=0, device=0 > Pincap 0x00000037: IN OUT Detect Trigger ImpSense > Pin Default 0x01813020: [Jack] Line In at Ext Rear > Conn = 1/8, Color = Blue > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x20: IN > Unsolicited: tag=04, enabled=1 > Connection: 1 > 0x0b > Node 0x10 [Pin Complex] wcaps 0x400181: Stereo > Control: name="Mic Jack", index=0, device=0 > Pincap 0x00001737: IN OUT Detect Trigger ImpSense > Vref caps: HIZ 50 GRD 80 > Pin Default 0x02a19021: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0x1 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=03, enabled=1 > Connection: 1 > 0x0b > Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out > Control: name="LFE Playback Volume", index=0, device=0 > ControlAmp: chs=1, dir=Out, idx=0, ofs=0 > Control: name="LFE Playback Switch", index=0, device=0 > ControlAmp: chs=1, dir=Out, idx=0, ofs=0 > Control: name="Speaker Phantom Jack", index=0, device=0 > Amp-Out caps: N/A > Amp-Out vals: [0x00] > Pincap 0x00000010: OUT > Pin Default 0x90100140: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x4, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 1 > 0x13 > Node 0x12 [Pin Complex] wcaps 0x400001: Stereo > Pincap 0x00000020: IN > Pin Default 0x400001f2: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x13 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x07 > Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > --endcollapse-- > > > !!USB Mixer information > !!--------------------------- > --startcollapse-- > > USB Mixer: usb_id=0x046d0a0c, ctrlif=0, ctlerr=0 > Card: Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed > Unit: 1 > Control: name="Speaker Playback Volume", index=0 > Info: id=1, control=2, cmask=0x3, channels=2, type="S16" > Volume: min=-10496, max=768, dBmin=-4100, dBmax=300 > Unit: 1 > Control: name="Speaker Playback Switch", index=0 > Info: id=1, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > Unit: 2 > Control: name="Mic Capture Volume", index=0 > Info: id=2, control=2, cmask=0x0, channels=1, type="S16" > Volume: min=4096, max=7424, dBmin=1600, dBmax=2900 > Unit: 2 > Control: name="Mic Capture Switch", index=0 > Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > Unit: 6 > Control: name="Mic Playback Volume", index=0 > Info: id=6, control=2, cmask=0x0, channels=1, type="S16" > Volume: min=-10496, max=-3072, dBmin=-4100, dBmax=-1200 > Unit: 6 > Control: name="Mic Playback Switch", index=0 > Info: id=6, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw----+ 1 root audio 116, 8 Aug 24 12:18 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 4 Aug 24 12:18 /dev/snd/controlC1 > crw-rw----+ 1 root audio 116, 7 Aug 24 12:18 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 6 Aug 24 12:19 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 5 Aug 24 12:19 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 3 Aug 24 12:19 /dev/snd/pcmC1D0c > crw-rw----+ 1 root audio 116, 2 Aug 24 12:30 /dev/snd/pcmC1D0p > crw-rw----+ 1 root audio 116, 1 Aug 24 12:18 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 Aug 24 12:18 /dev/snd/timer > > /dev/snd/by-id: > total 0 > drwxr-xr-x. 2 root root 60 Aug 24 12:18 . > drwxr-xr-x. 4 root root 260 Aug 24 12:18 .. > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 usb-Logitech_Logitech_USB_Headset-00 -> ../controlC1 > > /dev/snd/by-path: > total 0 > drwxr-xr-x. 2 root root 80 Aug 24 12:18 . > drwxr-xr-x. 4 root root 260 Aug 24 12:18 .. > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 pci-0000:00:1b.0 -> ../controlC0 > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 pci-0000:00:1d.7-usb-0:6.4:1.0 -> ../controlC1 > > > !!ALSA configuration files > !!------------------------ > > !!System wide config file (/etc/asound.conf) > > # > # Place your global alsa-lib configuration here... > # > > > !!Aplay/Arecord output > !!------------ > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: Headset [Logitech USB Headset], device 0: USB Audio [USB Audio] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: Headset [Logitech USB Headset], device 0: USB Audio [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [Intel] > > Card hw:0 'Intel'/'HDA Intel at 0xfcffc000 irq 66' > Mixer name : 'SigmaTel STAC9200' > Components : 'HDA:83847690,102801c0,00102201' > Controls : 15 > Simple ctrls : 6 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 31 > Mono: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'PCM',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 31 [100%] [0.00dB] [off] > Front Right: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'LFE',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 31 > Mono: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 7 [47%] [10.50dB] [on] > Front Right: Capture 7 [47%] [10.50dB] [on] > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Line' > Item0: 'Mic' > Simple mixer control 'Mux',0 > Capabilities: cvolume penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 4 > Front Left: Capture 0 [0%] [0.00dB] > Front Right: Capture 0 [0%] [0.00dB] > > !!-------Mixer controls for card 1 [Headset] > > Card hw:1 'Headset'/'Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed' > Mixer name : 'USB Mixer' > Components : 'USB046d:0a0c' > Controls : 6 > Simple ctrls : 2 > Simple mixer control 'Speaker',0 > Capabilities: pvolume pswitch pswitch-joined penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 44 > Mono: > Front Left: Playback 35 [80%] [-6.00dB] [on] > Front Right: Playback 35 [80%] [-6.00dB] [on] > Simple mixer control 'Mic',0 > Capabilities: pvolume pvolume-joined cvolume cvolume-joined pswitch pswitch-joined cswitch cswitch-joined penum > Playback channels: Mono > Capture channels: Mono > Limits: Playback 0 - 29 Capture 0 - 13 > Mono: Playback 0 [0%] [-41.00dB] [off] Capture 13 [100%] [29.00dB] [on] > > > !!Alsactl output > !!------------- > > --startcollapse-- > state.Intel { > control.1 { > iface MIXER > name 'PCM Playback Volume' > value.0 31 > value.1 31 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 31' > dbmin -9999999 > dbmax 0 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.2 { > iface MIXER > name 'PCM Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.3 { > iface MIXER > name 'Capture Volume' > value.0 7 > value.1 7 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 15' > dbmin 0 > dbmax 2250 > dbvalue.0 1050 > dbvalue.1 1050 > } > } > control.4 { > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.5 { > iface MIXER > name 'LFE Playback Volume' > value 31 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 31' > dbmin -4650 > dbmax 0 > dbvalue.0 0 > } > } > control.6 { > iface MIXER > name 'LFE Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.7 { > iface MIXER > name 'Mux Capture Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 4' > dbmin 0 > dbmax 4000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.8 { > iface MIXER > name 'Input Source' > value Mic > comment { > access 'read write' > type ENUMERATED > count 1 > item.0 Mic > item.1 Line > } > } > control.9 { > iface MIXER > name 'Master Playback Volume' > value 31 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 31' > dbmin -9999999 > dbmax 0 > dbvalue.0 0 > } > } > control.10 { > iface MIXER > name 'Master Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.11 { > iface CARD > name 'Line Out Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.12 { > iface CARD > name 'Front Headphone Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.13 { > iface CARD > name 'Speaker Phantom Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.14 { > iface CARD > name 'Mic Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.15 { > iface CARD > name 'Line Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > } > state.Headset { > control.1 { > iface MIXER > name 'Mic Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.2 { > iface MIXER > name 'Mic Playback Volume' > value 0 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 29' > dbmin -4100 > dbmax -1200 > dbvalue.0 -4100 > } > } > control.3 { > iface MIXER > name 'Speaker Playback Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.4 { > iface MIXER > name 'Speaker Playback Volume' > value.0 35 > value.1 35 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 44' > dbmin -4100 > dbmax 300 > dbvalue.0 -600 > dbvalue.1 -600 > } > } > control.5 { > iface MIXER > name 'Mic Capture Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.6 { > iface MIXER > name 'Mic Capture Volume' > value 13 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 13' > dbmin 1600 > dbmax 2900 > dbvalue.0 2900 > } > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > fuse > nf_conntrack_ipv4 > nf_defrag_ipv4 > xt_state > nf_conntrack > snd_hda_codec_idt > snd_usb_audio > snd_hda_intel > snd_usbmidi_lib > snd_hda_codec > snd_rawmidi > snd_hwdep > iTCO_wdt > iTCO_vendor_support > lpc_ich > i5000_edac > snd_seq > snd_seq_device > edac_core > snd_pcm > mfd_core > serio_raw > ppdev > snd_page_alloc > snd_timer > coretemp > snd > i5k_amb > soundcore > parport_pc > dcdbas > microcode > shpchp > parport > i2c_i801 > vhost_net > tun > macvtap > macvlan > kvm_intel > kvm > uinput > binfmt_misc > xts > gf128mul > dm_crypt > raid1 > radeon > i2c_algo_bit > drm_kms_helper > ttm > drm > tg3 > mptsas > mptscsih > mptbase > i2c_core > scsi_transport_sas > sunrpc > be2iscsi > bnx2i > cnic > uio > cxgb4i > cxgb4 > cxgb3 > mdio > libcxgbi > libiscsi_tcp > qla4xxx > iscsi_boot_sysfs > libiscsi > scsi_transport_iscsi > > > !!Sysfs Files > !!----------- > > /sys/class/sound/hwC0D0/init_pin_configs: > 0x08 0x40f000f0 > 0x09 0x40f000f1 > 0x0d 0x0221101f > 0x0e 0x01014010 > 0x0f 0x0181302e > 0x10 0x02a11020 > 0x11 0x40f000f2 > 0x12 0x40f000f3 > > /sys/class/sound/hwC0D0/driver_pin_configs: > 0x08 0x400001f0 > 0x09 0x400001f1 > 0x0d 0x0221401f > 0x0e 0x01014010 > 0x0f 0x01813020 > 0x10 0x02a19021 > 0x11 0x90100140 > 0x12 0x400001f2 > > /sys/class/sound/hwC0D0/user_pin_configs: > > /sys/class/sound/hwC0D0/init_verbs: > > > !!ALSA/HDA dmesg > !!------------------ > > [ 38.190306] SELinux: initialized (dev configfs, type configfs), uses genfs_contexts > [ 38.229616] snd_hda_intel 0000:00:1b.0: irq 66 for MSI/MSI-X > [ 38.270699] ALSA sound/usb/mixer.c:866 6:0: cannot get min/max values for control 2 (id 6) > [ 38.274097] ALSA sound/usb/mixer.c:866 1:0: cannot get min/max values for control 2 (id 1) > [ 38.276753] ALSA sound/usb/mixer.c:866 2:0: cannot get min/max values for control 2 (id 2) > [ 38.279322] ALSA sound/pci/hda/hda_auto_parser.c:322 autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) type:line > [ 38.279326] ALSA sound/pci/hda/hda_auto_parser.c:326 speaker_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 38.279329] ALSA sound/pci/hda/hda_auto_parser.c:330 hp_outs=1 (0xd/0x0/0x0/0x0/0x0) > [ 38.279331] ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0 > [ 38.279333] ALSA sound/pci/hda/hda_auto_parser.c:335 inputs: > [ 38.279336] ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x10 > [ 38.279342] ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0xf > [ 38.285983] usbcore: registered new interface driver snd-usb-audio > [ 38.290348] input: HDA Intel Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 > [ 38.294975] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 > [ 38.297406] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 > [ 38.301369] input: HDA Intel Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 > [ 39.377352] EXT4-fs: Warning: mounting with data=journal disables delayed allocation and O_DIRECT support! > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

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[alsa-devel] [PATCH] ASoC: samsung dma - prepare dma in start/resume trigger path
by Dylan Reid 24 Aug '12

24 Aug '12

On suspend all dma transactions are terminated (by the stop/suspend trigger calling ops->stop), they need to be set up again during resume. Move the prepare op to the start/resume/unpause trigger path so that dma and audio will restart on resume/unpause. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..5772762 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -250,6 +250,7 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dma_prepare(substream); prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; @@ -352,7 +353,6 @@ static struct snd_pcm_ops dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = dma_hw_params, .hw_free = dma_hw_free, - .prepare = dma_prepare, .trigger = dma_trigger, .pointer = dma_pointer, .mmap = dma_mmap, -- 1.7.7.3

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Re: [alsa-devel] linux-next: Tree for Aug 23 (sound/isa/cmi)
by Randy Dunlap 24 Aug '12

24 Aug '12

On 08/22/2012 10:22 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120822: > on i386: CC sound/isa/cmi8328.o sound/isa/cmi8328.c: In function 'snd_cmi8328_remove': sound/isa/cmi8328.c:416:24: error: 'cmi' undeclared (first use in this function) sound/isa/cmi8328.c:416:24: note: each undeclared identifier is reported only once for each function it appears in make[3]: *** [sound/isa/cmi8328.o] Error 1 Full randconfig file is attached. -- ~Randy

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[alsa-devel] [PATCH v2 1/1] [ALSA] hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
by mengdong.lin＠intel.com 24 Aug '12

24 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> The patch to support runtime PM introduced a bug: Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk' 'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE. This patch is to fix the bug. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 6 ++++++ sound/pci/hda/hda_intel.c | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d6b3a6d..e618394 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1320,11 +1320,13 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_CLKSTOP); if (!codec->d3_stop_clk) bus->power_keep_link_on = 1; +#endif /* power-up all before initialization */ hda_set_power_state(codec, @@ -3523,7 +3525,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk_ok = 0; +#endif if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); @@ -3548,9 +3552,11 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, break; } +#ifdef CONFIG_SND_HDA_POWER_SAVE if ((power_state == AC_PWRST_D3) && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) codec->d3_stop_clk_ok = 1; +#endif } #ifdef CONFIG_SND_HDA_HWDEP diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2baa3ee..97307df 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2515,8 +2515,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; +#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); -- 1.7.9.5

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Re: [alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Ricardo Neri 24 Aug '12

24 Aug '12

On 08/21/2012 07:39 AM, Clark, Rob wrote: > On Tue, Aug 21, 2012 at 1:01 AM, Tomi Valkeinen <tomi.valkeinen(a)ti.com> wrote: >> On Mon, 2012-08-20 at 20:47 -0500, Ricardo Neri wrote: >>> Hello! >>> >>> I have been working on prototypes for the ASoC OMAP HDMI audio driver to >>> propagate events from the HDMI output (e.g., display getting >>> enabled/disabled/suspended). This for the users of the driver to react >>> to such events. For instance, if the display is disabled or disconected, >>> audio could be stopped, rerouted or whatever other decision the user >>> makes. This is needed because, if, for instance, the HDMI IP goes off, >>> audio will stall and the audio users will only see a "playback write >>> error (DMA or IRQ trouble?)" >>> >>> In my prototypes I have used snd_soc_jack for this purpose and I have >>> some questions: >>> >>> *I see snd_soc_jack is used mostly for headsets and microphones with >>> actual external mechanical connections. Strictly, in my case I propagate >>> events originated by the OMAP display driver (changes in the power >>> state), and not from external events. Some of these events are generated >>> from an actual HDMI cable connection/disconnection, though. >>> >>> *Maybe the event should be propagated by omapdss/omapdrm/drm and the >>> entity in charge of the audio policy should listen those events instead. >>> >>> *I do see SND_JACK_VIDEOOUT and SND_JACK_AVOUT types so maybe it is >>> feasible for an audio driver to report events from an AV output. >>> >>> I was wondering about how much sense does it make to you guys use a >>> snd_soc_jack in this case? >> >> How does DRM handle audio? I made a quick grep, but I see the drm >> drivers only enabling the audio in the HW, nothing else. > > I confess to not knowing too much about audio/alsa, but what does > audio driver need from hdmi? Just hotplug events? At least for the case of the ASoC HDMI audio driver (but hopefully for other audio drivers as well), it needs to detect whether an HDMI output is available, if the display's current configuration supports audio (e.g., a 1080p display configured as VGA should be reported as not supporting audio). It also needs interfaces to configure/prepare/start/stop audio. Also, of course, needs to know if the display is off/on/suspended and when transitions occur. For OMAP, omapdss provide an interface for this functionality for ALSA or any other interested user. > > From a quick look, it seems most of what the existing drm drivers are > doing is detecting if the display supports audio, and then turning > on/off the hw.. (and some infoframe stuff in some cases). Yes, it seems to me that every driver makes its own audio implementation, mainly focused on configuration. I could not find any audio common interface so that users like ALSA can take advantage of. Also, I could not see any ALSA driver using functionality provided by a drm driver. Maybe the lack of audio support in drm is because the audio users should not talk to drm directly but to a lower level component (omapdrm, omapdss?). However, today there exists video technology supports audio as well, such as DisplayPort or HDMI. Could it make more sense now to provide audio support? > > Does ASoC support 'hotplug' of audio devices? If so, maybe it makes > some sense to have some support in drm core. At least all the edid > parsing stuff to determine if the display supports audio should be > generic and not driver specific. drm already supports generic edid parsing to check for audio support. This is a good example of how an audio driver could just use that functionality to probe audio support. Ricardo > > BR, > -R > >> If there's a common generic way to handle this, we should obviously use >> that. But if we need to choose between doing something custom or doing >> it in omapdrm driver, I think we should go for drm the only solution and >> forget about audio with omapfb. >> >> Tomi >>

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[alsa-devel] [PATCH] [ALSA] hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
by mengdong.lin＠intel.com 24 Aug '12

24 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> The patch to support runtime PM introduced a bug: Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk' 'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE. This patch is to fix the bug. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2baa3ee..97307df 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2515,8 +2515,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; +#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Don't force card bias level to be updated
by Mark Brown 23 Aug '12

23 Aug '12

Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c1fb6a6..53f69b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,8 +293,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, level); else dapm->bias_level = level; - } else + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: dapm: Don't force card bias level to be updated
by Mark Brown 23 Aug '12

23 Aug '12

Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c1fb6a6..53f69b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,8 +293,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, level); else dapm->bias_level = level; - } else + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; -- 1.7.10.4

1 0

[alsa-devel] Undelivered Mail Returned to Sender
by MAILER-DAEMON＠opensource.wolfsonmicro.com 23 Aug '12

23 Aug '12

This is the mail system at host opensource.wolfsonmicro.com. I'm sorry to have to inform you that your message could not be delivered to one or more recipients. It's attached below. For further assistance, please send mail to postmaster. If you do so, please include this problem report. You can delete your own text from the attached returned message. The mail system <alsa-devel(a)alsa-projects.org>: Host or domain name not found. Name service error for name=alsa-projects.org type=AAAA: Host not found

1 0

[alsa-devel] [PATCH] ASoC: wm0010: Fix passthrough routing
by Mark Brown 23 Aug '12

23 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8e0b6d6..e1315e0 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -119,7 +119,8 @@ struct wm0010_spi_msg { }; static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { - { "SDI2 Playback", NULL, "SDI1 Playback" }, + { "SDI2 Capture", NULL, "SDI1 Playback" }, + { "SDI1 Capture", NULL, "SDI2 Playback" }, }; static const char *wm0010_state_to_str(enum wm0010_state state) -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm0010: Add initial wm0010 DSP driver
by Dimitris Papastamos 23 Aug '12

23 Aug '12

The WM0010 is a compact digital signal processor that has been highly optimised for low-power audio applications. Extensive memory resources and core optimisation allow the device to manage all audio processing algorithms efficiently and autonomously, while the host processor sleeps or performs other tasks. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/wm0010.h | 27 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm0010.c | 930 ++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 963 insertions(+) create mode 100644 include/sound/wm0010.h create mode 100644 sound/soc/codecs/wm0010.c diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h new file mode 100644 index 0000000..3261e90 --- /dev/null +++ b/include/sound/wm0010.h @@ -0,0 +1,27 @@ +/* + * wm0010.h -- Platform data for WM0010 DSP Driver + * + * Copyright 2012 Wolfson Microelectronics PLC. + * + * Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef WM0010_PDATA_H +#define WM0010_PDATA_H + +struct wm0010_pdata { + int gpio_reset; + + /* Set if there is an inverter between the GPIO controlling + * the reset signal and the device. + */ + int reset_active_high; + int irq_flags; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e7b795..5149a46 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -66,6 +66,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE + select SND_SOC_WM0010 if SPI_MASTER select SND_SOC_WM1250_EV1 if I2C select SND_SOC_WM2000 if I2C select SND_SOC_WM2200 if I2C @@ -319,6 +320,9 @@ config SND_SOC_UDA1380 config SND_SOC_WL1273 tristate +config SND_SOC_WM0010 + tristate + config SND_SOC_WM1250_EV1 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a0cf18c..e9dd804 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -57,6 +57,7 @@ snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o +snd-soc-wm0010-objs := wm0010.o snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm2200-objs := wm2200.o @@ -175,6 +176,7 @@ obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o +obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o Obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c new file mode 100644 index 0000000..8e0b6d6 --- /dev/null +++ b/sound/soc/codecs/wm0010.c @@ -0,0 +1,930 @@ +/* + * wm0010.c -- WM0010 DSP Driver + * + * Copyright 2012 Wolfson Microelectronics PLC. + * + * Authors: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> + * Scott Ling <sl(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/irqreturn.h> +#include <linux/init.h> +#include <linux/spi/spi.h> +#include <linux/firmware.h> +#include <linux/delay.h> +#include <linux/fs.h> +#include <linux/miscdevice.h> +#include <linux/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <linux/workqueue.h> + +#include <sound/soc.h> +#include <sound/wm0010.h> + +#define DEVICE_ID_WM0010 10 + +enum dfw_cmd { + DFW_CMD_FUSE = 0x01, + DFW_CMD_CODE_HDR, + DFW_CMD_CODE_DATA, + DFW_CMD_PLL, + DFW_CMD_INFO = 0xff +}; + +struct dfw_binrec { + u8 command; + u32 length:24; + u32 address; + uint8_t data[0]; +} __packed; + +struct dfw_pllrec { + u8 command; + u32 length:24; + u32 address; + u32 clkctrl1; + u32 clkctrl2; + u32 clkctrl3; + u32 ldetctrl; + u32 uart_div; + u32 spi_div; +} __packed; + +static struct pll_clock_map { + int max_sysclk; + int max_pll_spi_speed; + u32 pll_clkctrl1; +} pll_clock_map[] = { /* Dividers */ + { 22000000, 26000000, 0x00201f11 }, /* 2,32,2 */ + { 18000000, 26000000, 0x00203f21 }, /* 2,64,4 */ + { 14000000, 26000000, 0x00202620 }, /* 1,39,4 */ + { 10000000, 22000000, 0x00203120 }, /* 1,50,4 */ + { 6500000, 22000000, 0x00204520 }, /* 1,70,4 */ + { 5500000, 22000000, 0x00103f10 }, /* 1,64,2 */ +}; + +enum wm0010_state { + WM0010_POWER_OFF, + WM0010_OUT_OF_RESET, + WM0010_BOOTROM, + WM0010_STAGE2, + WM0010_FIRMWARE, +}; + +struct wm0010_priv { + struct snd_soc_codec *codec; + + struct mutex lock; + struct device *dev; + + struct wm0010_pdata pdata; + + int gpio_reset; + int gpio_reset_value; + + struct regulator_bulk_data core_supplies[2]; + struct regulator *dbvdd; + + int sysclk; + + enum wm0010_state state; + bool boot_failed; + int boot_done; + bool ready; + bool pll_running; + int max_spi_freq; + int board_max_spi_speed; + u32 pll_clkctrl1; + + spinlock_t irq_lock; + int irq; + + struct completion boot_completion; +}; + +struct wm0010_spi_msg { + struct spi_message m; + struct spi_transfer t; + u8 *tx_buf; + u8 *rx_buf; + size_t len; +}; + +static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { + { "SDI2 Playback", NULL, "SDI1 Playback" }, +}; + +static const char *wm0010_state_to_str(enum wm0010_state state) +{ + const char *state_to_str[] = { + "Power off", + "Out of reset", + "Bootrom", + "Stage2", + "Firmware" + }; + + if (state < 0 || state >= ARRAY_SIZE(state_to_str)) + return "null"; + return state_to_str[state]; +} + +/* Called with wm0010->lock held */ +static void wm0010_halt(struct snd_soc_codec *codec) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; + enum wm0010_state state; + + /* Fetch the wm0010 state */ + spin_lock_irqsave(&wm0010->irq_lock, flags); + state = wm0010->state; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + switch (state) { + case WM0010_POWER_OFF: + /* If there's nothing to do, bail out */ + return; + case WM0010_OUT_OF_RESET: + case WM0010_BOOTROM: + case WM0010_STAGE2: + case WM0010_FIRMWARE: + /* Remember to put chip back into reset */ + gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + /* Disable the regulators */ + regulator_disable(wm0010->dbvdd); + regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + break; + } + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_POWER_OFF; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); +} + +struct wm0010_boot_xfer { + struct list_head list; + struct snd_soc_codec *codec; + struct completion *done; + struct spi_message m; + struct spi_transfer t; +}; + +/* Called with wm0010->lock held */ +static void wm0010_mark_boot_failure(struct wm0010_priv *wm0010) +{ + enum wm0010_state state; + unsigned long flags; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + state = wm0010->state; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + dev_err(wm0010->dev, "Failed to transition from `%s' state to `%s' state\n", + wm0010_state_to_str(state), wm0010_state_to_str(state + 1)); + + wm0010->boot_failed = true; +} + +static void wm0010_boot_xfer_complete(void *data) +{ + struct wm0010_boot_xfer *xfer = data; + struct snd_soc_codec *codec = xfer->codec; + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + u32 *out32 = xfer->t.rx_buf; + int i; + + if (xfer->m.status != 0) { + dev_err(codec->dev, "SPI transfer failed: %d\n", + xfer->m.status); + wm0010_mark_boot_failure(wm0010); + if (xfer->done) + complete(xfer->done); + return; + } + + for (i = 0; i < xfer->t.len / 4; i++) { + dev_dbg(codec->dev, "%d: %04x\n", i, out32[i]); + + switch (be32_to_cpu(out32[i])) { + case 0xe0e0e0e0: + dev_err(codec->dev, + "%d: ROM error reported in stage 2\n", i); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x55555555: + if (wm0010->boot_done == 0) + break; + dev_err(codec->dev, + "%d: ROM bootloader running in stage 2\n", i); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0000: + dev_dbg(codec->dev, "Stage2 loader running\n"); + break; + + case 0x0fed0007: + dev_dbg(codec->dev, "CODE_HDR packet received\n"); + break; + + case 0x0fed0008: + dev_dbg(codec->dev, "CODE_DATA packet received\n"); + break; + + case 0x0fed0009: + dev_dbg(codec->dev, "Download complete\n"); + break; + + case 0x0fed000c: + dev_dbg(codec->dev, "Application start\n"); + break; + + case 0x0fed000e: + dev_dbg(codec->dev, "PLL packet received\n"); + wm0010->pll_running = true; + break; + + case 0x0fed0025: + dev_err(codec->dev, "Device reports image too long\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed002c: + dev_err(codec->dev, "Device reports bad SPI packet\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0031: + dev_err(codec->dev, "Device reports SPI read overflow\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0032: + dev_err(codec->dev, "Device reports SPI underclock\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0033: + dev_err(codec->dev, "Device reports bad header packet\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0034: + dev_err(codec->dev, "Device reports invalid packet type\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0035: + dev_err(codec->dev, "Device reports data before header error\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0038: + dev_err(codec->dev, "Device reports invalid PLL packet\n"); + break; + + case 0x0fed003a: + dev_err(codec->dev, "Device reports packet alignment error\n"); + wm0010_mark_boot_failure(wm0010); + break; + + default: + dev_err(codec->dev, "Unrecognised return 0x%x\n", + be32_to_cpu(out32[i])); + wm0010_mark_boot_failure(wm0010); + break; + } + + if (wm0010->boot_failed) + break; + } + + wm0010->boot_done++; + if (xfer->done) + complete(xfer->done); +} + +static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len) +{ + int i; + + for (i = 0; i < len / 8; i++) + data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i])); +} + +static int wm0010_boot(struct snd_soc_codec *codec) +{ + struct spi_device *spi = to_spi_device(codec->dev); + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; + struct list_head xfer_list; + struct wm0010_boot_xfer *xfer; + int ret; + struct completion done; + const struct firmware *fw; + const struct dfw_binrec *rec; + struct spi_message m; + struct spi_transfer t; + struct dfw_pllrec pll_rec; + u32 *img, *p; + u64 *img_swap; + u8 *out; + u32 len, offset; + int i; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + if (wm0010->state != WM0010_POWER_OFF) + dev_warn(wm0010->dev, "DSP already powered up!\n"); + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + if (wm0010->sysclk > 26000000) { + dev_err(codec->dev, "Max DSP clock frequency is 26MHz\n"); + ret = -ECANCELED; + goto err; + } + + INIT_LIST_HEAD(&xfer_list); + + mutex_lock(&wm0010->lock); + wm0010->pll_running = false; + + dev_dbg(codec->dev, "max_spi_freq: %d\n", wm0010->max_spi_freq); + + ret = regulator_bulk_enable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to enable core supplies: %d\n", + ret); + mutex_unlock(&wm0010->lock); + goto err; + } + + ret = regulator_enable(wm0010->dbvdd); + if (ret != 0) { + dev_err(&spi->dev, "Failed to enable DBVDD: %d\n", ret); + goto err_core; + } + + /* Release reset */ + gpio_set_value(wm0010->gpio_reset, !wm0010->gpio_reset_value); + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_OUT_OF_RESET; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + /* First the bootloader */ + ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev); + if (ret != 0) { + dev_err(codec->dev, "Failed to request stage2 loader: %d\n", + ret); + goto abort; + } + + if (!wait_for_completion_timeout(&wm0010->boot_completion, + msecs_to_jiffies(10))) + dev_err(codec->dev, "Failed to get interrupt from DSP\n"); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_BOOTROM; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + dev_dbg(codec->dev, "Downloading %d byte stage 2 loader\n", fw->size); + + /* Copy to local buffer first as vmalloc causes problems for dma */ + img = kzalloc(fw->size, GFP_KERNEL); + if (!img) { + dev_err(codec->dev, "Failed to allocate image buffer\n"); + goto abort; + } + + out = kzalloc(fw->size, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, "Failed to allocate output buffer\n"); + goto abort; + } + + memcpy(img, &fw->data[0], fw->size); + + spi_message_init(&m); + memset(&t, 0, sizeof(t)); + t.rx_buf = out; + t.tx_buf = img; + t.len = fw->size; + t.bits_per_word = 8; + t.speed_hz = wm0010->sysclk / 10; + spi_message_add_tail(&t, &m); + + dev_dbg(codec->dev, "Starting initial download at %dHz\n", + t.speed_hz); + + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "Initial download failed: %d\n", ret); + goto abort; + } + + /* Look for errors from the boot ROM */ + for (i = 0; i < fw->size; i++) { + if (out[i] != 0x55) { + ret = -EBUSY; + dev_err(codec->dev, "Boot ROM error: %x in %d\n", + out[i], i); + wm0010_mark_boot_failure(wm0010); + goto abort; + } + } + + release_firmware(fw); + kfree(img); + kfree(out); + + if (!wait_for_completion_timeout(&wm0010->boot_completion, + msecs_to_jiffies(10))) + dev_err(codec->dev, "Failed to get interrupt from DSP loader.\n"); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_STAGE2; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + /* Only initialise PLL if max_spi_freq initialised */ + if (wm0010->max_spi_freq) { + + /* Initialise a PLL record */ + memset(&pll_rec, 0, sizeof(pll_rec)); + pll_rec.command = DFW_CMD_PLL; + pll_rec.length = (sizeof(pll_rec) - 8); + + /* On wm0010 only the CLKCTRL1 value is used */ + pll_rec.clkctrl1 = wm0010->pll_clkctrl1; + + len = pll_rec.length + 8; + out = kzalloc(len, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, + "Failed to allocate RX buffer\n"); + goto abort; + } + + img_swap = kzalloc(len, GFP_KERNEL); + if (!img_swap) { + dev_err(codec->dev, + "Failed to allocate image buffer\n"); + goto abort; + } + + /* We need to re-order for 0010 */ + byte_swap_64((u64 *)&pll_rec, img_swap, len); + + spi_message_init(&m); + memset(&t, 0, sizeof(t)); + t.rx_buf = out; + t.tx_buf = img_swap; + t.len = len; + t.bits_per_word = 8; + t.speed_hz = wm0010->sysclk / 6; + spi_message_add_tail(&t, &m); + + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "First PLL write failed: %d\n", ret); + goto abort; + } + + /* Use a second send of the message to get the return status */ + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "Second PLL write failed: %d\n", ret); + goto abort; + } + + p = (u32 *)out; + + /* Look for PLL active code from the DSP */ + for (i = 0; i < len / 4; i++) { + if (*p == 0x0e00ed0f) { + dev_dbg(codec->dev, "PLL packet received\n"); + wm0010->pll_running = true; + break; + } + p++; + } + + kfree(img_swap); + kfree(out); + } else + dev_dbg(codec->dev, "Not enabling DSP PLL."); + + ret = request_firmware(&fw, "wm0010.dfw", codec->dev); + if (ret != 0) { + dev_err(codec->dev, "Failed to request application: %d\n", + ret); + goto abort; + } + + rec = (const struct dfw_binrec *)fw->data; + offset = 0; + wm0010->boot_done = 0; + wm0010->boot_failed = false; + BUG_ON(!list_empty(&xfer_list)); + init_completion(&done); + + /* First record should be INFO */ + if (rec->command != DFW_CMD_INFO) { + dev_err(codec->dev, "First record not INFO\r\n"); + goto abort; + } + + /* Check it's a 0010 file */ + if (rec->data[0] != DEVICE_ID_WM0010) { + dev_err(codec->dev, "Not a WM0010 firmware file.\r\n"); + goto abort; + } + + /* Skip the info record as we don't need to send it */ + offset += ((rec->length) + 8); + rec = (void *)&rec->data[rec->length]; + + while (offset < fw->size) { + dev_dbg(codec->dev, + "Packet: command %d, data length = 0x%x\r\n", + rec->command, rec->length); + len = rec->length + 8; + + out = kzalloc(len, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, + "Failed to allocate RX buffer\n"); + goto abort; + } + + img_swap = kzalloc(len, GFP_KERNEL); + if (!img_swap) { + dev_err(codec->dev, + "Failed to allocate image buffer\n"); + goto abort; + } + + /* We need to re-order for 0010 */ + byte_swap_64((u64 *)&rec->command, img_swap, len); + + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + + spi_message_init(&xfer->m); + xfer->m.complete = wm0010_boot_xfer_complete; + xfer->m.context = xfer; + xfer->t.tx_buf = img_swap; + xfer->t.rx_buf = out; + xfer->t.len = len; + xfer->t.bits_per_word = 8; + + if (!wm0010->pll_running) { + xfer->t.speed_hz = wm0010->sysclk / 6; + } else { + xfer->t.speed_hz = wm0010->max_spi_freq; + + if (wm0010->board_max_spi_speed && + (wm0010->board_max_spi_speed < wm0010->max_spi_freq)) + xfer->t.speed_hz = wm0010->board_max_spi_speed; + } + + /* Store max usable spi frequency for later use */ + wm0010->max_spi_freq = xfer->t.speed_hz; + + spi_message_add_tail(&xfer->t, &xfer->m); + + offset += ((rec->length) + 8); + rec = (void *)&rec->data[rec->length]; + + if (offset >= fw->size) { + dev_dbg(codec->dev, "All transfers scheduled\n"); + xfer->done = &done; + } + + ret = spi_async(spi, &xfer->m); + if (ret != 0) { + dev_err(codec->dev, "Write failed: %d\n", ret); + goto abort; + } + + if (wm0010->boot_failed) + goto abort; + } + + wait_for_completion(&done); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_FIRMWARE; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + mutex_unlock(&wm0010->lock); + + release_firmware(fw); + + while (!list_empty(&xfer_list)) { + xfer = list_first_entry(&xfer_list, struct wm0010_boot_xfer, + list); + kfree(xfer->t.rx_buf); + kfree(xfer->t.tx_buf); + list_del(&xfer->list); + kfree(xfer); + } + + return 0; + +abort: + /* Put the chip back into reset */ + wm0010_halt(codec); + mutex_unlock(&wm0010->lock); + return ret; +err_core: + regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); +err: + return ret; +} + +static int wm0010_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) + wm0010_boot(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { + mutex_lock(&wm0010->lock); + wm0010_halt(codec); + mutex_unlock(&wm0010->lock); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int wm0010_set_sysclk(struct snd_soc_codec *codec, int source, + int clk_id, unsigned int freq, int dir) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned int i; + + wm0010->sysclk = freq; + + if (freq < pll_clock_map[ARRAY_SIZE(pll_clock_map)-1].max_sysclk) { + wm0010->max_spi_freq = 0; + } else { + for (i = 0; i < ARRAY_SIZE(pll_clock_map); i++) + if (freq >= pll_clock_map[i].max_sysclk) + break; + + wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed; + wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1; + } + + return 0; +} + +static int wm0010_probe(struct snd_soc_codec *codec); + +static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { + .probe = wm0010_probe, + .set_bias_level = wm0010_set_bias_level, + .set_sysclk = wm0010_set_sysclk, + + .dapm_routes = wm0010_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), +}; + +#define WM0010_RATES (SNDRV_PCM_RATE_48000) +#define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm0010_dai[] = { + { + .name = "wm0010-sdi1", + .playback = { + .stream_name = "SDI1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + .capture = { + .stream_name = "SDI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + }, + { + .name = "wm0010-sdi2", + .playback = { + .stream_name = "SDI2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + .capture = { + .stream_name = "SDI2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + }, +}; + +static irqreturn_t wm0010_irq(int irq, void *data) +{ + struct wm0010_priv *wm0010 = data; + + switch (wm0010->state) { + case WM0010_POWER_OFF: + case WM0010_OUT_OF_RESET: + case WM0010_BOOTROM: + case WM0010_STAGE2: + spin_lock(&wm0010->irq_lock); + complete(&wm0010->boot_completion); + spin_unlock(&wm0010->irq_lock); + return IRQ_HANDLED; + default: + return IRQ_NONE; + } + + return IRQ_NONE; +} + +static int wm0010_probe(struct snd_soc_codec *codec) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + struct spi_device *spi = to_spi_device(wm0010->dev); + unsigned long flags; + unsigned long gpio_flags; + int ret; + int trigger; + int irq; + + wm0010->codec = codec; + + init_completion(&wm0010->boot_completion); + + wm0010->core_supplies[0].supply = "AVDD"; + wm0010->core_supplies[1].supply = "DCVDD"; + ret = devm_regulator_bulk_get(wm0010->dev, ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + if (ret != 0) { + dev_err(wm0010->dev, "Failed to obtain core supplies: %d\n", + ret); + return ret; + } + + wm0010->dbvdd = devm_regulator_get(wm0010->dev, "DBVDD"); + if (IS_ERR(wm0010->dbvdd)) { + ret = PTR_ERR(wm0010->dbvdd); + dev_err(wm0010->dev, "Failed to obtain DBVDD: %d\n", ret); + return ret; + } + + if (wm0010->pdata.gpio_reset) { + wm0010->gpio_reset = wm0010->pdata.gpio_reset; + + if (wm0010->pdata.reset_active_high) + wm0010->gpio_reset_value = 1; + else + wm0010->gpio_reset_value = 0; + + if (wm0010->gpio_reset_value) + gpio_flags = GPIOF_OUT_INIT_HIGH; + else + gpio_flags = GPIOF_OUT_INIT_LOW; + + ret = devm_gpio_request_one(wm0010->dev, wm0010->gpio_reset, + gpio_flags, "wm0010 reset"); + if (ret < 0) { + dev_err(wm0010->dev, + "Failed to request GPIO for DSP reset: %d\n", + ret); + return ret; + } + } else { + dev_err(wm0010->dev, "No reset GPIO configured\n"); + return ret; + } + + irq = spi->irq; + if (wm0010->pdata.irq_flags) + trigger = wm0010->pdata.irq_flags; + else + trigger = IRQF_TRIGGER_FALLING; + trigger |= IRQF_ONESHOT; + + ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, + "wm0010", wm0010); + if (ret) + dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", + irq, ret); + wm0010->irq = irq; + + if (spi->max_speed_hz) + wm0010->board_max_spi_speed = spi->max_speed_hz; + else + wm0010->board_max_spi_speed = 0; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_POWER_OFF; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + return 0; +} + +static int __devinit wm0010_spi_probe(struct spi_device *spi) +{ + struct wm0010_priv *wm0010; + int ret; + + wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), + GFP_KERNEL); + if (!wm0010) + return -ENOMEM; + + mutex_init(&wm0010->lock); + spin_lock_init(&wm0010->irq_lock); + + spi_set_drvdata(spi, wm0010); + wm0010->dev = &spi->dev; + + if (dev_get_platdata(&spi->dev)) + memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), + sizeof(wm0010->pdata)); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_wm0010, wm0010_dai, + ARRAY_SIZE(wm0010_dai)); + if (ret < 0) + return ret; + + return 0; +} + +static int __devexit wm0010_spi_remove(struct spi_device *spi) +{ + struct wm0010_priv *wm0010 = spi_get_drvdata(spi); + + snd_soc_unregister_codec(&spi->dev); + + if (wm0010->gpio_reset) { + /* Remember to put chip back into reset */ + gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + gpio_free(wm0010->gpio_reset); + } + + if (wm0010->irq) + free_irq(wm0010->irq, wm0010); + + return 0; +} + +static struct spi_driver wm0010_spi_driver = { + .driver = { + .name = "wm0010", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm0010_spi_probe, + .remove = __devexit_p(wm0010_spi_remove), +}; + +module_spi_driver(wm0010_spi_driver); + +MODULE_DESCRIPTION("ASoC WM0010 driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); -- 1.7.12

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Update for new WM1811 variants
by Mark Brown 23 Aug '12

23 Aug '12

There are some new WM1811 variants distinguished by both revision and cust_id which need slightly different handling. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8d5b65a..c7dbbd4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3823,14 +3823,20 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.no_cache_dac_hp_direct = true; wm8994->fll_byp = true; - switch (wm8994->revision) { + switch (control->cust_id) { case 0: - case 1: case 2: - case 3: wm8994->hubs.dcs_codes_l = -9; wm8994->hubs.dcs_codes_r = -7; break; + case 1: + case 3: + wm8994->hubs.dcs_codes_l = -8; + wm8994->hubs.dcs_codes_r = -7; + break; + default: + break; + } default: break; } @@ -3919,7 +3925,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM1811: - if (wm8994->revision > 1) { + if (control->cust_id > 1 || wm8994->revision > 1) { ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm1811_jackdet_irq, "JACKDET", -- 1.7.10.4

1 0

[alsa-devel] SAIF configuration on imx28
by George Stefan 23 Aug '12

23 Aug '12

Hello, I am using a custom board based on imx28 (running 2.6.35) with the SAIF interface connected to the PCM output of a modem. I have developed a dummy codec for the modem and customized mxs-devb.c to use that codec instead of sgtl5000. Clock master, fixed at 256kHz, Frame sync master, fixed 125us frame duration (8kHz): 32 bits of total data; first 16 bits are valid (while FS is high), following 16 are "don't care" (while FS low) 16-bit linear samples, sent MSB first "Long Frame" sync: transmit and receive occur simultaneously while the common FS line is active (high). TX data with 0-bit delay; start at the rising edge of the clock, while FS is high. RX data with 0-bit delay; sample at the falling edge of the clock, while FS is high. This configuration is fixed so the only setup needed for SAIF is dai_format = SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); if (ret < 0) return ret; My problem is, although i am receiving dma irqs when i record ( arecord -f S16_LE -r 8000 -c 1 <file>.wav), when i play the resulting file its just silence. [ 5295.890000] dump @ mxs_saif_trigger [ 5295.890000] scr 101 , 107 [ 5295.890000] stat 80010041 , 80010071 [ 5295.890000] data 0 , ffffffff [ 5295.890000] version 1010000 , 1010000 This is a dump for arecord! Do you have any ideeas? Thanks, George.

3 14

[alsa-devel] [PATCH] ASoC: dapm: Fix return value in snd_soc_dapm_new_control()
by Dimitris Papastamos 23 Aug '12

23 Aug '12

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 99aac65..90b8fb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2884,7 +2884,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = PTR_ERR(w->priv); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); - return ret; + return NULL; } break; default: -- 1.7.12

2 1

[alsa-devel] [PATCH] ARM: OMAP2+: twl-common: Fix compile time error when omap-twl4030 audio is not enabled
by Peter Ujfalusi 23 Aug '12

23 Aug '12

Fixes: CC arch/arm/mach-omap2/twl-common.o arch/arm/mach-omap2/twl-common.c:562: error: conflicting types for ‘omap_twl4030_audio_init’ arch/arm/mach-omap2/twl-common.h:62: error: previous declaration of ‘omap_twl4030_audio_init’ was here make[1]: *** [arch/arm/mach-omap2/twl-common.o] Error 1 make: *** [arch/arm/mach-omap2] Error 2 When building the kernel with !SND_OMAP_SOC_OMAP_TWL4030 Reported-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- arch/arm/mach-omap2/twl-common.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap2/twl-common.c b/arch/arm/mach-omap2/twl-common.c index 1c6045a..2919d7a 100644 --- a/arch/arm/mach-omap2/twl-common.c +++ b/arch/arm/mach-omap2/twl-common.c @@ -544,7 +544,7 @@ void __init omap_twl4030_audio_init(char *card_name) } #else /* SOC_OMAP_TWL4030 */ -void __init omap_twl4030_audio_init(char *card_name, int codec_sysclk) +void __init omap_twl4030_audio_init(char *card_name) { return; } -- 1.7.8.6

3 4

[alsa-devel] [PATCH v5 1/1] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 23 Aug '12

23 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gerneral sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled: -- If a codec supports 'stop-clock' in D3, it will request suspending the controller after it enters D3 and request resuming the controller before back to D0. Thus the controller will be suspended only when all codecs are suspended and support stop-clock in D3. -- User IO operations and HW wakeup signal can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. More about 'stop-clock' feature: If a codec can support targeted pass-through operations in D3 state when there is no BCLK present on the link, it will set CLKSTOP flag in the supported power states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec section 7.3.3.10 Power state and 7.3.4.12 Supported Power State. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 19 ++++++++++-- sound/pci/hda/hda_codec.h | 5 ++- sound/pci/hda/hda_intel.c | 75 +++++++++++++++++++++++++++++++++++---------- 3 files changed, 80 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..d6b3a6d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1320,12 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_CLKSTOP); + if (!codec->d3_stop_clk) + bus->power_keep_link_on = 1; + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -3514,6 +3523,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; + codec->d3_stop_clk_ok = 0; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3536,6 +3547,10 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } + + if ((power_state == AC_PWRST_D3) + && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) + codec->d3_stop_clk_ok = 1; } #ifdef CONFIG_SND_HDA_HWDEP @@ -4385,7 +4400,7 @@ static void hda_power_work(struct work_struct *work) hda_call_codec_suspend(codec); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4440,7 +4455,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_unlock(&codec->power_lock); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fe6206b..b384c721 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -612,7 +612,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus); + void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); #endif }; @@ -870,6 +870,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..2baa3ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1022,7 +1023,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus); +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); #endif /* reset codec link */ @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2390,23 +2394,17 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus) +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - struct hda_codec *c; - int power_on = 0; - list_for_each_entry(c, &bus->codec_list, list) { - if (c->power_on) { - power_on = 1; - break; - } - } - if (power_on) - azx_init_chip(chip, 1); - else if (chip->running && power_save_controller && - !bus->power_keep_link_on) - azx_stop_chip(chip); + if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + return; + + if (codec->power_on) + pm_runtime_get_sync(&chip->pci->dev); + else + pm_runtime_put_sync(&chip->pci->dev); } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2509,35 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + if (!power_save_controller) + return -EAGAIN; + + azx_stop_chip(chip); + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + azx_init_chip(chip, 1); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3201,6 +3227,15 @@ static void power_down_all_codecs(struct azx *chip) #endif } +static void rpm_get_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &chip->bus->codec_list, list) { + pm_runtime_get_noresume(&chip->pci->dev); + } +} + static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -3238,6 +3273,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3327,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + rpm_get_all_codecs(chip); /* all codecs are active */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3342,10 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

3 3

[alsa-devel] [Cross compile issue] - I met problem when to cross compile alsa-lib and alsa-utils.Could you please help on it?
by Tim Jiang 23 Aug '12

23 Aug '12

Hi experts, #My question: I met problem when to cross compile alsa-lib and alsa-utils.Could you please help on it? Thanks in advance. #details My host PC: ubuntu 11.10 64-bit; Alsa-lib source code version: 1.0.23; Alsa-utils source code version: 1.0.23; My target platform: PandaBoard; My build shell script: #!/bin/bash echo "=======================start build alsa-lib-1.0.23==============" cd ./alsa-lib-1.0.23/ export CC=arm-linux-gnueabi-gcc echo "-------------->`pwd`" ./configure --target=arm-linux --host=i686-linux --prefix=/home/tim/work/dev_board/tools/ALSA/output/alsa_lib --disable-python make clean make make install cd - echo "========================end build alsa-lib-1.0.23===============" echo "=======================start build alsa-utils-1.0.23====================" cd ./alsa-utils-1.0.23/ echo "-------------->`pwd`" ./configure --target=arm-linux --host=i686-linux --with-alsa-inc-prefix=/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/include --with-alsa-prefix=/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/lib --disable-alsamixer --disable-xmlto make I can build Alsa-lib successfully but error happened when to build Alsa-utils. =======================start build alsa-utils-1.0.23==================== -------------->/home/tim/work/dev_board/tools/ALSA/alsa-utils-1.0.23 configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... no checking for mawk... mawk checking whether make sets $(MAKE)... yes checking for i686-linux-strip... no checking for strip... strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for i686-linux-gcc... arm-linux-gnueabi-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gnueabi-gcc accepts -g... yes checking for arm-linux-gnueabi-gcc option to accept ISO C89... none needed checking dependency style of arm-linux-gnueabi-gcc... gcc3 checking build system type... x86_64-unknown-linux-gnu checking host system type... i686-pc-linux-gnu checking for ld used by GCC... /usr/arm-linux-gnueabi/bin/ld checking if the linker (/usr/arm-linux-gnueabi/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking for CFPreferencesCopyAppValue... no checking for CFLocaleCopyCurrent... no checking for GNU gettext in libc... yes checking whether to use NLS... yes checking where the gettext function comes from... libc checking for i686-linux-gcc... (cached) arm-linux-gnueabi-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gnueabi-gcc accepts -g... (cached) yes checking for arm-linux-gnueabi-gcc option to accept ISO C89... (cached) none needed checking dependency style of arm-linux-gnueabi-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... -I/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/include checking for ALSA LDFLAGS... -L/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/lib -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.16... found. checking for snd_ctl_open in -lasound... no configure: error: No linkable libasound was found. make: *** No targets specified and no makefile found. Stop. Best Regards, Tim

1 0

[alsa-devel] [PATCH v4 1/1] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 23 Aug '12

23 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gerneral sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled: -- If a codec supports 'stop-clock' in D3, it will request suspending the controller after it enters D3 and request resuming the controller before back to D0. Thus the controller will be suspended only when all codecs are suspended and support stop-clock in D3. -- User IO operations and HW wakeup signal can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. More about 'stop-clock' feature: If a codec can support targeted pass-through operations in D3 state when there is no BCLK present on the link, it will set CLKSTOP flag in the supported power states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec section 7.3.3.10 Power state and 7.3.4.12 Supported Power State. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 20 ++++++++++-- sound/pci/hda/hda_codec.h | 5 ++- sound/pci/hda/hda_intel.c | 75 +++++++++++++++++++++++++++++++++++---------- 3 files changed, 81 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..72e6328 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -26,6 +26,7 @@ #include <linux/pci.h> #include <linux/mutex.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> @@ -1209,6 +1210,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1321,12 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_CLKSTOP); + if (!codec->d3_stop_clk) + bus->power_keep_link_on = 1; + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -3514,6 +3524,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; + codec->d3_stop_clk_ok = 0; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3536,6 +3548,10 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } + + if ((power_state == AC_PWRST_D3) + && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) + codec->d3_stop_clk_ok = 1; } #ifdef CONFIG_SND_HDA_HWDEP @@ -4385,7 +4401,7 @@ static void hda_power_work(struct work_struct *work) hda_call_codec_suspend(codec); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4440,7 +4456,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_unlock(&codec->power_lock); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fe6206b..b384c721 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -612,7 +612,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus); + void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); #endif }; @@ -870,6 +870,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..2baa3ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1022,7 +1023,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus); +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); #endif /* reset codec link */ @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2390,23 +2394,17 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus) +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - struct hda_codec *c; - int power_on = 0; - list_for_each_entry(c, &bus->codec_list, list) { - if (c->power_on) { - power_on = 1; - break; - } - } - if (power_on) - azx_init_chip(chip, 1); - else if (chip->running && power_save_controller && - !bus->power_keep_link_on) - azx_stop_chip(chip); + if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + return; + + if (codec->power_on) + pm_runtime_get_sync(&chip->pci->dev); + else + pm_runtime_put_sync(&chip->pci->dev); } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2509,35 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + if (!power_save_controller) + return -EAGAIN; + + azx_stop_chip(chip); + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + azx_init_chip(chip, 1); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3201,6 +3227,15 @@ static void power_down_all_codecs(struct azx *chip) #endif } +static void rpm_get_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &chip->bus->codec_list, list) { + pm_runtime_get_noresume(&chip->pci->dev); + } +} + static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -3238,6 +3273,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3327,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + rpm_get_all_codecs(chip); /* all codecs are active */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3342,10 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v3 0/9] ARM/ASoC: OMAP McBSP device tree support
by Peter Ujfalusi 23 Aug '12

23 Aug '12

Hello, Changes since v2: - Replaced the IS_ERR_OR_NULL() with IS_ERR() Changes since v1: - ICLK workaround for the sidetone module has been kept for legacy mode. Intro mail from v2: in order to be able to add DT support for the McBSP driver which is used on all OMAP platforms (OMAP1/2/3/4/5) I needed to make some cleanups to the stack: - Removing callbacks used by the driver for - clock re-parenting (code moved to sound driver) - FSR/CLKR mux configuration (from now on it need to be done at board level) This mux is board specific and depends on the wiring. - Replaced the cpu_is_omap* checks with future proof check. - The FIFO was not enabled on OMAP2430 SoC It has been tested on BeagleBoard (with legacy and DT boot). The series changes McBSP related files mostly. It would be great if the whole series could go via audio tree (if the patches are OK and it is fine by Tony). For DT booted kernel on OMAP3 proper ICLK woraround need to be implemented. At the moment I don't see this as a stopper since we can not properly boot OMAP3 devices (especially rx51 which is using the ST). Other OMAP revisions does not have ST block so the missing workaround is not going to hurt them. I'm working on a solution which can replace the current workaround with a generic implementation to support DT and non DT booted kernels. Regards, Peter --- Peter Ujfalusi (9): ARM/ASoC: omap-mcbsp: Move OMAP2+ clock parenting code to ASoC driver ARM: OMAP: mcbsp: Enable FIFO use for OMAP2430 ARM: OMAP: board-am3517evm: Configure McBSP1 CLKR/FSR signal source ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code ASoC: omap-mcbsp: Remove unused defines ASoC: omap-mcbsp: Remove cpu_is_omap* checks from the code ARM: OMAP2+: McBSP: Do not create legacy devices when booting with DT data ASoC: omap-mcbsp: Add device tree bindings .../devicetree/bindings/sound/omap-mcbsp.txt | 45 +++++++ arch/arm/mach-omap2/board-am3517evm.c | 13 ++ arch/arm/mach-omap2/mcbsp.c | 126 +------------------ arch/arm/plat-omap/include/plat/mcbsp.h | 2 - sound/soc/omap/am3517evm.c | 20 +--- sound/soc/omap/mcbsp.c | 54 ++++----- sound/soc/omap/mcbsp.h | 3 - sound/soc/omap/omap-mcbsp.c | 106 +++++++++++----- sound/soc/omap/omap-mcbsp.h | 20 --- 9 files changed, 162 insertions(+), 227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt -- 1.7.8.6

5 33

[alsa-devel] [PATCH 0/5] ARM/ASoC: Davinci: Add Device Tree Support
by Hebbar, Gururaja 23 Aug '12

23 Aug '12

This patch-set is a cumulative of below tasks 1. Split asp header into platform and audio specific Davinci driver is shared by few OMAP platforms (like TI81xx, AM335x). Splitting asp header into Davinci platform specific and Audio specific helps to share them across platforms. 2. Add DT support for Davinci Audio Module (McASP, machine, pcm & Codec) This patch was tested on AM18x Board (Tested to check if the commits doesn't break on compile or runtime) and is based on repo at http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git; a=shortlog;h=refs/heads/for-3.7 Note: Matt Porter from TI is working on DMA engine conversion of DaVinci EDMA support and DT conversion of DMA parts will be taken up post that is complete. Since Audio requires DMA to work, this patch set does not yet provide complete functionality required to use sound on DT-based platforms, but is a step towards that direction. Hebbar, Gururaja (5): ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific ASoC: Davinci: McASP: add device tree support for McASP ASoC: Davinci: pcm: Add device tree bindings ASoC: Davinci: machine: Add device tree binding ASoC: tlv320aic3x: Add device tree bindings .../bindings/sound/davinci-evm-audio.txt | 52 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 43 +++++ .../devicetree/bindings/sound/davinci-pcm.txt | 13 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 18 ++ arch/arm/mach-davinci/asp.h | 49 +++++ arch/arm/mach-davinci/davinci.h | 3 +- arch/arm/mach-davinci/devices-da8xx.c | 1 + arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- .../linux/platform_data/davinci_asp.h | 73 ++------ sound/soc/codecs/tlv320aic3x.c | 30 +++ sound/soc/davinci/davinci-evm.c | 190 +++++++++++++++++--- sound/soc/davinci/davinci-i2s.c | 3 +- sound/soc/davinci/davinci-mcasp.c | 113 ++++++++++++- sound/soc/davinci/davinci-mcasp.h | 3 +- sound/soc/davinci/davinci-pcm.c | 12 ++- sound/soc/davinci/davinci-pcm.h | 3 +- 20 files changed, 527 insertions(+), 89 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt create mode 100644 arch/arm/mach-davinci/asp.h rename arch/arm/mach-davinci/include/mach/asp.h => include/linux/platform_data/davinci_asp.h (56%)

2 17

[alsa-devel] [PATCH v3 2/2] [ALSA] hda - suspend the controller only if all codecs support stop-clk in D3
by mengdong.lin＠intel.com 23 Aug '12

23 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> If a codec can support targeted pass-through operations in D3 state when there is no BCLK present on the link, it will set CLKSTOP flag in the supported power states and report PS-ClkStopOk when entering D3 state. So only if a codec supports this stop-clock feature, it will request suspending the controller after it enters D3 and requst resuming the controller before back to D0. Thus the controller will be suspended only when all codecs are suspended and support stop-clock in D3. Please refer to HDA spec section 7.3.3.10 Power state and 7.3.4.12 Supported Power States. Details: - azx_power_notify() add 'codec' as 2nd parameter to check its stop-clock flag. - each codec count 1 on the controller's power usage counter, and inc/dec the count independently from each other. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 23 +++++++++++++++++++++-- sound/pci/hda/hda_codec.h | 5 ++++- sound/pci/hda/hda_intel.c | 25 +++++++++++-------------- 3 files changed, 36 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..5d09662 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -26,6 +26,7 @@ #include <linux/pci.h> #include <linux/mutex.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> @@ -1206,9 +1207,14 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec->chip_name); kfree(codec->modelname); kfree(codec->wcaps); + /* undo get in snd_hda_codec_new() */ + pm_runtime_put_noidle(&codec->bus->pci->dev); kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,10 +1323,17 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_CLKSTOP); + if (!codec->d3_stop_clk) + bus->power_keep_link_on = 1; + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + pm_runtime_get_noresume(&bus->pci->dev); /* active by default */ snd_hda_codec_proc_new(codec); @@ -3514,6 +3527,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; + codec->d3_stop_clk_ok = 0; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3536,6 +3551,10 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } + + if ((power_state == AC_PWRST_D3) + && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) + codec->d3_stop_clk_ok = 1; } #ifdef CONFIG_SND_HDA_HWDEP @@ -4385,7 +4404,7 @@ static void hda_power_work(struct work_struct *work) hda_call_codec_suspend(codec); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4440,7 +4459,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_unlock(&codec->power_lock); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fe6206b..b384c721 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -612,7 +612,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus); + void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); #endif }; @@ -870,6 +870,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4c30fd7..16f037c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1023,7 +1023,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus); +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); #endif /* reset codec link */ @@ -2394,7 +2394,7 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus) +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; struct hda_codec *c; @@ -2406,19 +2406,20 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) { - if (!chip->initialized) - pm_runtime_get_sync(&chip->pci->dev); + + if (!bus->power_keep_link_on + && codec->power_on && codec->d3_stop_clk_ok) + pm_runtime_get_sync(&chip->pci->dev); + + if (power_on) azx_init_chip(chip, 1); - } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) { + !bus->power_keep_link_on) azx_stop_chip(chip); - /* TODO: Suspend controller only if all codec support - stop-clock in D3, for wakeup consideration */ + if (!bus->power_keep_link_on + && !codec->power_on && codec->d3_stop_clk_ok) pm_runtime_put_sync(&chip->pci->dev); - } } static DEFINE_MUTEX(card_list_lock); @@ -3327,7 +3328,6 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; - pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3343,9 +3343,6 @@ static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); - /* undo 'get' in azx_probe_continue() */ - pm_runtime_put_noidle(&pci->dev); - if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Device tree binding documentation update
by Peter Ujfalusi 22 Aug '12

22 Aug '12

To reflect the final devicetree node structure of McBSPs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, the initial OMAP McBSP DT structure was not able to describe the IP (and it's versions) correctly. The main issue was the sidetone block of McBSP2/3 on OMAP3. With this change in the DT description the OS can get the needed information about the IP. The sidetone is still not supported when the Linux kernel is booted with DT since we still depend on hwmod to fill the resources. This patch depend on the McBSP DT support series going via audio tree. Regards, Peter .../devicetree/bindings/sound/omap-mcbsp.txt | 28 +++++++------------ 1 files changed, 10 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt index 447cb13..17cce44 100644 --- a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -8,38 +8,30 @@ Required properties: - reg: Register location and size, for OMAP4+ as an array: <MPU access base address, size>, <L3 interconnect address, size>; +- reg-names: Array of strings associated with the address space - interrupts: Interrupt numbers for the McBSP port, as an array in case the McBSP IP have more interrupt lines: <OCP compliant irq>, <TX irq>, <RX irq>; +- interrupt-names: Array of strings associated with the interrupt numbers - interrupt-parent: The parent interrupt controller - ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) - ti,hwmods: Name of the hwmod associated to the McBSP port -Sidetone support for OMAP3 McBSP2 and 3 ports: -- sidetone { }: Within this section the following parameters are required: -- reg: Register location and size for the ST block -- interrupts: The interrupt number for the ST block -- interrupt-parent: The parent interrupt controller for the ST block - Example: mcbsp2: mcbsp@49022000 { compatible = "ti,omap3-mcbsp"; - #address-cells = <1>; - #size-cells = <1>; - reg = <0x49022000 0xff>; - interrupts = <0 17 0x4>, /* OCP compliant interrup */ - <0 62 0x4>, /* TX interrup */ - <0 63 0x4>; /* RX interrup */ + reg = <0x49022000 0xff>, + <0x49028000 0xff>; + reg-names = "mpu", "sidetone"; + interrupts = <0 17 0x4>, /* OCP compliant interrupt */ + <0 62 0x4>, /* TX interrupt */ + <0 63 0x4>, /* RX interrupt */ + <0 4 0x4>; /* Sidetone */ + interrupt-names = "common", "tx", "rx", "sidetone"; interrupt-parent = <&intc>; ti,buffer-size = <1280>; ti,hwmods = "mcbsp2"; - - sidetone { - reg = <0x49028000 0xff>; - interrupts = <0 4 0x4>; - interrupt-parent = <&intc>; - }; }; -- 1.7.8.6

2 1

[alsa-devel] [PATCH] ALSA: hda - fix broken HDMI jack detection after S3
by David Henningsson 22 Aug '12

22 Aug '12

The HDMI codec (an NVIDIA one in this case) forgot that its pins were unsol enabled, while it was suspended. Therefore jack detection was broken after S3. With this patch, we reenable the unsol events on resume, and also do an extra check afterwards, to see if the HDMI monitor was plugged/unplugged while in S3. Cc: stable(a)kernel.org (3.3+) BugLink: https://bugs.launchpad.net/bugs/1040030 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8f23374..6a3ac05 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1315,6 +1315,16 @@ static int generic_hdmi_init(struct hda_codec *codec) return 0; } +#ifdef CONFIG_PM +static int generic_hdmi_resume(struct hda_codec *codec) +{ + snd_hda_codec_resume_cache(codec); + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_report_sync(codec); + return 0; +} +#endif + static void generic_hdmi_free(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -1338,6 +1348,9 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, +#ifdef CONFIG_PM + .resume = generic_hdmi_resume, +#endif }; static int patch_generic_hdmi(struct hda_codec *codec) -- 1.7.9.5

2 8

[alsa-devel] [PATCH] ASoC: wm9712: Provide TLV information for capture boost controls
by Mark Brown 22 Aug '12

22 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f6baf9c..1992a62 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -132,8 +132,9 @@ SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1), SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), -SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), -SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1), +SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv), +SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1, + boost_tlv), SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), -- 1.7.10.4

1 0

Re: [alsa-devel] ARM: OMAP: twl-common: Add helper function to register the omap-twl4030 audio driver
by Brian Austin 22 Aug '12

22 Aug '12

Hey guys, This commit 552d9dc316cfedd134ee882cd1ae24ca6934ffc0 breaks the build for OMAP3 Beagelboard. with the following error. CC arch/arm/mach-omap2/twl-common.o arch/arm/mach-omap2/twl-common.c:562: error: conflicting types for ‘omap_twl4030_audio_init’ arch/arm/mach-omap2/twl-common.h:62: error: previous declaration of ‘omap_twl4030_audio_init’ was here make[1]: *** [arch/arm/mach-omap2/twl-common.o] Error 1 make: *** [arch/arm/mach-omap2] Error 2 You added a redifinition of omap_twl4030_audio_init in twl-common.c with the extra parameter of sysclk, but the header file has the definition with just the card_name. I'm not sure which way ya'll wanted to go on this so I didnt submit a patch. Just letting it be known. This is off of for-next. Regards, Brian

1 0

[alsa-devel] [PATCH] ALSA: hda - Do not set GPIOs for speakers on IDT if there are no speakers
by David Henningsson 22 Aug '12

22 Aug '12

This fixes an issue with a machine where there were no speakers, but GPIO0 had to be data=1 for the headphone to be functioning. I'm not sure if we need a more advanced patch to solve all possible cases, but if so, this patch would still provide a minor optimisation. BugLink: https://bugs.launchpad.net/bugs/1040077 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) The alsa-info for this machine can not be released yet, sorry. diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea5775a..3edd73c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4543,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec, struct auto_pin_cfg *cfg = &spec->autocfg; int i; + if (cfg->speaker_outs == 0) + return; + for (i = 0; i < cfg->line_outs; i++) { if (presence) break; -- 1.7.9.5

2 1

[alsa-devel] [PATCHv2.1-CA0132 HDA Codec 2/2] Add DSP loader code to CA0132 codec
by Ian Minett 22 Aug '12

22 Aug '12

From: Ian Minett <ian_minett(a)creativelabs.com> Draft patch to add new DSP loader code, with calls to load_dsp_xxxx() functions. Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 825f5b4..45dce9c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1952,22 +1952,13 @@ struct dma_engine { unsigned short m_converter_format; void *m_buffer_addr; unsigned int m_buffer_size; - unsigned int m_req_size; - struct snd_pcm_substream *substream; }; enum dma_state { - DMA_STATE_RESET = 0, - DMA_STATE_STOP = 1, - DMA_STATE_RUN = 2 + DMA_STATE_STOP = 0, + DMA_STATE_RUN = 1 }; -#define azx_pcm_open(a) (a->ops->open(a)) -#define azx_pcm_close(a) (a->ops->close(a)) -#define azx_pcm_prepare(a) (a->ops->prepare(a)) -#define azx_pcm_trigger(a, b) (a->ops->trigger(a, b)) -#define azx_pcm_hw_free(a) (a->ops->hw_free(a)) - static int dma_convert_to_hda_format( struct hda_stream_format *stream_format, unsigned short *hda_format) @@ -1986,113 +1977,6 @@ static int dma_convert_to_hda_format( return 0; } -static int dma_init( - struct hda_codec *codec, - struct dma_engine **pp_dma, - struct hda_stream_format *stream_format, - unsigned short *format, - unsigned int req_size) -{ - struct dma_engine *dma; - struct snd_pcm_substream *substream; - struct snd_pcm *pcm; - struct snd_pcm_runtime *runtime; - unsigned int bits; - snd_pcm_uframes_t frames; - - *pp_dma = NULL; - dma = kzalloc(sizeof(*dma), GFP_KERNEL); - memset((void *)dma, 0, sizeof(*dma)); - - dma_convert_to_hda_format(stream_format, format); - dma->m_converter_format = *format; - - dma->substream = NULL; - pcm = codec->pcm_info->pcm; - for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - substream; substream = substream->next) { - if (codec->pcm_info->pcm_type == HDA_PCM_TYPE_SPDIF) - continue; - - if (!SUBSTREAM_BUSY(substream)) { - dma->substream = substream; - break; - } - } - - if (NULL == dma->substream) { - kfree(dma); - return -1; - } - - runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); - memset((void *)runtime, 0, sizeof(*runtime)); - dma->substream->runtime = runtime; - dma->substream->private_data = pcm->private_data; - - azx_pcm_open(dma->substream); - req_size = req_size * 2; - snd_pcm_lib_malloc_pages(dma->substream, req_size); - - runtime->rate = stream_format->sample_rate; - runtime->channels = stream_format->number_channels; - runtime->format = SNDRV_PCM_FORMAT_S32_LE; - runtime->no_period_wakeup = 1; - - bits = snd_pcm_format_physical_width(runtime->format); - runtime->sample_bits = bits; - bits *= runtime->channels; - runtime->frame_bits = bits; - frames = 1; - while (bits % 8 != 0) { - bits *= 2; - frames *= 2; - } - runtime->byte_align = bits / 8; - runtime->min_align = frames; - runtime->buffer_size = bytes_to_frames(runtime, req_size); - runtime->period_size = runtime->buffer_size; - dma->m_req_size = req_size; - dma->codec = codec; - - *pp_dma = dma; - CA0132_LOG("dma_init: succeeded.\n"); - return 0; -} - -static int dma_prepare(struct dma_engine *dma) -{ - struct snd_pcm_runtime *runtime; - int err; - - CA0132_LOG("dma_prepare: begin\n"); - runtime = dma->substream->runtime; - - err = azx_pcm_prepare(dma->substream); - if (err < 0) - return -1; - - dma->m_buffer_size = snd_pcm_lib_buffer_bytes(dma->substream); - dma->m_buffer_addr = runtime->dma_area; - - return 0; -} - -static int dma_reset(struct dma_engine *dma) -{ - struct snd_pcm_runtime *runtime = dma->substream->runtime; - - CA0132_LOG("dma_reset: begin\n"); - azx_pcm_hw_free(dma->substream); - snd_pcm_lib_malloc_pages(dma->substream, dma->m_req_size); - - azx_pcm_prepare(dma->substream); - dma->m_buffer_size = snd_pcm_lib_buffer_bytes(dma->substream); - dma->m_buffer_addr = runtime->dma_area; - - return 0; -} - static int dma_set_state(struct dma_engine *dma, enum dma_state state) { int cmd; @@ -2100,9 +1984,6 @@ static int dma_set_state(struct dma_engine *dma, enum dma_state state) CA0132_LOG("dma_set_state state=%d\n", state); switch (state) { - case DMA_STATE_RESET: - dma_reset(dma); - return 0; case DMA_STATE_STOP: cmd = SNDRV_PCM_TRIGGER_STOP; break; @@ -2113,9 +1994,7 @@ static int dma_set_state(struct dma_engine *dma, enum dma_state state) return 0; } - azx_pcm_trigger(dma->substream, cmd); - - return 0; + return load_dsp_trigger(dma->codec->bus, cmd); } static unsigned int dma_get_buffer_size(struct dma_engine *dma) @@ -2128,15 +2007,6 @@ static unsigned int *dma_get_buffer_addr(struct dma_engine *dma) return dma->m_buffer_addr; } -static int dma_free_buffer(struct dma_engine *dma) -{ - azx_pcm_hw_free(dma->substream); - azx_pcm_close(dma->substream); - kfree(dma->substream->runtime); - dma->substream->runtime = NULL; - return 0; -} - static int dma_xfer(struct dma_engine *dma, const unsigned int *data, unsigned int count) @@ -2160,13 +2030,6 @@ static unsigned int dma_get_stream_id(struct dma_engine *dma) return spec->dsp_stream_id; } -static int dma_exit(struct dma_engine *dma) -{ - CA0132_LOG("dma_exit\n"); - kfree(dma); - return 0; -} - /* * CA0132 chip DSP image segment stuffs */ @@ -2384,7 +2247,6 @@ static int dspxfr_one_seg(struct hda_codec *codec, } CA0132_DSP_LOG("+++++ DMA complete"); dma_set_state(dma_engine, DMA_STATE_STOP); - dma_set_state(dma_engine, DMA_STATE_RESET); } CTASSERT(run_size_words <= words_to_write); @@ -2413,17 +2275,29 @@ static int dspxfr_image(struct hda_codec *codec, struct dma_engine *dma_engine; unsigned int dma_chan; unsigned int port_map_mask; - unsigned int buf_size; CTASSERT(fls_data != NULL); if (fls_data == NULL) return -1; - buf_size = ovly ? DSP_DMA_WRITE_BUFLEN_OVLY : DSP_DMA_WRITE_BUFLEN_INIT; - status = dma_init(codec, &dma_engine, format, &hda_format, buf_size); - - if (FAILED(status)) - return -1; + dma_engine = kzalloc(sizeof(*dma_engine), GFP_KERNEL); + if (!dma_engine) { + status = -ENOMEM; + goto exit; + } + memset((void*)dma_engine, 0, sizeof(*dma_engine)); + + dma_engine->codec = codec; + dma_convert_to_hda_format(format, &hda_format); + dma_engine->m_converter_format = hda_format; + dma_engine->m_buffer_size = + ovly ? DSP_DMA_WRITE_BUFLEN_OVLY : DSP_DMA_WRITE_BUFLEN_INIT; + dma_engine->m_buffer_addr = kzalloc(dma_engine->m_buffer_size, + GFP_KERNEL); + if (!dma_engine->m_buffer_addr) { + status = -ENOMEM; + goto exit; + } dma_chan = 0; do { @@ -2434,7 +2308,10 @@ static int dspxfr_image(struct hda_codec *codec, break; } - status = dma_prepare(dma_engine); + status = load_dsp_prepare(codec->bus, + dma_engine->m_converter_format, + dma_engine->m_buffer_addr, + dma_engine->m_buffer_size); if (FAILED(status)) break; @@ -2495,15 +2372,9 @@ static int dspxfr_image(struct hda_codec *codec, if (ovly && (dma_chan != INVALID_DMA_CHANNEL)) status = dspio_free_dma_chan(codec, dma_chan); - status = dma_set_state(dma_engine, DMA_STATE_RESET); - if (FAILED(status)) - status = FAIL_MSG(status, "dma set state Reset fail"); - - status = dma_free_buffer(dma_engine); - if (FAILED(status)) - status = FAIL_MSG(status, "dma free buffer fail"); - - dma_exit(dma_engine); +exit: + kfree(dma_engine->m_buffer_addr); + kfree(dma_engine); return status; } -- 1.7.4.1

2 6

Re: [alsa-devel] [pulseaudio-discuss] [PATCH v0 07/20] Consider unknown availability in module-switch-on-port-available
by Raymond Yau 22 Aug '12

22 Aug '12

2012-8-22 下午5:13 於 "David Henningsson" <david.henningsson(a)canonical.com> 寫道： > > On 08/22/2012 10:08 AM, Mikel Astiz wrote: >> >> From: Mikel Astiz <mikel.astiz(a)bmw-carit.de> >> >> A port with the availability state flag set to unknown should not be >> considered an active port. >> --- >> src/modules/module-switch-on-port-available.c | 2 +- >> 1 files changed, 1 insertions(+), 1 deletions(-) >> >> diff --git a/src/modules/module-switch-on-port-available.c b/src/modules/module-switch-on-port-available.c >> index 1da612d..8c16821 100644 >> --- a/src/modules/module-switch-on-port-available.c >> +++ b/src/modules/module-switch-on-port-available.c >> @@ -85,7 +85,7 @@ static pa_bool_t try_to_switch_profile(pa_card *card, pa_device_port *port) { >> PA_IDXSET_FOREACH(sink, card->sinks, state2) { >> if (!sink->active_port) >> continue; >> - if (sink->active_port->available != PA_PORT_AVAILABLE_NO) >> + if (sink->active_port->available == PA_PORT_AVAILABLE_YES) >> found_active_port = TRUE; >> } >> >> > > I think this will break the following use case (on a normal laptop) : > > Assume we plug in HDMI. With this patch we will get inconsistent behaviour: if headphones are plugged in (available = yes), we won't switch to HDMI, but if not (speakers are available = unknown), we will switch to HDMI. > > if you look at those bug reports of no sound for those on-board ac97 codecs with ubuntu 12.04 https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1033996 how does pulseaudio determine which controls are effective in the headphone path and speaker path of those ac97 codecs ? the analog-output-stereo profile for the line out jack is missing ( 0.056| 0.000) D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-headphone is there any solution for those snd-intel8x0 and those onboard ac97 codecs with headphone playback volume control ( no ctljack control and no phantom port)? http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… http://git.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/al…

2 1

[alsa-devel] [PATCH v3 1/2] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 22 Aug '12

22 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gernel sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled, and if all codecs are suspended (in D3), the HDA controller can also be suspended to a low power state by decreasing the power usage counter. And the user IO operation can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. TODO: Suspend the controller only if all codecs support CLKSTOP in D3, for pass-through operations and wakeup. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 52 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 49 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..4c30fd7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2402,11 +2406,19 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) + if (power_on) { + if (!chip->initialized) + pm_runtime_get_sync(&chip->pci->dev); azx_init_chip(chip, 1); + } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) + !bus->power_keep_link_on) { azx_stop_chip(chip); + + /* TODO: Suspend controller only if all codec support + stop-clock in D3, for wakeup consideration */ + pm_runtime_put_sync(&chip->pci->dev); + } } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2523,30 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3238,6 +3273,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3327,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3342,13 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + /* undo 'get' in azx_probe_continue() */ + pm_runtime_put_noidle(&pci->dev); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

1 0

Re: [alsa-devel] [PATCH 1/5] ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific
by Hebbar, Gururaja 22 Aug '12

22 Aug '12

On Fri, Aug 17, 2012 at 23:43:44, Nori, Sekhar wrote: > Hi Gururaja, > > On 8/14/2012 6:12 PM, Hebbar, Gururaja wrote: > > Davinci McASP header & driver are shared by few OMAP platforms (like > > TI81xx, AM335x). Splitting asp header into Davinci platform specific > > header and Audio specific header helps to share them across platforms. > > > > Audio specific defines is moved to to common > > <linux/platform_data/davinci_asp.h> so that the header can be > > accessed by all related platforms. > > > > While here, correct the header usage (remove multiple header > > re-definitions and unused headers) and remove platform names from > > structures comments and enum > > > > Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> > > Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> > > This patch looks good to me. See a minor nit below. > > Acked-by: Sekhar Nori <nsekhar(a)ti.com> > > > diff --git a/arch/arm/mach-davinci/asp.h b/arch/arm/mach-davinci/asp.h > > new file mode 100644 > > index 0000000..8fd5238 > > --- /dev/null > > +++ b/arch/arm/mach-davinci/asp.h > > @@ -0,0 +1,49 @@ > > +/* > > + * TI DaVinci Audio definitions > > + */ > > +#ifndef __ASM_ARCH_DAVINCI_ASP_H > > +#define __ASM_ARCH_DAVINCI_ASP_H > > + > > +/* Bases of dm644x and dm355 register banks */ > > +#define DAVINCI_ASP0_BASE 0x01E02000 > > +#define DAVINCI_ASP1_BASE 0x01E04000 > > + > > +/* Bases of dm365 register banks */ > > +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 > > + > > +/* Bases of dm646x register banks */ > > +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 > > +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 > > Can you remove usage of tabs after 'define'. I understand this is coming > from existing code. Ok. I have got few more comments from Mark. Will add this into it. > > Thanks, > Sekhar > Regards, Gururaja

1 0

[alsa-devel] [RFC] Channel mapping API
by Takashi Iwai 21 Aug '12

21 Aug '12

Hi, this is a progress report of my longstanding TODO, the channel map API implementation. I'm going to cover this at Plumbers audio uconf, so we can discuss details there, too. The channel mapping API provides a method for user-space to query, get and set the channel map of a PCM stream. It's required for assigning channels properly for multi-channel streams. * KERNEL IMPLEMENTATION In my latest attempt, I implemented with control elements. A control element is created for each PCM substream with the corresponding device and substream index. Then it gives a TLV for querying maps, a read op for obtaining the current map, and optionally a write op for setting the map. The obvious merit by this way is that no extra kernel ABI is required. A couple of new helper functions are provided for assigning standard channel maps. Currently, HD-audio and AC97 drivers has some implementation. * ALSA-LIB IMPLEMENTATION The additional alsa-lib API functions look like: int **snd_pcm_query_chmaps(snd_pcm_t *pcm); int *snd_pcm_get_chmap(snd_pcm_t *pcm); int snd_pcm_set_chmap(snd_pcm_t *pcm, const int *map); snd_pcm_query_chmaps() returns the list of channel maps. A channel map is represented by an integer array, beginning with the channel map type, followed by the number of channels, and the position of each channel, e.g. { SND_CHMAP_FIXED, 4, SND_CHMAP_FL, SND_CHMAP_FR, SND_CHMAP_RL, SND_CHMAP_RR } snd_pcm_get_chmap() returns the currently assigned channel map for the given PCM stream. If the PCM is before prepared, it fills UNKNOWN. When a driver allows user to change the channel map, user can call snd_pcm_set_chmap(). For example, HDMI allows you to choose whether it's 4.0 or 3.1 output. * RESOURCES The latest kernel patchset is found in sound-unstable tree topic/tlv-chmap branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git topic/tlv-chmap The alsa-lib portion is found in a tree on github topic/chmap branch. git://github.com/tiwai/alsa-lib.git topic/chmap thanks, Takashi

6 31

[alsa-devel] [PATCH] ASoC: wm_hubs: Allow configuration of MICBIAS power up delay via pdata
by Mark Brown 21 Aug '12

21 Aug '12

Sometimes the analogue circuitry connected to the microphone needs some time to settle after power up. Allow systems to configure this delay in the platform data, the driver will then insert the required delay during power up of paths that involve the microphone. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/pdata.h | 4 ++++ include/sound/wm8993.h | 4 ++++ sound/soc/codecs/wm8993.c | 2 ++ sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 35 +++++++++++++++++++++++++++++++---- sound/soc/codecs/wm_hubs.h | 4 ++++ 6 files changed, 47 insertions(+), 4 deletions(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index f0361c0..fc87be4 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -164,6 +164,10 @@ struct wm8994_pdata { int num_micd_rates; struct wm8958_micd_rate *micd_rates; + /* Power up delays to add after microphone bias power up (ms) */ + int micb1_delay; + int micb2_delay; + /* LINEOUT can be differential or single ended */ unsigned int lineout1_diff:1; unsigned int lineout2_diff:1; diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h index eee19f6..8016fd8 100644 --- a/include/sound/wm8993.h +++ b/include/sound/wm8993.h @@ -32,6 +32,10 @@ struct wm8993_platform_data { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* Delay to add for microphones to stabalise after power up */ + int micbias1_delay; + int micbias2_delay; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 9fd80d6..94737a3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1520,6 +1520,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->pdata.lineout2fb, wm8993->pdata.jd_scthr, wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_delay, + wm8993->pdata.micbias2_delay, wm8993->pdata.micbias1_lvl, wm8993->pdata.micbias2_lvl); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2d60ed3..a51431c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3233,6 +3233,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) pdata->lineout2fb, pdata->jd_scthr, pdata->jd_thr, + pdata->micb1_delay, + pdata->micb2_delay, pdata->micbias1_lvl, pdata->micbias2_lvl); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 8dc4e89..c72eaef 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -859,6 +859,28 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + switch (w->shift) { + case WM8993_MICB1_ENA_SHIFT: + if (hubs->micb1_delay) + msleep(hubs->micb1_delay); + break; + case WM8993_MICB2_ENA_SHIFT: + if (hubs->micb2_delay) + msleep(hubs->micb2_delay); + break; + default: + return -EINVAL; + } + + return 0; +} + void wm_hubs_update_class_w(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); @@ -1049,8 +1071,10 @@ SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), SND_SOC_DAPM_INPUT("IN2RP:VXRP"), -SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), @@ -1388,13 +1412,16 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, - int jd_scthr, int jd_thr, int micbias1_lvl, - int micbias2_lvl) + int jd_scthr, int jd_thr, + int micbias1_delay, int micbias2_delay, + int micbias1_lvl, int micbias2_lvl) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); hubs->lineout1_se = !lineout1_diff; hubs->lineout2_se = !lineout2_diff; + hubs->micb1_delay = micbias1_delay; + hubs->micb2_delay = micbias2_delay; if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 14927f6..704e53f 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -55,6 +55,9 @@ struct wm_hubs_data { struct list_head dcs_cache; bool (*check_class_w_digital)(struct snd_soc_codec *); + int micb1_delay; + int micb2_delay; + bool lineout1_se; bool lineout1n_ena; bool lineout1p_ena; @@ -75,6 +78,7 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, int jd_scthr, int jd_thr, + int micbias1_dly, int micbias2_dly, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); -- 1.7.10.4

1 0

Re: [alsa-devel] mx27_3ds : Sound doesn't work anymore on linux-next
by Gaëtan Carlier 21 Aug '12

21 Aug '12

Just some more informations, process enter in D+ state and this is so impossible to kill them. On 08/17/2012 05:16 PM, Gaëtan Carlier wrote: > Hi, > It is several days that I tried to make sound work on mx27_3ds platform > from linux-next (daily cloned/pulled) but each time I try start start > aplay or alsamixer, the process freeze and CTRL+C has not effect. I can > open another SSH session and execute other commands but alsa commands > always freeze. > The sound card is correctly installer at boot : > > imx_mc13783 imx_mc13783.0: mc13783-hifi <-> imx-ssi.0 mapping ok > ALSA device list: > #0: imx_mc13783 > > I only apply the patch "[PATCH 1/2] ARM: mx27pdk: Add audio support". I > attach my config file. > > I try to apply patch on a previous commit when I told that this patch > was working but it didn't succeed the is still freezing. > I try to make it working on another mx27-based platform but same > problem. I try to update Alsa from version 1.0.23 to 1.0.25 but same > behaviour. > What is wrong ? > > Regards, > Gaëtan Carlier. Regards, Gaëtan.

1 1

Re: [alsa-devel] [PATCH] ASoC: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 04:51:59PM +0200, Emil Goode wrote: > This patch removes a unused struct snd_soc_dai. > > Sparse is warning: > sound/soc/soc-compress.c:156:22: warning: > unused variable ‘codec_dai’ [-Wunused-variable] Already fixed by someone else.

1 0

[alsa-devel] [PATCH 0/3] ASoC: omap-mcbsp: Sidetone related changes
by Peter Ujfalusi 21 Aug '12

21 Aug '12

Hello, When we boot with DT blob the sidetone is not supported at the moment. Prepare the McBSP code for the time when we are going to be able to support ST with DT booted kernel. Meanwhile do not block the audio card if the st_data is missing from the port, but print a warning about it. This will allow us to debug the DT booted kernels with audio. Mark: This series as such does not depend on the McBSP DT support, but it can be attached to that branch. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcbsp: Check mcbsp->id instead of cpu_dai->id when adding ST controls ASoC: omap-mcbsp: Only print warning if the st_data is missing for the port ASoC: omap-mcbsp: Use macro to create the McBSP2/3 ST controls sound/soc/omap/omap-mcbsp.c | 47 ++++++++++++++++++------------------------ 1 files changed, 20 insertions(+), 27 deletions(-) -- 1.7.8.6

1 4

Re: [alsa-devel] [PATCH] ASoC: soc-compress: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 12:20:22PM +0530, Sachin Kamat wrote: > codec_dai is not used in the function. > > sound/soc/soc-compress.c: In function ‘soc_compr_set_params’: > sound/soc/soc-compress.c:156:22: warning: > unused variable ‘codec_dai’ [-Wunused-variable] Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: compress - fix code alignment
by Vinod Koul 21 Aug '12

21 Aug '12

Reported-by: Fengguang Wu <wfg(a)linux.intel.com> Signed-off-by: Namarta Kohli <namartax.kohli(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/soc/soc-compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 88d85ba..cc0562d 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -104,7 +104,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->free) platform->driver->compr_ops->free(cstream); - cpu_dai->runtime = NULL; + cpu_dai->runtime = NULL; if (cstream->direction == SND_COMPRESS_PLAYBACK) { if (!rtd->pmdown_time || codec->ignore_pmdown_time || -- 1.7.0.4

2 1

Re: [alsa-devel] [PATCH] ASoC: soc-compress: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 10:48:30AM -0300, Fabio Estevam wrote: > Remove unused variable ‘codec_dai’ and fix the following warning: > > sound/soc/soc-compress.c: In function ‘soc_compr_set_params’: > sound/soc/soc-compress.c:156:22: warning: unused variable ‘codec_dai’ [-Wunused-variable] Someone already sent a patch for this.

1 0

[alsa-devel] [PATCH 08/22] ASoC: codecs: Enable AB8500 CODEC for Device Tree
by Lee Jones 21 Aug '12

21 Aug '12

We continue to allow the AB8500 CODEC to be registered via the AB8500 Multi Functional Device API, only this time we extract its configuration from the Device Tree binary. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- include/linux/mfd/abx500/ab8500-codec.h | 6 ++- sound/soc/codecs/ab8500-codec.c | 81 +++++++++++++++++++++++++++++++ 2 files changed, 85 insertions(+), 2 deletions(-) diff --git a/include/linux/mfd/abx500/ab8500-codec.h b/include/linux/mfd/abx500/ab8500-codec.h index dc65292..d707941 100644 --- a/include/linux/mfd/abx500/ab8500-codec.h +++ b/include/linux/mfd/abx500/ab8500-codec.h @@ -23,7 +23,8 @@ enum amic_type { /* Mic-biases */ enum amic_micbias { AMIC_MICBIAS_VAMIC1, - AMIC_MICBIAS_VAMIC2 + AMIC_MICBIAS_VAMIC2, + AMIC_MICBIAS_UNKNOWN }; /* Bias-voltage */ @@ -31,7 +32,8 @@ enum ear_cm_voltage { EAR_CMV_0_95V, EAR_CMV_1_10V, EAR_CMV_1_27V, - EAR_CMV_1_58V + EAR_CMV_1_58V, + EAR_CMV_UNKNOWN }; /* Analog microphone settings */ diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 23b4018..fa80961 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -34,6 +34,7 @@ #include <linux/mfd/abx500/ab8500-sysctrl.h> #include <linux/mfd/abx500/ab8500-codec.h> #include <linux/regulator/consumer.h> +#include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> @@ -2394,9 +2395,65 @@ struct snd_soc_dai_driver ab8500_codec_dai[] = { } }; +static void ab8500_codec_of_probe(struct device *dev, struct device_node *np, + struct ab8500_codec_platform_data *codec) +{ + u32 value; + + if (of_get_property(np, "stericsson,amic1-type-single-ended", NULL)) + codec->amics.mic1_type = AMIC_TYPE_SINGLE_ENDED; + else + codec->amics.mic1_type = AMIC_TYPE_DIFFERENTIAL; + + if (of_get_property(np, "stericsson,amic2-type-single-ended", NULL)) + codec->amics.mic2_type = AMIC_TYPE_SINGLE_ENDED; + else + codec->amics.mic2_type = AMIC_TYPE_DIFFERENTIAL; + + /* Has a non-standard Vamic been requested? */ + if (of_get_property(np, "stericsson,amic1a-bias-vamic2", NULL)) + codec->amics.mic1a_micbias = AMIC_MICBIAS_VAMIC2; + else + codec->amics.mic1a_micbias = AMIC_MICBIAS_VAMIC1; + + if (of_get_property(np, "stericsson,amic1b-bias-vamic2", NULL)) + codec->amics.mic1b_micbias = AMIC_MICBIAS_VAMIC2; + else + codec->amics.mic1b_micbias = AMIC_MICBIAS_VAMIC1; + + if (of_get_property(np, "stericsson,amic2-bias-vamic1", NULL)) + codec->amics.mic2_micbias = AMIC_MICBIAS_VAMIC1; + else + codec->amics.mic2_micbias = AMIC_MICBIAS_VAMIC2; + + if (!of_property_read_u32(np, "stericsson,earpeice-cmv", &value)) { + switch (value) { + case 950 : + codec->ear_cmv = EAR_CMV_0_95V; + break; + case 1100 : + codec->ear_cmv = EAR_CMV_1_10V; + break; + case 1270 : + codec->ear_cmv = EAR_CMV_1_27V; + break; + case 1580 : + codec->ear_cmv = EAR_CMV_1_58V; + break; + default : + codec->ear_cmv = EAR_CMV_UNKNOWN; + dev_err(dev, "Unsuitable earpiece voltage found in DT\n"); + } + } else { + dev_warn(dev, "No earpiece voltage found in DT - using default\n"); + codec->ear_cmv = EAR_CMV_0_95V; + } +} + static int ab8500_codec_probe(struct snd_soc_codec *codec) { struct device *dev = codec->dev; + struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); struct ab8500_platform_data *pdata; struct filter_control *fc; @@ -2407,6 +2464,30 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + if (np) { + if (!pdata) + pdata = devm_kzalloc(dev, + sizeof(struct ab8500_platform_data), + GFP_KERNEL); + + if (!pdata->codec) + pdata->codec + = devm_kzalloc(dev, + sizeof(struct ab8500_codec_platform_data), + GFP_KERNEL); + + if (!(pdata && pdata->codec)) + return -ENOMEM; + + ab8500_codec_of_probe(dev, np, pdata->codec); + + } else { + if (!(pdata && pdata->codec)) { + dev_err(dev, "No codec platform data or DT found\n"); + return -EINVAL; + } + } + /* Inform SoC Core that we have our own I/O arrangements. */ codec->control_data = (void *)true; -- 1.7.9.5

4 8

Re: [alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 07:39:55AM -0500, Clark, Rob wrote: > Does ASoC support 'hotplug' of audio devices? If so, maybe it makes > some sense to have some support in drm core. At least all the edid > parsing stuff to determine if the display supports audio should be > generic and not driver specific. Not really (except on a card level), and it'd probably confuse most userspaces. What I'd expect to happen currently is that the current state of the connector would get reported via extcon.

1 0

[alsa-devel] [PATCH 0/5] hda-emu: Add playback/capture test to test suite
by David Henningsson 21 Aug '12

21 Aug '12

When I added those tests, I did not discover anything that needed to be fixed in the driver code, but there were some additional things in hda-emu to fix up. David Henningsson (5): hda-emu: Store pointers to pcm streams instead of their content hda-emu: Add support for get/set converter channel count hda-emu: Add CX20585 to modem whitelist hda-emu: Add playback/capture test to test suite hda-emu: improve test suite summary script hda-emu.c | 24 ++++++++++++------------ hda-int.c | 15 +++++++++++++++ hda-parse.c | 1 + include/hda-types.h | 1 + tester/runall.sh | 3 --- tester/runner.py | 25 ++++++++++++++++++++++--- tester/summary.py | 9 +++++++++ 7 files changed, 60 insertions(+), 18 deletions(-) delete mode 100755 tester/runall.sh -- 1.7.9.5

2 6

[alsa-devel] Can module snd-aloop (virtual looback device) be made clock slave of a real audio device ?
by James Warden 21 Aug '12

21 Aug '12

Hello alsa-devel's, A while back, I had written a rather hackish method to hook up the ALSA virtual loopback device to jack (jackaudio.org) via Jack clients alsa_in / alsa_out so that non jack-aware applications that can however use ALSA devices "directly" (via alsa-lib) can have their audio stream redirected to jack (see my rather lengthy prose at http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge). There are a few advantages in this method (depends on your use-cases of course). But it could be improved (or so I think) by removing the following drawbacks: The alsa_in/out jack clients are resampling the audio stream to synchronize with the device that jack is using. The resampling, done by libsamplerate, takes some CPU power. Since many users are using my recipe on laptops, this is not always welcome. Because snd-aloop is a virtual device, couldn't it be made a clock slave of the h/w device that jack is operating on ? This would in principle remove the need for resampling. I could imagine that the module could be passed an option at loading time (e.g. modprobe snd-aloop clock-master=<real h/w device index> or something like that) Since I have no clue as to the kernel code required to implement / extend snd-aloop, I would like some opinions about this. Thanks for any hints. J.

2 3

[alsa-devel] Query on PCM lib: wait_for_avail() API
by Sarawadi, Ravishankar 21 Aug '12

21 Aug '12

Greetings, I recently started working with ALSA architecture. I am very excited to be part of ALSA group and this is my first query as ALSA developer. This is in regards to function wait_for_avail() defined in pcm_lib.c If audio path is broken the wait_time inside this function is 10 seconds. wait_time = msecs_to_jiffies(wait_time * 1000); I have a use case where I need to force exit out of this wait when audio hardware path is broken. Waiting for 10 seconds is too long and causing synchronization issues. My Use-case: 1 PCM Audio playback is going on. PCM write is a blocking call. Waiting on DSP WRITE-DONE event. 2 Audio hardware path failure 3 PCM driver notified of hardware path failure. 4 Meanwhile DSP ring buffer gets full as buffers are not getting consumed. 5 Next WRITE is blocked on wait_for_avail() buffer call. 6 PCM close from driver as part process to handle audio path failure is blocked on Step 5 I am trying to figure out a way to exit out of this wait forcefully by some external trigger. Another option is to reduce this wait time to less than 5 seconds. Could someone please recommend the best way to handle this use case? Thanks, RaviShankar (Ravi)

2 1

[alsa-devel] [PATCH] introduce snd-cmi8328: C-Media CMI8328 driver
by Ondrej Zary 21 Aug '12

21 Aug '12

Introduce snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as AudioExcel AV500. It supports PCM playback and capture (full-duplex) through wss_lib, gameport, OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected to the MPU401 port and Aux 1 input internally which works too. The CDROM interface is not supported (as the drivers for these CDROMs were removed from the kernel some time ago). A separate driver is needed because CMI8328 is completely different chip to CMI8329/CMI8330. It's configured by magic registers (there's no PnP). Sound is provided by a real WSS codec (CS4231A) and the SB part is just a SB Pro emulation (for DOS games, useless for Linux). When SB is enabled, the CMI8328 chip disables access to the WSS codec, emulates SoundBlaster on one side and outputs sound data to the codec - so SB and WSS can't work together with this card. The WSS codec can do full duplex by itself so there's no need for crazy things like snd-cmi8330 does (combining SB and WSS parts into one driver). Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> commit ebd26fe68c73eb979ace14b51a4fbdd899233952 Author: Ondrej Zary <linux(a)rainbow-software.org> Date: Thu Aug 2 09:17:57 2012 +0200 introduce snd-cmi8328: C-Media CMI8328 driver Introduce snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as AudioExcel AV500. It supports PCM playback and capture (full-duplex) through wss_lib, gameport, OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected to the MPU401 port and Aux 1 input internally which works too. The CDROM interface is not supported (as the drivers for these CDROMs were removed from the kernel some time ago). diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 52064cf..a38d964 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -117,6 +117,18 @@ config SND_AZT2320 To compile this driver as a module, choose M here: the module will be called snd-azt2320. +config SND_CMI8328 + tristate "C-Media CMI8328" + select SND_WSS_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for soundcards based on the + C-Media CMI8328 chip. + + To compile this driver as a module, choose M here: the module + will be called snd-cmi8328. + config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 8d781e4..9a15f14 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -6,6 +6,7 @@ snd-adlib-objs := adlib.o snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o +snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o @@ -16,6 +17,7 @@ snd-sscape-objs := sscape.o obj-$(CONFIG_SND_ADLIB) += snd-adlib.o obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o +obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c new file mode 100644 index 0000000..aefafff --- /dev/null +++ b/sound/isa/cmi8328.c @@ -0,0 +1,482 @@ +/* + * Driver for C-Media CMI8328-based soundcards, such as AudioExcel AV500 + * Copyright (c) 2012 Ondrej Zary + * + * AudioExcel AV500 card consists of: + * - CMI8328 - main chip (SB Pro emulation, gameport, OPL3, MPU401, CD-ROM) + * - CS4231A - WSS codec + * - Dream SAM9233+GMS950400+RAM+ROM: Wavetable MIDI, connected to MPU401 + */ + +#include <linux/init.h> +#include <linux/isa.h> +#include <linux/module.h> +#include <linux/gameport.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/wss.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> +#define SNDRV_LEGACY_FIND_FREE_IOPORT +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Ondrej Zary <linux(a)rainbow-software.org>"); +MODULE_DESCRIPTION("C-Media CMI8328"); +MODULE_LICENSE("GPL"); + +#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) +#define SUPPORT_JOYSTICK 1 +#endif + +/* I/O port is configured by jumpers on the card to one of these */ +static int cmi8328_ports[] = { 0x530, 0xe80, 0xf40, 0x604 }; +#define CMI8328_MAX ARRAY_SIZE(cmi8328_ports) + +static int index[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = -1}; +static char *id[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = NULL}; +static long port[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int irq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +static int dma1[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static int dma2[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static long mpuport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int mpuirq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +#ifdef SUPPORT_JOYSTICK +static bool gameport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = true}; +#endif + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for CMI8328 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for CMI8328 soundcard."); + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for CMI8328 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for CMI8328 driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 for CMI8328 driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 for CMI8328 driver."); + +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8328 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8328 MPU-401 port."); +#ifdef SUPPORT_JOYSTICK +module_param_array(gameport, bool, NULL, 0444); +MODULE_PARM_DESC(gameport, "Enable gameport."); +#endif + +struct snd_cmi8328 { + u16 port; + u8 cfg[3]; + u8 wss_cfg; + struct snd_card *card; + struct snd_wss *wss; +#ifdef SUPPORT_JOYSTICK + struct gameport *gameport; +#endif +}; + +/* CMI8328 configuration registers */ +#define CFG1 0x61 +#define CFG1_SB_DISABLE (1 << 0) +#define CFG1_GAMEPORT (1 << 1) +/* + * bit 0: SB: 0=enabled, 1=disabled + * bit 1: gameport: 0=disabled, 1=enabled + * bits 2-4: SB IRQ: 001=3, 010=5, 011=7, 100=9, 101=10, 110=11 + * bits 5-6: SB DMA: 00=disabled (when SB disabled), 01=DMA0, 10=DMA1, 11=DMA3 + * bit 7: SB port: 0=0x220, 1=0x240 + */ +#define CFG2 0x62 +#define CFG2_MPU_ENABLE (1 << 2) +/* + * bits 0-1: CD-ROM mode: 00=disabled, 01=Panasonic, 10=Sony/Mitsumi/Wearnes, + 11=IDE + * bit 2: MPU401: 0=disabled, 1=enabled + * bits 3-4: MPU401 IRQ: 00=3, 01=5, 10=7, 11=9, + * bits 5-7: MPU401 port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x332, + 101=0x334, 110=0x336 + */ +#define CFG3 0x63 +/* + * bits 0-2: CD-ROM IRQ: 000=disabled, 001=3, 010=5, 011=7, 100=9, 101=10, + 110=11 + * bits 3-4: CD-ROM DMA: 00=disabled, 01=DMA0, 10=DMA1, 11=DMA3 + * bits 5-7: CD-ROM port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x340, + 101=0x350, 110=0x360, 111=0x370 + */ + +static u8 snd_cmi8328_cfg_read(u16 port, u8 reg) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + return inb(port); +} + +static void snd_cmi8328_cfg_write(u16 port, u8 reg, u8 val) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + outb(val, port + 3); /* yes, value goes to the same port as index */ +} + +static void snd_cmi8328_cfg_save(u16 port, u8 cfg[]) +{ + cfg[0] = snd_cmi8328_cfg_read(port, CFG1); + cfg[1] = snd_cmi8328_cfg_read(port, CFG2); + cfg[2] = snd_cmi8328_cfg_read(port, CFG3); +} + +static void snd_cmi8328_cfg_restore(u16 port, u8 cfg[]) +{ + snd_cmi8328_cfg_write(port, CFG1, cfg[0]); + snd_cmi8328_cfg_write(port, CFG2, cfg[1]); + snd_cmi8328_cfg_write(port, CFG3, cfg[2]); +} + +static int __devinit snd_cmi8328_mixer(struct snd_wss *chip) +{ + struct snd_card *card; + struct snd_ctl_elem_id id1, id2; + int err; + + card = chip->card; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* rename AUX0 switch to CD */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX0 volume to CD */ + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 switch to Synth */ + strcpy(id1.name, "Aux Playback Switch"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 volume to Synth */ + strcpy(id1.name, "Aux Playback Volume"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + + return 0; +} + +/* find index of an item in "-1"-ended array */ +int array_find(int array[], int item) +{ + int i; + + for (i = 0; array[i] != -1; i++) + if (array[i] == item) + return i; + + return -1; +} +/* the same for long */ +int array_find_l(long array[], long item) +{ + int i; + + for (i = 0; array[i] != -1; i++) + if (array[i] == item) + return i; + + return -1; +} + +static int __devinit snd_cmi8328_probe(struct device *pdev, unsigned int ndev) +{ + struct snd_card *card; + struct snd_opl3 *opl3; + struct snd_cmi8328 *cmi; +#ifdef SUPPORT_JOYSTICK + struct resource *res; +#endif + int err, pos; + static long mpu_ports[] = { 0x330, 0x300, 0x310, 0x320, 0x332, 0x334, + 0x336, -1 }; + static u8 mpu_port_bits[] = { 3, 0, 1, 2, 4, 5, 6 }; + static int mpu_irqs[] = { 9, 7, 5, 3, -1 }; + static u8 mpu_irq_bits[] = { 3, 2, 1, 0 }; + static int irqs[] = { 9, 10, 11, 7, -1 }; + static u8 irq_bits[] = { 2, 3, 4, 1 }; + static int dma1s[] = { 3, 1, 0, -1 }; + static u8 dma_bits[] = { 3, 2, 1 }; + static int dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, {0, -1} }; + u16 port = cmi8328_ports[ndev]; + u8 val; + + /* 0xff is invalid configuration (but settable - hope it isn't set) */ + if (snd_cmi8328_cfg_read(port, CFG1) == 0xff) + return -ENODEV; + /* the SB disable bit must NEVER EVER be cleared or the WSS dies */ + snd_cmi8328_cfg_write(port, CFG1, CFG1_SB_DISABLE); + if (snd_cmi8328_cfg_read(port, CFG1) != CFG1_SB_DISABLE) + return -ENODEV; + /* disable everything first */ + snd_cmi8328_cfg_write(port, CFG2, 0); /* disable CDROM and MPU401 */ + snd_cmi8328_cfg_write(port, CFG3, 0); /* disable CDROM IRQ and DMA */ + + if (irq[ndev] == SNDRV_AUTO_IRQ) { + irq[ndev] = snd_legacy_find_free_irq(irqs); + if (irq[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (dma1[ndev] == SNDRV_AUTO_DMA) { + dma1[ndev] = snd_legacy_find_free_dma(dma1s); + if (dma1[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2[ndev] == SNDRV_AUTO_DMA) { + dma2[ndev] = snd_legacy_find_free_dma(dma2s[dma1[ndev] % 4]); + if (dma2[ndev] < 0) { + snd_printk(KERN_WARNING "unable to find a free DMA2, full-duplex will not work\n"); + dma2[ndev] = -1; + } + } + /* configure WSS IRQ... */ + pos = array_find(irqs, irq[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid IRQ %d\n", irq[ndev]); + return -EINVAL; + } + val = irq_bits[pos] << 3; + /* ...and DMA... */ + pos = array_find(dma1s, dma1[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA1 %d\n", dma1[ndev]); + return -EINVAL; + } + val |= dma_bits[pos]; + /* ...and DMA2 */ + if (dma2[ndev] >= 0 && dma1[ndev] != dma2[ndev]) { + pos = array_find(dma2s[dma1[ndev]], dma2[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA2 %d\n", dma2[ndev]); + return -EINVAL; + } + val |= 0x04; /* enable separate capture DMA */ + } + outb(val, port); + + err = snd_card_create(index[ndev], id[ndev], THIS_MODULE, + sizeof(struct snd_cmi8328), &card); + if (err < 0) + return err; + cmi = card->private_data; + cmi->card = card; + cmi->port = port; + cmi->wss_cfg = val; + snd_card_set_dev(card, pdev); + + err = snd_wss_create(card, port + 4, -1, irq[ndev], dma1[ndev], + dma2[ndev], WSS_HW_DETECT, 0, &cmi->wss); + if (err < 0) + goto error; + + err = snd_wss_pcm(cmi->wss, 0, NULL); + if (err < 0) + goto error; + + err = snd_wss_mixer(cmi->wss); + if (err < 0) + goto error; + err = snd_cmi8328_mixer(cmi->wss); + if (err < 0) + goto error; + + if (snd_wss_timer(cmi->wss, 0, NULL) < 0) + snd_printk(KERN_WARNING "error initializing WSS timer\n"); + + if (mpuport[ndev] == SNDRV_AUTO_PORT) { + mpuport[ndev] = snd_legacy_find_free_ioport(mpu_ports, 2); + if (mpuport[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); + } + if (mpuirq[ndev] == SNDRV_AUTO_IRQ) { + mpuirq[ndev] = snd_legacy_find_free_irq(mpu_irqs); + if (mpuirq[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); + } + /* enable and configure MPU401 */ + if (mpuport[ndev] > 0 && mpuirq[ndev] > 0) { + val = CFG2_MPU_ENABLE; + pos = array_find_l(mpu_ports, mpuport[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 port 0x%lx\n", + mpuport[ndev]); + else { + val |= mpu_port_bits[pos] << 5; + pos = array_find(mpu_irqs, mpuirq[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 IRQ %d\n", + mpuirq[ndev]); + else { + val |= mpu_irq_bits[pos] << 3; + snd_cmi8328_cfg_write(port, CFG2, val); + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, mpuport[ndev], + 0, mpuirq[ndev], NULL) < 0) + snd_printk(KERN_ERR "error initializing MPU401\n"); + } + } + } + /* OPL3 is hardwired to 0x388 and cannot be disabled */ + if (snd_opl3_create(card, 0x388, 0x38a, OPL3_HW_AUTO, 0, &opl3) < 0) + snd_printk(KERN_ERR "error initializing OPL3\n"); + else + if (snd_opl3_hwdep_new(opl3, 0, 1, NULL) < 0) + snd_printk(KERN_WARNING "error initializing OPL3 hwdep\n"); + + strcpy(card->driver, "CMI8328"); + strcpy(card->shortname, "C-Media CMI8328"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d,%d", + card->shortname, cmi->wss->port, irq[ndev], dma1[ndev], + (dma2[ndev] >= 0) ? dma2[ndev] : dma1[ndev]); + + dev_set_drvdata(pdev, card); + err = snd_card_register(card); + if (err < 0) + goto error; +#ifdef SUPPORT_JOYSTICK + if (!gameport[ndev]) + return 0; + /* gameport is hardwired to 0x200 */ + res = request_region(0x200, 8, "CMI8328 gameport"); + if (!res) + snd_printk(KERN_WARNING "unable to allocate gameport I/O port\n"); + else { + struct gameport *gp = cmi->gameport = gameport_allocate_port(); + if (!cmi->gameport) + release_and_free_resource(res); + else { + gameport_set_name(gp, "CMI8328 Gameport"); + gameport_set_phys(gp, "%s/gameport0", dev_name(pdev)); + gameport_set_dev_parent(gp, pdev); + gp->io = 0x200; + gameport_set_port_data(gp, res); + /* Enable gameport */ + snd_cmi8328_cfg_write(port, CFG1, + CFG1_SB_DISABLE | CFG1_GAMEPORT); + gameport_register_port(gp); + } + } +#endif + return 0; +error: + snd_card_free(card); + + return err; +} + +static int __devexit snd_cmi8328_remove(struct device *pdev, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(pdev); +#ifdef SUPPORT_JOYSTICK + struct snd_cmi8328 *cmi = card->private_data; + if (cmi->gameport) { + struct resource *res = gameport_get_port_data(cmi->gameport); + gameport_unregister_port(cmi->gameport); + release_and_free_resource(res); + } +#endif + /* disable everything */ + snd_cmi8328_cfg_write(cmi->port, CFG1, CFG1_SB_DISABLE); + snd_cmi8328_cfg_write(cmi->port, CFG2, 0); + snd_cmi8328_cfg_write(cmi->port, CFG3, 0); + snd_card_free(card); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#ifdef CONFIG_PM +static int snd_cmi8328_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_save(cmi->port, cmi->cfg); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(cmi->wss->pcm); + cmi->wss->suspend(cmi->wss); + + return 0; +} + +static int snd_cmi8328_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_restore(cmi->port, cmi->cfg); + outb(cmi->wss_cfg, cmi->port); + cmi->wss->resume(cmi->wss); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif + +static struct isa_driver snd_cmi8328_driver = { + .probe = snd_cmi8328_probe, + .remove = __devexit_p(snd_cmi8328_remove), +#ifdef CONFIG_PM + .suspend = snd_cmi8328_suspend, + .resume = snd_cmi8328_resume, +#endif + .driver = { + .name = "cmi8328" + }, +}; + +static int __init alsa_card_cmi8328_init(void) +{ + return isa_register_driver(&snd_cmi8328_driver, CMI8328_MAX); +} + +static void __exit alsa_card_cmi8328_exit(void) +{ + isa_unregister_driver(&snd_cmi8328_driver); +} + +module_init(alsa_card_cmi8328_init) +module_exit(alsa_card_cmi8328_exit) -- Ondrej Zary

2 1

[alsa-devel] [PATCH] snd-als100: fix suspend/resume
by Ondrej Zary 21 Aug '12

21 Aug '12

snd_card_als100_probe() does not set pcm field in struct snd_sb. As a result, PCM is not suspended and applications don't know that they need to resume the playback. Tested with Labway A381-F20 card (ALS120). Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev, irq[dev], dma8[dev], dma16[dev]); } - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { + if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) { snd_card_free(card); return error; } -- Ondrej Zary

2 1

[alsa-devel] [PATCH v2] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 21 Aug '12

21 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gernel sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled, and if all codecs are suspended (in D3), the HDA controller can also be suspended to a low power state by decreasing the power usage counter. And the user IO operation can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. TODO: Suspend the controller only if all codecs support CLKSTOP in D3, for pass-through operations and wakeup. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 55 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 52 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..cd02971 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2402,11 +2406,19 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) + if (power_on) { + if (!chip->initialized) + pm_runtime_get_sync(&chip->pci->dev); azx_init_chip(chip, 1); + } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) + !bus->power_keep_link_on) { azx_stop_chip(chip); + + /* TODO: Suspend controller only if all codec support + stop-clock in D3, for wakeup consideration */ + pm_runtime_put_sync(&chip->pci->dev); + } } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2523,33 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3238,6 +3276,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3330,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3345,13 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + /* undo 'get' in azx_probe_continue() */ + pm_runtime_put_noidle(&pci->dev); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

3 2

[alsa-devel] [RFC PATCH] ALSA: hda - add support for runtime power management
by mengdong.lin＠intel.com 21 Aug '12

21 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a basic idea about adding runtime PM support to HD audio. Support for runtime PM is based on current HDA power saving implementation. When both power saving and runtime PM are enabled, and if all codecs are suspended (in D3), the HDA controller can also be suspended to a low power state by decreasing the power usage counter and triggering PCI device runtime suspend. And the user IO operation can resume the controller back to D0. The user can enable runtime PM on HDA controller on platfroms that can provide acceptable latency on transition from D3 to D0. If the runtime PM is disabled, power saving just works as without runtime PM before. And if power saving is disabled, the power usage counter will never be 0 so the HDA controller will not be suspended. I've verified this patch on a platfrom on which the HD-A controller can enter D3hot when suspended. And I plan to allow the controller to suspened only if all codecs support "stop-clock" in D3, for wakeup request. Is this idea doable? Any comments are welcome. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_intel.c | 70 +++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 68 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..c88a95e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4383,6 +4383,7 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); + snd_printd("hda codec %d suspend\n", codec->addr); hda_call_codec_suspend(codec); if (bus->ops.pm_notify) bus->ops.pm_notify(bus); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..543a5c4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -2402,11 +2403,22 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) + if (power_on) { + if (!chip->initialized) { + snd_printd(SFX "request resume controller\n"); + pm_runtime_get_sync(&chip->pci->dev); + } azx_init_chip(chip, 1); + } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) + !bus->power_keep_link_on) { azx_stop_chip(chip); + + /* TODO: Suspend controller only if all codec support + stop-clock in D3, for wakeup consideration */ + snd_printd(SFX "request suspend controller\n"); + pm_runtime_put_sync(&chip->pci->dev); + } } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2523,48 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + snd_printd(SFX "controller runtime suspend\n"); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + + if (chip->irq >= 0) { + free_irq(chip->irq, chip); + chip->irq = -1; + } + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + snd_printd(SFX "controller runtime resume\n"); + + if (azx_acquire_irq(chip, 1) < 0) + return -EIO; + + azx_init_pci(chip); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3238,6 +3291,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3345,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3360,13 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + /* undo 'get' in azx_probe_continue() */ + pm_runtime_put_noidle(&pci->dev); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

3 7

Re: [alsa-devel] Broken powersave in 3.5.2, codec is always active
by Takashi Iwai 20 Aug '12

20 Aug '12

[Re-added Cc to ML; please don't drop it] At Mon, 20 Aug 2012 22:23:02 +0700, ValdikSS wrote: > > Takashi Iwai <tiwai(a)suse.de> писал(а) в своём письме Mon, 20 Aug 2012 > 16:17:04 +0700: > > > At Sat, 18 Aug 2012 12:44:29 +0700, > > ValdikSS wrote: > >> > >> Hello! > >> I have a Lenovo ThinkPad X220 laptop with a conexant codec and recently > >> (after kernel 3.5.2) it is active all the time after KDE login, as > >> powertop says. This couldn't be fixed by logging of from KDE or playing > >> some music in mplayer from console. The only kernel I tried in 3.5.x is > >> 3.5.2, so I can't say is this bug reproducible in 3.5.0 or 3.5.1, but > >> everything is fine with 3.4.9. This never happens if you don't start KDE > >> after reboot, you can listen to some music in mplayer in console and > >> everything seems to be fine. lsof and fuser /dev/snd/* shows nothing. > > > > Just to be sure, check /sys/module/snd_hda_intel/parameters/power_save > > value after logging into KDE. This value can be changed dynamically. > > > > If the value is still set properly, try to build the kernel with the > > tracing option with the patch below. Then try to trace the power > > up/down sequences in the tracing buffer. Some brief description is > > found in Documentation/sound/alsa/HD-Audio.txt Tracepoints section. > > > > > > thanks, > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Add tracepoints at snd_hda_power_up/down > > entrances. > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/hda_codec.c | 2 ++ > > sound/pci/hda/hda_trace.h | 24 ++++++++++++++++++++++++ > > 2 files changed, 26 insertions(+) > > > > diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c > > index 408babc..ad0101e 100644 > > --- a/sound/pci/hda/hda_codec.c > > +++ b/sound/pci/hda/hda_codec.c > > @@ -4417,6 +4417,7 @@ static void __snd_hda_power_up(struct hda_codec > > *codec, bool wait_power_down) > > spin_lock(&codec->power_lock); > > codec->power_count++; > > + trace_hda_power_count(codec); > > /* Return if power_on or transitioning to power_on, unless currently > > * powering down. */ > > if ((codec->power_on || codec->power_transition > 0) && > > @@ -4496,6 +4497,7 @@ void snd_hda_power_down(struct hda_codec *codec) > > { > > spin_lock(&codec->power_lock); > > --codec->power_count; > > + trace_hda_power_count(codec); > > if (!codec->power_on || codec->power_count || codec->power_transition) > > { > > spin_unlock(&codec->power_lock); > > return; > > diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h > > index 9884871..9a34b86 100644 > > --- a/sound/pci/hda/hda_trace.h > > +++ b/sound/pci/hda/hda_trace.h > > @@ -87,6 +87,30 @@ DEFINE_EVENT(hda_power, hda_power_up, > > TP_ARGS(codec) > > ); > > +TRACE_EVENT(hda_power_count, > > + TP_PROTO(struct hda_codec *codec), > > + TP_ARGS(codec), > > + TP_STRUCT__entry( > > + __field( unsigned int, card ) > > + __field( unsigned int, addr ) > > + __field( int, power_count ) > > + __field( int, power_on ) > > + __field( int, power_transition ) > > + ), > > + > > + TP_fast_assign( > > + __entry->card = (codec)->bus->card->number; > > + __entry->addr = (codec)->addr; > > + __entry->power_count = (codec)->power_count; > > + __entry->power_on = (codec)->power_on; > > + __entry->power_transition = (codec)->power_transition; > > + ), > > + > > + TP_printk("[%d:%d] power_count=%d, power_on=%d, power_transition=%d", > > + __entry->card, __entry->addr, __entry->power_count, > > + __entry->power_on, __entry->power_transition) > > +); > > + > > TRACE_EVENT(hda_unsol_event, > > TP_PROTO(struct hda_bus *bus, u32 res, u32 res_ex), > > Thanks for the quick reply! I compiled 3.5.2 kernel with this patch, > enabled hda tracing, ran mplayer once in the console and started KDE. Once > KDE is started, card is used 100% of time, which can be seen in powertop. > When I logout from KDE and kill kdm, no events are generated in trace > since KDE start (no sounds are played from KDE start till KDM kill) and > card is still used 100% of time. If you play any sounds in mplayer in > console after kdm kill, the card is still used 100% time. This can be > fixed by laptop suspend. > > The traces are quite huge so I pasted it to pastebin: > 01-trace-just-enabled http://pastebin.com/JNRWXUPy > 02-mplayer-played-and-stopped-card-is-normal http://pastebin.com/r0jZLDey > 03-kde-started-and-stopped-100-percent-card http://pastebin.com/Z7mzbY4c > The last log begins with latest rows from 02 It's still not clear from the log. You can reduce the trace pattern to hda_power_* for reducing the trace log size. In anyway, could you try the patch below? thanks, Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20..b94e04d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4433,6 +4433,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; }

2 2

[alsa-devel] ASoC updates for 3.6
by Mark Brown 20 Aug '12

20 Aug '12

The following changes since commit 0d7614f09c1ebdbaa1599a5aba7593f147bf96ee: Linux 3.6-rc1 (2012-08-02 16:38:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 28c42c28309244d0b15d1b385e33429d59997679: ASoC: wm9712: Fix inverted capture volume (2012-08-17 22:43:18 +0100) ---------------------------------------------------------------- ASoC: Additional updates for 3.6 A batch more bugfixes, all driver-specific and fairly small and unremarkable in a global context. The biggest batch are for the newly added Arizona drivers. ---------------------------------------------------------------- Chris Rattray (1): ASoC: wm8994: Add missing dapm routes for WM8958 rev A Fabio Estevam (2): ASoC: mxs: Fix the name of the SoC family ASoC: imx-ssi: Remove mono support Mark Brown (10): Merge tag 'asoc-3.6' of git://git.kernel.org/.../broonie/sound into for-3.6 ASoC: wm8962: Don't duplicate bias level management in resume ASoC: core: Upgrade the severity of probe deferral errors to dev_err() MAINTAINERS: Add entries for Wolfson Arizona class devices ASoC: wm5102: Add missing input PGA routes ASoC: wm5110: Add missing input PGA routes ASoC: jack: Always notify full jack status ASoC: wm5102: Remove DRC2 ASoC: wm9712: Fix microphone source selection ASoC: wm9712: Fix inverted capture volume Peter Ujfalusi (1): ASoC: omap-mcbsp: Fix 6pin mux configuration Sachin Kamat (1): ASoC: Samsung: Fix build error Scott Jiang (1): ASoC: bfin: fix memory leak in sport3 controller driver Vaibhav Bedia (1): ASoC: Davinci: McASP: Flush the FIFO before enabling MAINTAINERS | 7 ++++++- sound/soc/blackfin/bf6xx-sport.c | 7 +++++++ sound/soc/codecs/ab8500-codec.c | 4 ++++ sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/mc13783.c | 2 ++ sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/stac9766.c | 1 + sound/soc/codecs/wm5102.c | 25 +++++++++---------------- sound/soc/codecs/wm5110.c | 12 ++++++++++++ sound/soc/codecs/wm8962.c | 18 +++--------------- sound/soc/codecs/wm8994.c | 17 ++++++++++++++++- sound/soc/codecs/wm9712.c | 22 +++++++++++++++++++--- sound/soc/codecs/wm9713.c | 1 + sound/soc/davinci/davinci-mcasp.c | 10 ++++++++-- sound/soc/fsl/imx-ssi.c | 5 +++-- sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ sound/soc/omap/mcbsp.c | 2 +- sound/soc/omap/omap-mcbsp.c | 1 + sound/soc/omap/omap-pcm.c | 1 + sound/soc/samsung/pcm.c | 2 +- sound/soc/soc-core.c | 12 +++++++----- sound/soc/soc-jack.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 10 +++++----- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- 28 files changed, 139 insertions(+), 60 deletions(-)

2 1

[alsa-devel] [PATCH 0/3] ASoC: add compress stream support
by Vinod Koul 20 Aug '12

20 Aug '12

Here is the much delayed support for compress audio streams in ASoC. The v2 incorprates feedback discussed on v1 patches and this also adds the support in Intel drivers. Namarta Kohli (1): ASoC: add compress stream support Vinod Koul (2): ASoC: add definations for compressed operations ASoC: mid-x86 - add support for compressed streams include/sound/compress_driver.h | 1 + include/sound/soc-dai.h | 3 + include/sound/soc.h | 16 ++- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/mid-x86/mfld_machine.c | 9 ++ sound/soc/mid-x86/sst_dsp.h | 134 +++++++++++++++++ sound/soc/mid-x86/sst_platform.c | 204 ++++++++++++++++++++++++++- sound/soc/mid-x86/sst_platform.h | 26 ++++- sound/soc/soc-compress.c | 295 ++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 57 +++++--- 11 files changed, 721 insertions(+), 27 deletions(-) create mode 100644 sound/soc/mid-x86/sst_dsp.h create mode 100644 sound/soc/soc-compress.c

2 4

[alsa-devel] [PATCH 0/10] fix error return code
by Julia Lawall 20 Aug '12

20 Aug '12

These patches fix cases where the return code appears to be unintentially nonnegative. The complete semantic match that finds the problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ f(...) { <+... ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ...+> } @r exists@ identifier ret,l,ok.f; expression e1,e2,e3; statement S; position p1,p2,p3; @@ f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(...) { ... when != ret = e2 when forall return@p3 ret; } ) ... when any } @bad exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any ret = e1 ... when any if@p2(...) S2 @bad2@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; @@ ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when any if@p2(...) S2 @script:python depends on !bad && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

4 13

Re: [alsa-devel] [sound:master 88/92] sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_count'
by Takashi Iwai 20 Aug '12

20 Aug '12

At Mon, 20 Aug 2012 23:51:31 +0800, Fengguang Wu wrote: > > Hi Takashi, > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git master > head: b9687805cf4e70e08221856b1bbadfa5e37c3342 > commit: b244d335609f9bbf4c0204a365bf03b7ac53c3c8 [88/92] ALSA: hda - Add tracepoints at snd_hda_power_up/down entrances. > config: i386-randconfig-k767 (attached as .config) > > All related error/warning messages: > > make[4]: *** [sound/pci/hda/hda_codec.o] Error 1 > make[4]: Target `__build' not remade because of errors. > -- > In file included from include/trace/ftrace.h:567:0, > from include/trace/define_trace.h:86, > from sound/pci/hda/hda_trace.h:141, > from sound/pci/hda/hda_codec.c:41: > sound/pci/hda/./hda_trace.h: In function 'ftrace_raw_event_hda_power_count': > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_count' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_on' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_transition' > In file included from include/trace/ftrace.h:775:0, > from include/trace/define_trace.h:86, > from sound/pci/hda/hda_trace.h:141, > from sound/pci/hda/hda_codec.c:41: > sound/pci/hda/./hda_trace.h: In function 'perf_trace_hda_power_count': > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_count' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_on' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_transition' > > vim +90 sound/pci/hda/./hda_trace.h > 87 TP_ARGS(codec) > 88 ); > 89 > > 90 TRACE_EVENT(hda_power_count, > 91 TP_PROTO(struct hda_codec *codec), > 92 TP_ARGS(codec), > 93 TP_STRUCT__entry( Thanks, looks like the missing ifdef. Fixed now. Takashi

1 0

[alsa-devel] Lockups reading from i.MX51 SSI registers
by Mark Brown 20 Aug '12

20 Aug '12

Hi, I'm trying to use -next to test AC'97 register I/O on an i.MX51 board but I'm seeing the CPU hang during probe at: lreg = (reg & 0x7f) << 12 ; writel(lreg, base + SSI_SACADD); in imx_ssi_ac97_read(). I'm somewhat suspicious this might be because the IP block isn't clocked properly, I do notice the recent conversion to the clock API which looks rather involved but it's possible something else broke. Does anyone have any bright ideas what might be going on here? The board doesn't have the reste functions defined so this is the first interaction with the hardware block AFAICT. Thanks, Mark

5 11

[alsa-devel] [alsa-plugins PATCH] pulse: Support 24-bit formats
by Andrew Eikum 20 Aug '12

20 Aug '12

These formats were introduced in PulseAudio 0.9.15. There aren't any new formats supported since then, so this also brings the PulseAudio plugin in sync with current PulseAudio. Signed-off-by: Andrew Eikum <aeikum(a)codeweavers.com> --- There was a problem once before with a patch I submitted. This is using format-patch and imap-send, so hopefully it goes through better than it did before :) pulse/pcm_pulse.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+)

2 4

[alsa-devel] [PATCH 0/2] Two fixes for ca0132
by David Henningsson 20 Aug '12

20 Aug '12

I came across the following codec: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1038651/+attachment/32… ...and the codec wouldn't load due to mixer errors. I don't know more about the codec than the codec proc shows, and looking at that file it seems a bit broken, but maybe Ian would be able to make it work, with the help of some secret verbs? At least now the codec would load correctly, so that the output side could be used. David Henningsson (2): ALSA: hda - Fix "Line" and "Mic" names for CA0132 ALSA: hda - don't create dysfunctional mixer controls for ca0132 sound/pci/hda/patch_ca0132.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) -- 1.7.9.5

2 6

[alsa-devel] Broken powersave in 3.5.2, codec is always active
by ValdikSS 20 Aug '12

20 Aug '12

Hello! I have a Lenovo ThinkPad X220 laptop with a conexant codec and recently (after kernel 3.5.2) it is active all the time after KDE login, as powertop says. This couldn't be fixed by logging of from KDE or playing some music in mplayer from console. The only kernel I tried in 3.5.x is 3.5.2, so I can't say is this bug reproducible in 3.5.0 or 3.5.1, but everything is fine with 3.4.9. This never happens if you don't start KDE after reboot, you can listen to some music in mplayer in console and everything seems to be fine. lsof and fuser /dev/snd/* shows nothing. Thank you! $ cat /etc/modprobe.d/options.conf options snd-hda-intel power_save=1 power_save_controller=1 $ lspci -nn | grep Audio 00:1b.0 Audio device [0403]: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller [8086:1c20] (rev 04) $ lsmod | grep snd snd_hda_codec_hdmi 23704 1 snd_hda_codec_conexant 47158 1 snd_hda_intel 25700 2 snd_hda_codec 97400 3 snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_hda_intel snd_hwdep 6300 1 snd_hda_codec snd_pcm 74926 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 7185 2 snd_pcm,snd_hda_intel snd_timer 18966 1 snd_pcm snd 60021 12 snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_pcm,snd_hda_codec,snd_hda_intel,thinkpad_acpi soundcore 5442 1 snd

2 1

[alsa-devel] [PATCH 1/2] snd-ad1816a: remove useless struct snd_card_ad1816a
by Ondrej Zary 20 Aug '12

20 Aug '12

struct snd_card_ad1816a is only set but the values are never used then. Removing it allows struct snd_card's private_data to be used for struct snd_ad1816a, simplifying the code. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index d010858..62da41e 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -165,7 +165,7 @@ struct snd_ad1816a { extern int snd_ad1816a_create(struct snd_card *card, unsigned long port, int irq, int dma1, int dma2, - struct snd_ad1816a **chip); + struct snd_ad1816a *chip); extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm); extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 94b83b6..1a374e6 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -63,11 +63,6 @@ MODULE_PARM_DESC(enable, "Enable ad1816a based soundcard."); module_param_array(clockfreq, int, NULL, 0444); MODULE_PARM_DESC(clockfreq, "Clock frequency for ad1816a driver (default = 0)."); -struct snd_card_ad1816a { - struct pnp_dev *dev; - struct pnp_dev *devmpu; -}; - static struct pnp_card_device_id snd_ad1816a_pnpids[] = { /* Analog Devices AD1815 */ { .id = "ADS7150", .devs = { { .id = "ADS7150" }, { .id = "ADS7151" } } }, @@ -99,25 +94,16 @@ MODULE_DEVICE_TABLE(pnp_card, snd_ad1816a_pnpids); #define DRIVER_NAME "snd-card-ad1816a" -static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acard, - struct pnp_card_link *card, +static int __devinit snd_card_ad1816a_pnp(int dev, struct pnp_card_link *card, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; int err; - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) + pdev = pnp_request_card_device(card, id->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devmpu == NULL) { - mpu_port[dev] = -1; - snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n"); - } - - pdev = acard->dev; - err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "AUDIO PnP configure failure\n"); @@ -130,16 +116,17 @@ static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acar dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - if (acard->devmpu == NULL) + pdev = pnp_request_card_device(card, id->devs[1].id, NULL); + if (pdev == NULL) { + mpu_port[dev] = -1; + snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n"); return 0; - - pdev = acard->devmpu; + } err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "MPU401 PnP configure failure\n"); mpu_port[dev] = -1; - acard->devmpu = NULL; } else { mpu_port[dev] = pnp_port_start(pdev, 0); mpu_irq[dev] = pnp_irq(pdev, 0); @@ -153,18 +140,17 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard { int error; struct snd_card *card; - struct snd_card_ad1816a *acard; struct snd_ad1816a *chip; struct snd_opl3 *opl3; struct snd_timer *timer; error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_ad1816a), &card); + sizeof(struct snd_ad1816a), &card); if (error < 0) return error; - acard = card->private_data; + chip = card->private_data; - if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) { + if ((error = snd_card_ad1816a_pnp(dev, pcard, pid))) { snd_card_free(card); return error; } @@ -174,7 +160,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard irq[dev], dma1[dev], dma2[dev], - &chip)) < 0) { + chip)) < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 177eed3..de5cc1c 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -548,7 +548,6 @@ static int snd_ad1816a_free(struct snd_ad1816a *chip) snd_dma_disable(chip->dma2); free_dma(chip->dma2); } - kfree(chip); return 0; } @@ -573,19 +572,13 @@ static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip) int __devinit snd_ad1816a_create(struct snd_card *card, unsigned long port, int irq, int dma1, int dma2, - struct snd_ad1816a **rchip) + struct snd_ad1816a *chip) { static struct snd_device_ops ops = { .dev_free = snd_ad1816a_dev_free, }; int error; - struct snd_ad1816a *chip; - *rchip = NULL; - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; @@ -631,7 +624,6 @@ int __devinit snd_ad1816a_create(struct snd_card *card, return error; } - *rchip = chip; return 0; } -- Ondrej Zary

2 1

[alsa-devel] [PATCH] sound: fix pcm.h kernel-doc warning and notation
by Randy Dunlap 20 Aug '12

20 Aug '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix kernel-doc warning in <sound/pcm.h> and add function name to make the kernel-doc notation complete. Warning(include/sound/pcm.h:1081): No description found for parameter 'substream' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org (moderated for non-subscribers) --- include/sound/pcm.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) --- lnx-36-rc2.orig/include/sound/pcm.h +++ lnx-36-rc2/include/sound/pcm.h @@ -1075,7 +1075,8 @@ static inline void snd_pcm_limit_isa_dma const char *snd_pcm_format_name(snd_pcm_format_t format); /** - * Get a string naming the direction of a stream + * snd_pcm_stream_str - Get a string naming the direction of a stream + * @substream: the pcm substream instance */ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) {

2 1

[alsa-devel] [patch] oss/sb_audio: prevent divide by zero bug
by Dan Carpenter 20 Aug '12

20 Aug '12

Speed comes from get_user() in audio_ioctl(). We use it to set the "s" variable before clamping it to valid values so it could lead to a divide by zero bug. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 733b014..b2b3c01 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed) if (speed > 0) { int tmp; - int s = speed * devc->channels; + int s; if (speed < 5000) speed = 5000; if (speed > 44100) speed = 44100; + s = speed * devc->channels; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst;

2 1

Re: [alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Takashi Iwai 20 Aug '12

20 Aug '12

At Mon, 20 Aug 2012 14:57:51 +0900 (JST), etsuko suzuki wrote: > > Hello, Takashi. > > > Hm, I guess your message didn't reach properly, or was filtered by some spam > filter. > > Better to contact with Jaroslav (Cc'ed), who maintains alsa-devel ML. > > I contacted him in August 17. > However, there is no reply yet. Didn't you read his mail in Aug. 16? You were in Cc. > > [No.1] > > > > http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx > > > > > > In this, I found the statement 'left-alinged'. > > > (In 'Details about wValidBitsPerSample') > > > > > > I understood 'left-alinged' was 'LSB padding'. > > > So I thought S24_LE was LSB padding. > > > > No. S24_LE has the MSB padding of 8 bits in 32 bit container. > > That is, does it mean that this document is not correct? > (I don't understand why this document differs from what you say.) Well, Microsoft doesn't conquer the whole world yet. Their definition isn't what we define for our purpose. > http://www-mmsp.ece.mcgill.ca/documents/audioformats/wave/Docs/multichaudP.… > (This PDF is the same as what was written before. > The page is written on this.) > > On the 6 page of this PDF, it is written to like this. > > 'If wValidBitsPerSample is less than wBitsPerSample, > then the actual > PCM data is "left-aligned" within the container. > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > The sample itself is justified most significant; > all extra bits are at the least-significant > portion of the container. > All non-valid data bits must be set to 0.' > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > In S24_LE, I think 'wBitsPerSample' indicates > 'the container size'(=32) and > 'wValidBitsPerSample' indicates 'the bit of actual valid data'(=24). You _think_ but you utterly misunderstand the terms. ALSA S24 is not what you think. S24 is a special format, 3 bytes LSB in 4 bytes container. This _is_ the definition. There is no such a format in MS's WAV definition. wValidBitsPerSample is rather corresponding to hw_params ms_bits parameter. > > [No.2] > > It was written that Microsoft has defined > the new structure 'WAVE_FORMAT_EXTENSIBLE'(FORMAT_CODE:0xFFFE). > (On the 4 page of this PDF.) > > I think that the new structure should be used > in this(the container size > the bit of actual valid data) case. > > But the structure recording > by arecord was 'WAVE_FORMAT_PCM'(FORMAT_CODE:0x0001). > > > % od -t x4 ubuntu_rec_S24_LE_48000.wav > > 0000000 46464952 003a9824 45564157 20746d66 > > 0000010 00000010 00020001 0000bb80 0005dc00 > ^^^^ 0x0001:WAVE_FORMAT_PCM > > (The 'FORMAT_CODE' is written in this page. > http://www-mmsp.ece.mcgill.ca/documents/audioformats/wave/wave.html) > > > Why does ALSA NOT use the new structure? Because it's new, and no one cared such a thing. > Has ALSA corresponded to the new structure? Please send a fix patch if you'd like to improve :) thanks, Takashi

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[alsa-devel] [PATCH 2/2] snd-ad1816a: Implement suspend/resume
by Ondrej Zary 19 Aug '12

19 Aug '12

Implement suspend/resume support for AD1816 chips. Tested with Terratec SoundSystem Base-1. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index 62da41e..2a89f0d 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -147,6 +147,9 @@ struct snd_ad1816a { unsigned int c_dma_size; struct snd_timer *timer; +#ifdef CONFIG_PM + unsigned short image[48]; +#endif }; @@ -171,5 +174,9 @@ extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer); +#ifdef CONFIG_PM +extern void snd_ad1816a_suspend(struct snd_ad1816a *chip); +extern void snd_ad1816a_resume(struct snd_ad1816a *chip); +#endif #endif /* __SOUND_AD1816A_H */ diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 1a374e6..2c2f829 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -244,13 +244,37 @@ static void __devexit snd_ad1816a_pnp_remove(struct pnp_card_link * pcard) pnp_set_card_drvdata(pcard, NULL); } +#ifdef CONFIG_PM +static int snd_ad1816a_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_ad1816a_suspend(card->private_data); + return 0; +} + +static int snd_ad1816a_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_ad1816a_resume(card->private_data); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + static struct pnp_card_driver ad1816a_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, .name = "ad1816a", .id_table = snd_ad1816a_pnpids, .probe = snd_ad1816a_pnp_detect, .remove = __devexit_p(snd_ad1816a_pnp_remove), - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_ad1816a_pnp_suspend, + .resume = snd_ad1816a_pnp_resume, +#endif }; static int __init alsa_card_ad1816a_init(void) diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index de5cc1c..db64df6 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -491,7 +491,7 @@ static int snd_ad1816a_capture_close(struct snd_pcm_substream *substream) } -static void __devinit snd_ad1816a_init(struct snd_ad1816a *chip) +static void snd_ad1816a_init(struct snd_ad1816a *chip) { unsigned long flags; @@ -511,6 +511,32 @@ static void __devinit snd_ad1816a_init(struct snd_ad1816a *chip) spin_unlock_irqrestore(&chip->lock, flags); } +#ifdef CONFIG_PM +void snd_ad1816a_suspend(struct snd_ad1816a *chip) +{ + int reg; + unsigned long flags; + + snd_pcm_suspend_all(chip->pcm); + spin_lock_irqsave(&chip->lock, flags); + for (reg = 0; reg < 48; reg++) + chip->image[reg] = snd_ad1816a_read(chip, reg); + spin_unlock_irqrestore(&chip->lock, flags); +} + +void snd_ad1816a_resume(struct snd_ad1816a *chip) +{ + int reg; + unsigned long flags; + + snd_ad1816a_init(chip); + spin_lock_irqsave(&chip->lock, flags); + for (reg = 0; reg < 48; reg++) + snd_ad1816a_write(chip, reg, chip->image[reg]); + spin_unlock_irqrestore(&chip->lock, flags); +} +#endif + static int __devinit snd_ad1816a_probe(struct snd_ad1816a *chip) { unsigned long flags; -- Ondrej Zary

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Re: [alsa-devel] M-Audio FastTrack C400
by Grant Diffey 18 Aug '12

18 Aug '12

So this looks reasonably familiar, int0 is DFU int1 is mixer int2 is audio out int3 is audio in int4 is midi int5 is HID Quirk should look like the fast track ultra basically. maybe the mixer is less broken in this device? Should we start with just QUIRK_STANDARD_AUDIO_INTERFACE or copy the whole data section from the Ultra quirk? This has isosyncronous feedback for clock control needs current sound tree usb-audio tree Grant On Wed, Apr 18, 2012 at 8:45 AM, Chris Cavey <chris-alsa(a)rauros.net> wrote: > Will to help with development, may need pointers to get started and > willing to test things. I am mostly concerned with simple playback. > Recording, DSP, and Midi can all come later if at all. > > As requested the output of sudo lsusb -v -d 0763:2030" > > Bus 002 Device 003: ID 0763:2030 Midiman > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > idVendor 0x0763 Midiman > idProduct 0x2030 > bcdDevice 1.01 > iManufacturer 1 M-Audio > iProduct 2 Fast Track C400 > iSerial 0 > bNumConfigurations 1 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 455 > bNumInterfaces 6 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0x80 > (Bus Powered) > MaxPower 500mA > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 254 Application Specific Interface > bInterfaceSubClass 1 Device Firmware Update > bInterfaceProtocol 0 > iInterface 6 Fast Track C400 DFU > Device Firmware Upgrade Interface Descriptor: > bLength 9 > bDescriptorType 33 > bmAttributes 1 > Will Not Detach > Manifestation Intolerant > Upload Unsupported > Download Supported > wDetachTimeout 2000 milliseconds > wTransferSize 1024 bytes > bcdDFUVersion 1.10 > Interface Association: > bLength 8 > bDescriptorType 11 > bFirstInterface 1 > bInterfaceCount 4 > bFunctionClass 255 Vendor Specific Class > bFunctionSubClass 0 > bFunctionProtocol 32 > iFunction 2 Fast Track C400 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 1 > bInterfaceProtocol 32 > iInterface 2 Fast Track C400 > ** UNRECOGNIZED: 09 24 01 00 02 0a a3 00 00 > ** UNRECOGNIZED: 09 24 0b 80 02 81 82 03 00 > ** UNRECOGNIZED: 08 24 0a 81 03 03 03 00 > ** UNRECOGNIZED: 08 24 0a 82 00 05 03 00 > ** UNRECOGNIZED: 11 24 02 00 01 01 00 80 06 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 01 01 01 00 02 80 00 00 00 > ** UNRECOGNIZED: 11 24 02 02 03 06 00 80 04 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 03 03 06 00 40 80 00 00 00 > ** UNRECOGNIZED: 11 24 09 20 00 00 02 02 00 04 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 15 24 04 40 03 02 00 43 04 00 00 00 00 00 ff ff ff > ff 0f 00 00 > ** UNRECOGNIZED: 10 24 04 42 02 02 00 01 00 00 00 00 00 ff 00 00 > ** UNRECOGNIZED: 10 24 07 43 02 00 42 fd 03 00 00 fd 03 00 00 00 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 06 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x00ea 1x 234 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 2 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 d0 00 00 00 02 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 03 02 10 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x004e 1x 78 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 04 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x81 EP 1 IN > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x009c 1x 156 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 4 > bAlternateSetting 0 > bNumEndpoints 2 > bInterfaceClass 1 Audio > bInterfaceSubClass 3 MIDI Streaming > bInterfaceProtocol 0 > iInterface 0 > MIDIStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 35 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 2 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 1 Embedded > bJackID 3 > bNrInputPins 1 > baSourceID( 0) 2 > BaSourcePin( 0) 1 > iJack 8 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 2 External > bJackID 4 > bNrInputPins 1 > baSourceID( 0) 1 > BaSourcePin( 0) 1 > iJack 105 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x07 EP 7 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 1 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x87 EP 7 IN > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 3 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 5 > bAlternateSetting 0 > bNumEndpoints 1 > bInterfaceClass 3 Human Interface Device > bInterfaceSubClass 0 No Subclass > bInterfaceProtocol 1 Keyboard > iInterface 0 > HID Device Descriptor: > bLength 9 > bDescriptorType 33 > bcdHID 1.10 > bCountryCode 0 Not supported > bNumDescriptors 1 > bDescriptorType 34 Report > wDescriptorLength 63 > Report Descriptors: > ** UNAVAILABLE ** > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x86 EP 6 IN > bmAttributes 3 > Transfer Type Interrupt > Synch Type None > Usage Type Data > wMaxPacketSize 0x0008 1x 8 bytes > bInterval 8 > Device Qualifier (for other device speed): > bLength 10 > bDescriptorType 6 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > bNumConfigurations 1 > Device Status: 0x0000 > (Bus Powered) > > > On 04/17/2012 09:15 AM, Grant Diffey wrote: > > start with an lsusb -v -d (vid:pid) pasted here > > On Tue, Apr 17, 2012 at 10:37 PM, Chris Cavey <chris-alsa(a)rauros.net>wrote: > >> I'd love to be able to help ALSA support this USB sound device. I've >> currently got the device working in a VM on my Ubuntu 12.04 workstation. >> I believe I can capture USB packets. Would this be helpful to anyone? >> Any pointers on where to get started? >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > > >

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[alsa-devel] [PATCH 1/2 v2] ASoC: bf5xx-ad1836: convert to use snd_soc_register_card
by Scott Jiang 18 Aug '12

18 Aug '12

Cpu dai and codec name are passed in through platform data. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 73 +++++++++++++++++++------------------ 1 files changed, 37 insertions(+), 36 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d542d40..16b9c9e 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -59,62 +59,63 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { #define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) -static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.0", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.1", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .codec_dai_name = "ad1836-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }; static struct snd_soc_card bf5xx_ad1836 = { .name = "bfin-ad1836", .owner = THIS_MODULE, - .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM], + .dai_link = &bf5xx_ad1836_dai, .num_links = 1, }; -static struct platform_device *bfxx_ad1836_snd_device; - -static int __init bf5xx_ad1836_init(void) +static __devinit int bf5xx_ad1836_driver_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &bf5xx_ad1836; + const char **link_name; int ret; - bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1836_snd_device) - return -ENOMEM; + link_name = pdev->dev.platform_data; + if (!link_name) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + bf5xx_ad1836_dai.cpu_dai_name = link_name[0]; + bf5xx_ad1836_dai.codec_name = link_name[1]; - platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836); - ret = platform_device_add(bfxx_ad1836_snd_device); + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + ret = snd_soc_register_card(card); if (ret) - platform_device_put(bfxx_ad1836_snd_device); - + dev_err(&pdev->dev, "Failed to register card\n"); return ret; } -static void __exit bf5xx_ad1836_exit(void) +static int __devexit bf5xx_ad1836_driver_remove(struct platform_device *pdev) { - platform_device_unregister(bfxx_ad1836_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(bf5xx_ad1836_init); -module_exit(bf5xx_ad1836_exit); +static struct platform_driver bf5xx_ad1836_driver = { + .driver = { + .name = "bfin-snd-ad1836", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bf5xx_ad1836_driver_probe, + .remove = __devexit_p(bf5xx_ad1836_driver_remove), +}; +module_platform_driver(bf5xx_ad1836_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); -- 1.7.0.4

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Re: [alsa-devel] [PATCH 1/5] ARM: tegra: config: enable dmaengine based APB DMA driver
by Stephen Warren 17 Aug '12

17 Aug '12

On 08/17/2012 12:38 AM, Laxman Dewangan wrote: > On Thursday 16 August 2012 11:23 PM, Stephen Warren wrote: >> On 08/16/2012 08:13 AM, Laxman Dewangan wrote: >>> Enable config for dmaengine based Tegra APB DMA driver and >>> disable the legacy APB DMA driver (SYSTEM_DMA). >> Laxman, if I apply this series to next-20120816 (plus a few patches in >> my local work branch plus the CPU hotplug patches from Joseph, although >> I suspect none of that matters), then audio playback on Tegra20 is >> broken; the pitch is far too high. Audio playback on Tegra30 works as >> expected. > > I run the test again on ventana and did not see any issue. > I play one song and saw similar behavior with/without this series. OK, I'll test some other boards. I tested Whistler, although all the clocking logic is identical between all Tegra20 boards. > With this series: > -------------------- > > ubuntu@tegra-ubuntu:/home$ aplay "1-04 Hungry Like The Wolf.wav" > Playing WAVE '1-04 Hungry Like The Wolf.wav' : Signed 16 bit Little > Endian, Rate > 44100 Hz, Stereo > underrun!!! (at least 5039.388 ms long) Why are you getting all these underruns? I almost never see this, except when something else is hammering the SD card containing the .wav file. If I saw those, I'd count it as a test failure.

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