December 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 1/7 v2] ARM: OMAP1: ams-delta: register latch dependent devices later
by Janusz Krzysztofik 22 Dec '11

22 Dec '11

In preparation to converting Amstrad Delta on-board latches to basic_mmio_gpio devices, registration of platform devices which depend on latches and will require initialization of their GPIO pins first, should be moved out of .machine_init down to late_initcall level, as the gpio-generic driver is not available until device_initcall time. The latch reset operation, which will be replaced with GPIO initialization, must also be moved to late_initcall for the same reason. Since there was already another, separate arch_initcall function for setting up one of those latch dependent devices, the on-board modem device, reuse that function, i.e., rename it to a name that matches the new purpose, extend with other device setup relocated from .machine_init, and move down to the late_initcall level. While being at it, add missing gpio_free() in case the modem platform device registration fails. Thanks to Tony Lindgren <tony(a)atomide.com> who suggested this approach instead of shifting up the gpio-generic driver initialization. In addition, defer registration of the Amstrad Delta ASoC and serio devices, done from their device driver files, until late_initcall time, as those drivers will depend on their GPIO pins already requested from the board late_init() function until updated to register their GPIO pins themselves. Created and tested against linux-3.2-rc6. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes since the initial version of this patch: * use late_initcall() unconditionally in modules, requested by Russell King (thanks!); an updated patch 7/7 will follow, * in the changelog message, move credits to Tony up a bit, since the person responsible for other changes (he might not necessarily like) is me, not him ;). Comments copied from this patch initial submission: This patch was not present in the initial submission, it replaces the old patch 1/10, providing an alternative solution not touching the gpio-generic driver. arch/arm/mach-omap1/board-ams-delta.c | 28 ++++++++++++++++++++-------- drivers/input/serio/ams_delta_serio.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- 3 files changed, 22 insertions(+), 10 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index b0f15d2..50987c9 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -275,11 +275,14 @@ static struct omap1_cam_platform_data ams_delta_camera_platform_data = { }; static struct platform_device *ams_delta_devices[] __initdata = { - &ams_delta_nand_device, &ams_delta_kp_device, + &ams_delta_camera_device, +}; + +static struct platform_device *late_devices[] __initdata = { + &ams_delta_nand_device, &ams_delta_lcd_device, &ams_delta_led_device, - &ams_delta_camera_device, }; static void __init ams_delta_init(void) @@ -307,9 +310,6 @@ static void __init ams_delta_init(void) omap_serial_init(); omap_register_i2c_bus(1, 100, NULL, 0); - /* Clear latch2 (NAND, LCD, modem enable) */ - ams_delta_latch2_write(~0, 0); - omap1_usb_init(&ams_delta_usb_config); omap1_set_camera_info(&ams_delta_camera_platform_data); #ifdef CONFIG_LEDS_TRIGGERS @@ -345,13 +345,18 @@ static struct platform_device ams_delta_modem_device = { }, }; -static int __init ams_delta_modem_init(void) +static int __init late_init(void) { int err; if (!machine_is_ams_delta()) return -ENODEV; + /* Clear latch2 (NAND, LCD, modem enable) */ + ams_delta_latch2_write(~0, 0); + + platform_add_devices(late_devices, ARRAY_SIZE(late_devices)); + omap_cfg_reg(M14_1510_GPIO2); ams_delta_modem_ports[0].irq = gpio_to_irq(AMS_DELTA_GPIO_PIN_MODEM_IRQ); @@ -367,9 +372,16 @@ static int __init ams_delta_modem_init(void) AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC, AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC); - return platform_device_register(&ams_delta_modem_device); + err = platform_device_register(&ams_delta_modem_device); + if (err) + goto gpio_free; + return 0; + +gpio_free: + gpio_free(AMS_DELTA_GPIO_PIN_MODEM_IRQ); + return err; } -arch_initcall(ams_delta_modem_init); +late_initcall(late_init); static void __init ams_delta_map_io(void) { diff --git a/drivers/input/serio/ams_delta_serio.c b/drivers/input/serio/ams_delta_serio.c index d4d08bd..835d37a 100644 --- a/drivers/input/serio/ams_delta_serio.c +++ b/drivers/input/serio/ams_delta_serio.c @@ -165,7 +165,7 @@ serio: kfree(ams_delta_serio); return err; } -module_init(ams_delta_serio_init); +late_initcall(ams_delta_serio_init); static void __exit ams_delta_serio_exit(void) { diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index ccb8a6a..be81bc7 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -636,7 +636,7 @@ err: platform_device_put(ams_delta_audio_platform_device); return ret; } -module_init(ams_delta_module_init); +late_initcall(ams_delta_module_init); static void __exit ams_delta_module_exit(void) { -- 1.7.3.4

4 6

[alsa-devel] [PATCH] ASoC: samsung: Add .owner to struct snd_soc_card
by Axel Lin 22 Dec '11

22 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/goni_wm8994.c | 1 + sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/jive_wm8750.c | 1 + sound/soc/samsung/littlemill.c | 1 + sound/soc/samsung/ln2440sbc_alc650.c | 1 + sound/soc/samsung/lowland.c | 1 + sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 1 + sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 1 + sound/soc/samsung/s3c24xx_uda134x.c | 1 + sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk2443_wm9710.c | 1 + sound/soc/samsung/smdk_spdif.c | 1 + sound/soc/samsung/smdk_wm8580.c | 1 + sound/soc/samsung/smdk_wm8580pcm.c | 1 + sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/smdk_wm8994pcm.c | 1 + sound/soc/samsung/smdk_wm9713.c | 1 + sound/soc/samsung/speyside.c | 1 + sound/soc/samsung/tobermory.c | 1 + 21 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 84f9c3c..c23c2ae 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -244,6 +244,7 @@ static struct snd_soc_dai_link goni_dai[] = { static struct snd_soc_card goni = { .name = "goni", + .owner = THIS_MODULE, .dai_link = goni_dai, .num_links = ARRAY_SIZE(goni_dai), diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 03cfa5f..6e32577 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -215,6 +215,7 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { static struct snd_soc_card h1940_asoc = { .name = "h1940", + .owner = THIS_MODULE, .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 8e523fd..1578663 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -127,6 +127,7 @@ static struct snd_soc_dai_link jive_dai = { /* jive audio machine driver */ static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", + .owner = THIS_MODULE, .dai_link = &jive_dai, .num_links = 1, diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 5cea59b..9dd818b 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -194,6 +194,7 @@ static int littlemill_late_probe(struct snd_soc_card *card) static struct snd_soc_card littlemill = { .name = "Littlemill", + .owner = THIS_MODULE, .dai_link = littlemill_dai, .num_links = ARRAY_SIZE(littlemill_dai), diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c index cde38b8..69c4a59 100644 --- a/sound/soc/samsung/ln2440sbc_alc650.c +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -34,6 +34,7 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { static struct snd_soc_card ln2440sbc = { .name = "LN2440SBC", + .owner = THIS_MODULE, .dai_link = ln2440sbc_dai, .num_links = ARRAY_SIZE(ln2440sbc_dai), }; diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 4216a06..4adff93 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -177,6 +177,7 @@ static struct snd_soc_dapm_route audio_paths[] = { static struct snd_soc_card lowland = { .name = "Lowland", + .owner = THIS_MODULE, .dai_link = lowland_dai, .num_links = ARRAY_SIZE(lowland_dai), .aux_dev = lowland_aux_dev, diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7207189..7ac0ba2 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -465,6 +465,7 @@ static const struct gpio neo1973_gta02_gpios[] = {}; static struct snd_soc_card neo1973 = { .name = "neo1973", + .owner = THIS_MODULE, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), .aux_dev = neo1973_aux_devs, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 71b4c02..21e1236 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -114,6 +114,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_card rx1950_asoc = { .name = "rx1950", + .owner = THIS_MODULE, .dai_link = rx1950_uda1380_dai, .num_links = ARRAY_SIZE(rx1950_uda1380_dai), diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 5027981..7ace6a8 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -89,6 +89,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = { /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic33 = { .name = "Simtec-Hermes", + .owner = THIS_MODULE, .dai_link = &simtec_dai_aic33, .num_links = 1, diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 89b57b5..c42d5f0 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -78,6 +78,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = { /* simtec audio machine driver */ static struct snd_soc_card snd_soc_machine_simtec_aic23 = { .name = "Simtec", + .owner = THIS_MODULE, .dai_link = &simtec_dai_aic23, .num_links = 1, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 62b69fb..d731042 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -229,6 +229,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { static struct snd_soc_card snd_soc_s3c24xx_uda134x = { .name = "S3C24XX_UDA134X", + .owner = THIS_MODULE, .dai_link = &s3c24xx_uda134x_dai_link, .num_links = 1, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index a22fc44..f2dcb42 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -198,6 +198,7 @@ static struct snd_soc_dai_link smartq_dai[] = { static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", + .owner = THIS_MODULE, .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 8bd1dc5..720ba29 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -30,6 +30,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { static struct snd_soc_card smdk2443 = { .name = "SMDK2443", + .owner = THIS_MODULE, .dai_link = smdk2443_dai, .num_links = ARRAY_SIZE(smdk2443_dai), }; diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index e0fd8ad..beaa9c1 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -160,6 +160,7 @@ static struct snd_soc_dai_link smdk_dai = { static struct snd_soc_card smdk = { .name = "SMDK-S/PDIF", + .owner = THIS_MODULE, .dai_link = &smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 81b4478..bff8758 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -203,6 +203,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk = { .name = "SMDK-I2S", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 2, diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c index 49dfafb..fab5322 100644 --- a/sound/soc/samsung/smdk_wm8580pcm.c +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -143,6 +143,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk_pcm = { .name = "SMDK-PCM", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 2, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index ad9ac42..8eb309f 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -144,6 +144,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk = { .name = "SMDK-I2S", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = ARRAY_SIZE(smdk_dai), }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 23c7fb7..77ecba9 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -124,6 +124,7 @@ static struct snd_soc_dai_link smdk_dai[] = { static struct snd_soc_card smdk_pcm = { .name = "SMDK-PCM", + .owner = THIS_MODULE, .dai_link = smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index 31c6daf..8e26a73 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -50,6 +50,7 @@ static struct snd_soc_dai_link smdk_dai = { static struct snd_soc_card smdk = { .name = "SMDK WM9713", + .owner = THIS_MODULE, .dai_link = &smdk_dai, .num_links = 1, }; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 0222d86..f9ab770 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -291,6 +291,7 @@ static struct snd_soc_dapm_route audio_paths[] = { static struct snd_soc_card speyside = { .name = "Speyside", + .owner = THIS_MODULE, .dai_link = speyside_dai, .num_links = ARRAY_SIZE(speyside_dai), .aux_dev = speyside_aux_dev, diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 6f91c65..9199649 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -196,6 +196,7 @@ static int tobermory_late_probe(struct snd_soc_card *card) static struct snd_soc_card tobermory = { .name = "Tobermory", + .owner = THIS_MODULE, .dai_link = tobermory_dai, .num_links = ARRAY_SIZE(tobermory_dai), -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: pxa: Add .owner to struct snd_soc_card
by Axel Lin 22 Dec '11

22 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 1 + sound/soc/pxa/e750_wm9705.c | 1 + sound/soc/pxa/e800_wm9712.c | 1 + sound/soc/pxa/em-x270.c | 1 + sound/soc/pxa/hx4700.c | 1 + sound/soc/pxa/imote2.c | 1 + sound/soc/pxa/magician.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/raumfeld.c | 2 ++ sound/soc/pxa/saarb.c | 1 + sound/soc/pxa/spitz.c | 1 + sound/soc/pxa/tavorevb3.c | 1 + sound/soc/pxa/tosa.c | 1 + sound/soc/pxa/z2.c | 1 + sound/soc/pxa/zylonite.c | 1 + 17 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b0e2fb7..5e5004a 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -317,6 +317,7 @@ static struct snd_soc_dai_link corgi_dai = { /* corgi audio machine driver */ static struct snd_soc_card snd_soc_corgi = { .name = "Corgi", + .owner = THIS_MODULE, .dai_link = &corgi_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 203ab78a..7b1bc2390 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -133,6 +133,7 @@ static struct snd_soc_dai_link e740_dai[] = { static struct snd_soc_card e740 = { .name = "Toshiba e740", + .owner = THIS_MODULE, .dai_link = e740_dai, .num_links = ARRAY_SIZE(e740_dai), }; diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 27f90cc..47b89d7 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -116,6 +116,7 @@ static struct snd_soc_dai_link e750_dai[] = { static struct snd_soc_card e750 = { .name = "Toshiba e750", + .owner = THIS_MODULE, .dai_link = e750_dai, .num_links = ARRAY_SIZE(e750_dai), }; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 858bf94..ea9707e 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -106,6 +106,7 @@ static struct snd_soc_dai_link e800_dai[] = { static struct snd_soc_card e800 = { .name = "Toshiba e800", + .owner = THIS_MODULE, .dai_link = e800_dai, .num_links = ARRAY_SIZE(e800_dai), }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index b13a425..64743a0 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -54,6 +54,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { static struct snd_soc_card em_x270 = { .name = "EM-X270", + .owner = THIS_MODULE, .dai_link = em_x270_dai, .num_links = ARRAY_SIZE(em_x270_dai), }; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 8260207..2a342c9 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -169,6 +169,7 @@ static struct snd_soc_dai_link hx4700_dai = { /* hx4700 audio machine driver */ static struct snd_soc_card snd_soc_card_hx4700 = { .name = "iPAQ hx4700", + .owner = THIS_MODULE, .dai_link = &hx4700_dai, .num_links = 1, .dapm_widgets = hx4700_dapm_widgets, diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index dc905ae..b93dafd 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -60,6 +60,7 @@ static struct snd_soc_dai_link imote2_dai = { static struct snd_soc_card imote2 = { .name = "Imote2", + .owner = THIS_MODULE, .dai_link = &imote2_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index e79f516..3f7a8ec 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -452,6 +452,7 @@ static struct snd_soc_dai_link magician_dai[] = { /* magician audio machine driver */ static struct snd_soc_card snd_soc_card_magician = { .name = "Magician", + .owner = THIS_MODULE, .dai_link = magician_dai, .num_links = ARRAY_SIZE(magician_dai), diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 0e73a7f..9c585af 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -181,6 +181,7 @@ static struct snd_soc_dai_link mioa701_dai[] = { static struct snd_soc_card mioa701 = { .name = "MioA701", + .owner = THIS_MODULE, .dai_link = mioa701_dai, .num_links = ARRAY_SIZE(mioa701_dai), }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index f313eca..db24bc6 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -146,6 +146,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { static struct snd_soc_card palm27x_asoc = { .name = "Palm/PXA27x", + .owner = THIS_MODULE, .dai_link = palm27x_dai, .num_links = ARRAY_SIZE(palm27x_dai), }; diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index b899a3b..ba15451 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -260,6 +260,7 @@ static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = static struct snd_soc_card snd_soc_raumfeld_connector = { .name = "Raumfeld Connector", + .owner = THIS_MODULE, .dai_link = snd_soc_raumfeld_connector_dai, .num_links = ARRAY_SIZE(snd_soc_raumfeld_connector_dai), .suspend_post = raumfeld_analog_suspend, @@ -268,6 +269,7 @@ static struct snd_soc_card snd_soc_raumfeld_connector = { static struct snd_soc_card snd_soc_raumfeld_speaker = { .name = "Raumfeld Speaker", + .owner = THIS_MODULE, .dai_link = snd_soc_raumfeld_speaker_dai, .num_links = ARRAY_SIZE(snd_soc_raumfeld_speaker_dai), .suspend_post = raumfeld_analog_suspend, diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index d9467a2..2e21712 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -125,6 +125,7 @@ static struct snd_soc_dai_link saarb_dai[] = { static struct snd_soc_card snd_soc_card_saarb = { .name = "Saarb", + .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), }; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c2d6ff9..bb06048 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -319,6 +319,7 @@ static struct snd_soc_dai_link spitz_dai = { /* spitz audio machine driver */ static struct snd_soc_card snd_soc_spitz = { .name = "Spitz", + .owner = THIS_MODULE, .dai_link = &spitz_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c index eeec892..4bef12c 100644 --- a/sound/soc/pxa/tavorevb3.c +++ b/sound/soc/pxa/tavorevb3.c @@ -125,6 +125,7 @@ static struct snd_soc_dai_link evb3_dai[] = { static struct snd_soc_card snd_soc_card_evb3 = { .name = "Tavor EVB3", + .owner = THIS_MODULE, .dai_link = evb3_dai, .num_links = ARRAY_SIZE(evb3_dai), }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 3f394de..564ef08 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -236,6 +236,7 @@ static struct snd_soc_dai_link tosa_dai[] = { static struct snd_soc_card tosa = { .name = "Tosa", + .owner = THIS_MODULE, .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), }; diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index b311ffe..d6807e0 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -202,6 +202,7 @@ static struct snd_soc_dai_link z2_dai = { /* z2 audio machine driver */ static struct snd_soc_card snd_soc_z2 = { .name = "Z2", + .owner = THIS_MODULE, .dai_link = &z2_dai, .num_links = 1, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 580aae3..ceb6566 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -249,6 +249,7 @@ static int zylonite_resume_pre(struct snd_soc_card *card) static struct snd_soc_card zylonite = { .name = "Zylonite", + .owner = THIS_MODULE, .probe = &zylonite_probe, .remove = &zylonite_remove, .suspend_post = &zylonite_suspend_post, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: omap: Add .owner to struct snd_soc_card
by Axel Lin 22 Dec '11

22 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/omap/am3517evm.c | 1 + sound/soc/omap/ams-delta.c | 1 + sound/soc/omap/igep0020.c | 1 + sound/soc/omap/n810.c | 1 + sound/soc/omap/omap3evm.c | 1 + sound/soc/omap/omap3pandora.c | 1 + sound/soc/omap/omap4-hdmi-card.c | 1 + sound/soc/omap/osk5912.c | 1 + sound/soc/omap/overo.c | 1 + sound/soc/omap/rx51.c | 1 + sound/soc/omap/sdp3430.c | 1 + sound/soc/omap/sdp4430.c | 1 + sound/soc/omap/zoom2.c | 1 + 13 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index c1cd4a0..add4866 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -107,6 +107,7 @@ static struct snd_soc_dai_link am3517evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", + .owner = THIS_MODULE, .dai_link = &am3517evm_dai, .num_links = 1, diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index a04a433..3e523a7 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -597,6 +597,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { /* Audio card driver */ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", + .owner = THIS_MODULE, .dai_link = &ams_delta_dai_link, .num_links = 1, .set_bias_level = ams_delta_set_bias_level, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 591fbf8..ccae58a 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link igep2_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_igep2 = { .name = "igep2", + .owner = THIS_MODULE, .dai_link = &igep2_dai, .num_links = 1, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fc6209b..597be41 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -289,6 +289,7 @@ static struct snd_soc_dai_link n810_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_n810 = { .name = "N810", + .owner = THIS_MODULE, .dai_link = &n810_dai, .num_links = 1, diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 6857895..071fcb0 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -70,6 +70,7 @@ static struct snd_soc_dai_link omap3evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_omap3evm = { .name = "omap3evm", + .owner = THIS_MODULE, .dai_link = &omap3evm_dai, .num_links = 1, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 7605c37..07794bd 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -233,6 +233,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_omap3pandora = { .name = "omap3pandora", + .owner = THIS_MODULE, .dai_link = omap3pandora_dai, .num_links = ARRAY_SIZE(omap3pandora_dai), }; diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c index 52d471c..28d689b 100644 --- a/sound/soc/omap/omap4-hdmi-card.c +++ b/sound/soc/omap/omap4-hdmi-card.c @@ -74,6 +74,7 @@ static struct snd_soc_dai_link omap4_hdmi_dai = { static struct snd_soc_card snd_soc_omap4_hdmi = { .name = "OMAP4HDMI", + .owner = THIS_MODULE, .dai_link = &omap4_hdmi_dai, .num_links = 1, }; diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 351ec9d..d859b59 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link osk_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", + .owner = THIS_MODULE, .dai_link = &osk_dai, .num_links = 1, diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index c3550ae..2ee889c 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link overo_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_overo = { .name = "overo", + .owner = THIS_MODULE, .dai_link = &overo_dai, .num_links = 1, }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index ad16db5..fada6ef 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -383,6 +383,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { /* Audio card */ static struct snd_soc_card rx51_sound_card = { .name = "RX-51", + .owner = THIS_MODULE, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e8fbf8e..2c85066 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -213,6 +213,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", + .owner = THIS_MODULE, .dai_link = sdp3430_dai, .num_links = ARRAY_SIZE(sdp3430_dai), diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 2735fa0..175ba9a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -226,6 +226,7 @@ static struct snd_soc_dai_link sdp4430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", + .owner = THIS_MODULE, .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 7641a7f..981616d 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -157,6 +157,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_zoom2 = { .name = "Zoom2", + .owner = THIS_MODULE, .dai_link = zoom2_dai, .num_links = ARRAY_SIZE(zoom2_dai), -- 1.7.5.4

2 1

[alsa-devel] [PATCH V5] ASoC: Allow DAI links to be specified using device tree nodes
by Stephen Warren 22 Dec '11

22 Dec '11

DAI link endpoints and platform (DMA) devices are currently specified by name. When instantiating sound cards from device tree, it may be more convenient to refer to these devices by phandle in the device tree, and for code to describe DAI links using the "struct device_node *" ("of_node") those phandles map to. This change adds new fields to snd_soc_dai_link which can "name" devices using of_node, enhances soc_bind_dai_link() to allow binding based on of_node, and enhances snd_soc_register_card() to ensure that illegal combinations of name and of_node are not used. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- v2: New patch implementing a new feature v3: Add Liam's ack v4: Rebased to re-order this patch directly onto ASoC for-next v5: Rebased again now that my other patches are ASoC for-next include/sound/soc.h | 4 +++ sound/soc/soc-core.c | 64 ++++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 61 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index db8acd2..8391b0e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -231,6 +231,7 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +struct device_node; struct snd_jack; struct snd_soc_card; struct snd_soc_pcm_stream; @@ -703,8 +704,11 @@ struct snd_soc_dai_link { const char *name; /* Codec name */ const char *stream_name; /* Stream name */ const char *codec_name; /* for multi-codec */ + const struct device_node *codec_of_node; const char *platform_name; /* for multi-platform */ + const struct device_node *platform_of_node; const char *cpu_dai_name; + const struct device_node *cpu_dai_of_node; const char *codec_dai_name; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42ad2db..a4592cb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -764,8 +764,13 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } /* no, then find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; + if (dai_link->cpu_dai_of_node) { + if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) + continue; + } else { + if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; + } rtd->cpu_dai = cpu_dai; goto find_codec; @@ -781,8 +786,13 @@ find_codec: /* no, then find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { - if (strcmp(codec->name, dai_link->codec_name)) - continue; + if (dai_link->codec_of_node) { + if (codec->dev->of_node != dai_link->codec_of_node) + continue; + } else { + if (strcmp(codec->name, dai_link->codec_name)) + continue; + } rtd->codec = codec; @@ -814,13 +824,19 @@ find_platform: /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; - if (!platform_name) + if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; /* no, then find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { - if (strcmp(platform->name, platform_name)) - continue; + if (dai_link->platform_of_node) { + if (platform->dev->of_node != + dai_link->platform_of_node) + continue; + } else { + if (strcmp(platform->name, platform_name)) + continue; + } rtd->platform = platform; goto out; @@ -2831,6 +2847,40 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai_link *link = &card->dai_link[i]; + + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codec_name == !!link->codec_of_node) { + dev_err(card->dev, + "Neither/both codec name/of_node are set\n"); + return -EINVAL; + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "Both platform name/of_node are set\n"); + return -EINVAL; + } + + /* + * CPU DAI must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + dev_err(card->dev, + "Neither/both cpu_dai name/of_node are set\n"); + return -EINVAL; + } + } + dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/3 v5] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Tomoya MORINAGA 22 Dec '11

22 Dec '11

ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high resistance to voltage noise. On chip regulator realizes power supply rejection ratio be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ ML26124-02GD can deliver stable audio performance without being affected by noise from the power supply circuit and peripheral components. The chip also includes a composite video signal output, which can be applied to various portable device requirements. The ML26124 is realized these functions into very small package the size is only 2.56mm x 2.46mm therefore can be construct high quality sound system easily. ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package. Signed-off-by: Tomoya MORINAGA <tomoya.rohm(a)gmail.com> --- V5: Add clock setting processing --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/ml26124.c | 594 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ml26124.h | 129 ++++++++++ 4 files changed, 728 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ml26124.c create mode 100644 sound/soc/codecs/ml26124.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7c205e7..2b4435f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_ML26124 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -224,6 +225,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_ML26124 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9aa6e66..22adcbd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-ml26124-objs := ml26124.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c new file mode 100644 index 0000000..fe2dc26 --- /dev/null +++ b/sound/soc/codecs/ml26124.c @@ -0,0 +1,594 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "ml26124.h" + +struct ml26124_priv { + u32 mclk; + u32 channels; +}; + +/* ML26124 configuration */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7150, 50, 0); +static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); + +static const char * const ml26124_companding[] = {"16bit PCM", "u-law", + "A-law"}; +static const struct soc_enum ml26124_adc_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); +static const struct soc_enum ml26124_dac_companding_enum + = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); + +static const struct snd_kcontrol_new ml26124_snd_controls[] = { + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0, + 0xff, 1, digital_tlv), + SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0, + 0xf, 1, alclvl), + SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0, + 7, 0, mingain), + SOC_SINGLE_TLV("ALC Max Input Volume", ML26124_ALC_MAXMIN_GAIN, 4, + 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Limitter Min Input Volume", + ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain), + SOC_SINGLE_TLV("Playback Limitter Max Input Volume", + ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain), + SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0, + 0x3f, 0, boost_vol), + SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0), + SOC_SINGLE("Noise High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0), + SOC_SINGLE("Zero Cross Comparator Switch", ML26124_PW_ZCCMP_PW_MNG, 1, + 1, 0), + SOC_SINGLE("EQ Band0 Switch", ML26124_FILTER_EN, 2, 1, 0), + SOC_SINGLE("EQ Band1 Switch", ML26124_FILTER_EN, 3, 1, 0), + SOC_SINGLE("EQ Band2 Switch", ML26124_FILTER_EN, 4, 1, 0), + SOC_SINGLE("EQ Band3 Switch", ML26124_FILTER_EN, 5, 1, 0), + SOC_SINGLE("EQ Band4 Switch", ML26124_FILTER_EN, 6, 1, 0), + SOC_SINGLE("Play Limitter", ML26124_DVOL_CTL, 0, 1, 0), + SOC_SINGLE("Capture Limitter", ML26124_DVOL_CTL, 1, 1, 0), + SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0), + SOC_SINGLE("Digital Switch", ML26124_DVOL_CTL, 4, 1, 0), + SOC_ENUM("DAC Companding", ml26124_dac_companding_enum), + SOC_ENUM("ADC Companding", ml26124_adc_companding_enum), +}; + +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0), + SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0), + SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0), +}; + +/* Input mux */ +static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", + "Analog MIC Differential in", + "Digital MIC in"}; + +static const struct soc_enum ml26124_insel_enum = + SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 2, ml26124_input_select); + +static const struct snd_kcontrol_new ml26124_input_mux_controls = + SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); + +static const struct snd_kcontrol_new ml26124_line_control = + SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0); + +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0, + &ml26124_output_mixer_controls[0], + ARRAY_SIZE(ml26124_output_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0), + SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ml26124_input_mux_controls), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ml26124_line_control), + SND_SOC_DAPM_INPUT("MDIN"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_OUTPUT("SPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), +}; + +static const struct snd_soc_dapm_route ml26124_intercon[] = { + /* Supply */ + {"DAC", NULL, "MCLK"}, + {"ADC", NULL, "MCLK"}, + {"DAC", NULL, "PLL"}, + {"ADC", NULL, "PLL"}, + + /* output mixer */ + {"Output Mixer", "PGA Switch", "PGA"}, + {"Output Mixer", "DAC Switch", "DAC"}, + {"Output Mixer", "Line in Switch", "LIN"}, + + /* outputs */ + {"LOUT", NULL, "Output Mixer"}, + {"SPOUT", NULL, "Output Mixer"}, + + /* input */ + {"Input Mux", "Analog MIC SingleEnded in", "MIN"}, + {"Input Mux", "Analog MIC Differential in", "MIN"}, + {"ADC", NULL, "Input Mux"}, +}; + +struct clk_coeff { + u32 mclk; + u32 rate; + u8 pllnl; + u8 pllnh; + u8 pllml; + u8 pllmh; + u8 plldiv; +}; + +/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */ +static const struct clk_coeff coeff_div[] = { + {12288800, 16000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 32000, 0xc, 0x0, 0x20, 0x0, 0x4}, + {12288800, 48000, 0xc, 0x0, 0x30, 0x0, 0x4}, +}; + +/* Get sampling rate value of sampling rate setting register (0x0) */ +static inline int get_srate(int rate) +{ + int srate; + + switch (rate) { + case 16000: + srate = 3; + break; + case 32000: + srate = 6; + break; + case 48000: + srate = 8; + break; + default: + return -EINVAL; + } + return srate; +} + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return -EINVAL; +} + +static int ml26124_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + int i = get_coeff(priv->mclk, params_rate(hw_params)); + + priv->channels = params_channels(hw_params); + + switch (params_rate(hw_params)) { + case 16000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 32000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + case 48000: + snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, + get_srate(coeff_div[i].rate)); + snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, + coeff_div[i].pllnl); + snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, + coeff_div[i].pllnh); + snd_soc_update_bits(codec, ML26124_PLLML, 0xff, + coeff_div[i].pllml); + snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, + coeff_div[i].pllmh); + snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, + coeff_div[i].plldiv); + break; + default: + pr_err("%s:this rate is no support for ml26124\n", __func__); + return -EINVAL; + } + + return 0; +} + +#define DVOL_CTL_DVMUTE_ON BIT(4) /* Digital volume MUTE On */ +#define DVOL_CTL_DVMUTE_OFF 0 /* Digital volume MUTE Off */ + +static int ml26124_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + if (mute) + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_ON); + else + snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4), + DVOL_CTL_DVMUTE_OFF); + return 0; +} + +#define ML26124_SAI_NO_DELAY BIT(1) +#define ML26124_SAI_FRAME_SYNC (BIT(5) | BIT(0)) /* For mono (Telecodec) */ + +static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + unsigned char mode; + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + mode = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode); + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + /* SAI configuration */ + if (priv->channels == 1) { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC); + } else { + snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, + ML26124_SAI_NO_DELAY); + snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, + ML26124_SAI_NO_DELAY); + } + + return 0; +} + +static int ml26124_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ML26124_USE_PLL: + break; + default: + return -EINVAL; + } + + switch (freq) { + case 12288000: + priv->mclk = freq; + break; + default: + return -EINVAL; + } + return 0; +} + +#define ML26134_CACHESIZE 79 +static const u16 ml26124_reg[ML26134_CACHESIZE] = { + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, + 0x0, 0x0, 0x0, /* System Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* Power Management */ + 0x4, /* Analog Reference Control */ + 0x10, 0x0, 0x33, 0x0, 0x0, /* Input/Output Amplifier Control */ + 0x0, 0x0, 0x1, /* Analog Path Contorl */ + 0x0, 0x0, 0x0, /* Audio Interface Control */ + 0x1, 0x0, 0x0, 0xff, 0xff, 0x10, /* DSP Control */ + 0x7, 0x7, 0x7, 0x7, 0x7, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x0, 0x0, /* DSP Control */ + 0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0, /* ALC Control */ + 0x4, 0x5, 0xd, 0x70, 0x10, 0x0, /* Playback Limiter Control */ + 0x1, 0x1, 0x1 /* Video Amplifer Control */ +}; + +#define ML26124_VMID BIT(0) + +static void ml26124_sync_cache(struct snd_soc_codec *codec) +{ + u8 *reg_cache = codec->reg_cache; + int i; + + if (!codec->cache_sync) + return; + + codec->cache_only = 0; + + for (i = 0; i < codec->driver->reg_cache_size; i++) { + if (reg_cache[i] == ml26124_reg[i]) + continue; + + snd_soc_write(codec, i, reg_cache[i]); + } + + codec->cache_sync = 0; +} + +static int ml26124_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* VMID ON */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, ML26124_VMID); + msleep(500); + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ml26124_sync_cache(codec); + break; + case SND_SOC_BIAS_OFF: + /* VMID OFF */ + snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, + ML26124_VMID, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int ml26124_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0); + return 0; + } else if (cmd == SNDRV_PCM_TRIGGER_START) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 1); + else + snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, + 2); + } + + return 0; +} + +#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_48000) +#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static const struct snd_soc_dai_ops ml26124_dai_ops = { + .hw_params = ml26124_hw_params, + .digital_mute = ml26124_mute, + .set_fmt = ml26124_set_dai_fmt, + .set_sysclk = ml26124_set_dai_sysclk, + .trigger = ml26124_pcm_trigger, +}; + +static struct snd_soc_dai_driver ml26124_dai = { + .name = "ml26124-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ML26124_RATES, + .formats = ML26124_FORMATS,}, + .ops = &ml26124_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int ml26124_resume(struct snd_soc_codec *codec) +{ + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define ml26124_suspend NULL +#define ml26124_resume NULL +#endif + +static int ml26124_probe(struct snd_soc_codec *codec) +{ + int ret; + + 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: %d\n", ret); + return ret; + } + + /* Software Reset */ + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); + + ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = { + .probe = ml26124_probe, + .suspend = ml26124_suspend, + .resume = ml26124_resume, + .set_bias_level = ml26124_set_bias_level, + .reg_cache_size = ML26134_CACHESIZE, + .reg_word_size = sizeof(u8), + .reg_cache_default = ml26124_reg, + .dapm_widgets = ml26124_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets), + .dapm_routes = ml26124_intercon, + .num_dapm_routes = ARRAY_SIZE(ml26124_intercon), + .controls = ml26124_snd_controls, + .num_controls = ARRAY_SIZE(ml26124_snd_controls), +}; + +static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ml26124_priv *priv; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_ml26124, &ml26124_dai, 1); +} + +static __devexit int ml26124_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ml26124_i2c_id[] = { + { "ml26124", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); + +static struct i2c_driver ml26124_i2c_driver = { + .driver = { + .name = "ml26124", + .owner = THIS_MODULE, + }, + .probe = ml26124_i2c_probe, + .remove = __devexit_p(ml26124_i2c_remove), + .id_table = ml26124_i2c_id, +}; + +static int __init ml26124_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&ml26124_i2c_driver); + if (ret != 0) + pr_err("Failed to register ML26124 I2C driver: %d\n", ret); + + return ret; +} +module_init(ml26124_modinit); + +static void __exit ml26124_exit(void) +{ + i2c_del_driver(&ml26124_i2c_driver); +} +module_exit(ml26124_exit); + +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); +MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h new file mode 100644 index 0000000..4fe7337 --- /dev/null +++ b/sound/soc/codecs/ml26124.h @@ -0,0 +1,129 @@ +/* + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd. + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef ML26124_H +#define ML26124_H + +/* Clock Control Register */ +#define ML26124_SMPLING_RATE 0x00 +#define ML26124_PLLNL 0x02 +#define ML26124_PLLNH 0x04 +#define ML26124_PLLML 0x06 +#define ML26124_PLLMH 0x08 +#define ML26124_PLLDIV 0x0a +#define ML26124_CLK_EN 0x0c +#define ML26124_CLK_CTL 0x0e + +/* System Control Register */ +#define ML26124_SW_RST 0x10 +#define ML26124_REC_PLYBAK_RUN 0x12 +#define ML26124_MIC_TIM 0x14 + +/* Power Mnagement Register */ +#define ML26124_PW_REF_PW_MNG 0x20 +#define ML26124_PW_IN_PW_MNG 0x22 +#define ML26124_PW_DAC_PW_MNG 0x24 +#define ML26124_PW_SPAMP_PW_MNG 0x26 +#define ML26124_PW_LOUT_PW_MNG 0x28 +#define ML26124_PW_VOUT_PW_MNG 0x2a +#define ML26124_PW_ZCCMP_PW_MNG 0x2e + +/* Analog Reference Control Register */ +#define ML26124_PW_MICBIAS_VOL 0x30 + +/* Input/Output Amplifier Control Register */ +#define ML26124_PW_MIC_IN_VOL 0x32 +#define ML26124_PW_MIC_BOST_VOL 0x38 +#define ML26124_PW_SPK_AMP_VOL 0x3a +#define ML26124_PW_AMP_VOL_FUNC 0x48 +#define ML26124_PW_AMP_VOL_FADE 0x4a + +/* Analog Path Control Register */ +#define ML26124_SPK_AMP_OUT 0x54 +#define ML26124_MIC_IF_CTL 0x5a +#define ML26124_MIC_SELECT 0xe8 + +/* Audio Interface Control Register */ +#define ML26124_SAI_TRANS_CTL 0x60 +#define ML26124_SAI_RCV_CTL 0x62 +#define ML26124_SAI_MODE_SEL 0x64 + +/* DSP Control Register */ +#define ML26124_FILTER_EN 0x66 +#define ML26124_DVOL_CTL 0x68 +#define ML26124_MIXER_VOL_CTL 0x6a +#define ML26124_RECORD_DIG_VOL 0x6c +#define ML26124_PLBAK_DIG_VOL 0x70 +#define ML26124_DIGI_BOOST_VOL 0x72 +#define ML26124_EQ_GAIN_BRAND0 0x74 +#define ML26124_EQ_GAIN_BRAND1 0x76 +#define ML26124_EQ_GAIN_BRAND2 0x78 +#define ML26124_EQ_GAIN_BRAND3 0x7a +#define ML26124_EQ_GAIN_BRAND4 0x7c +#define ML26124_HPF2_CUTOFF 0x7e +#define ML26124_EQBRAND0_F0L 0x80 +#define ML26124_EQBRAND0_F0H 0x82 +#define ML26124_EQBRAND0_F1L 0x84 +#define ML26124_EQBRAND0_F1H 0x86 +#define ML26124_EQBRAND1_F0L 0x88 +#define ML26124_EQBRAND1_F0H 0x8a +#define ML26124_EQBRAND1_F1L 0x8c +#define ML26124_EQBRAND1_F1H 0x8e +#define ML26124_EQBRAND2_F0L 0x90 +#define ML26124_EQBRAND2_F0H 0x92 +#define ML26124_EQBRAND2_F1L 0x94 +#define ML26124_EQBRAND2_F1H 0x96 +#define ML26124_EQBRAND3_F0L 0x98 +#define ML26124_EQBRAND3_F0H 0x9a +#define ML26124_EQBRAND3_F1L 0x9c +#define ML26124_EQBRAND3_F1H 0x9e +#define ML26124_EQBRAND4_F0L 0xa0 +#define ML26124_EQBRAND4_F0H 0xa2 +#define ML26124_EQBRAND4_F1L 0xa4 +#define ML26124_EQBRAND4_F1H 0xa6 + +/* ALC Control Register */ +#define ML26124_ALC_MODE 0xb0 +#define ML26124_ALC_ATTACK_TIM 0xb2 +#define ML26124_ALC_DECAY_TIM 0xb4 +#define ML26124_ALC_HOLD_TIM 0xb6 +#define ML26124_ALC_TARGET_LEV 0xb8 +#define ML26124_ALC_MAXMIN_GAIN 0xba +#define ML26124_NOIS_GATE_THRSH 0xbc +#define ML26124_ALC_ZERO_TIMOUT 0xbe + +/* Playback Limiter Control Register */ +#define ML26124_PL_ATTACKTIME 0xc0 +#define ML26124_PL_DECAYTIME 0xc2 +#define ML26124_PL_TARGETTIME 0xc4 +#define ML26124_PL_MAXMIN_GAIN 0xc6 +#define ML26124_PLYBAK_BOST_VOL 0xc8 +#define ML26124_PL_0CROSS_TIMOUT 0xca + +/* Video Amplifer Control Register */ +#define ML26124_VIDEO_AMP_GAIN_CTL 0xd0 +#define ML26124_VIDEO_AMP_SETUP1 0xd2 +#define ML26124_VIDEO_AMP_CTL2 0xd4 + +/* Clock select for machine driver */ +#define ML26124_USE_PLL 0 +#define ML26124_USE_MCLKI_256FS 1 +#define ML26124_USE_MCLKI_512FS 2 +#define ML26124_USE_MCLKI_1024FS 3 + + +#endif -- 1.7.4.4

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Re: [alsa-devel] [PATCH 07/10] ASoC: OMAP: ams-delta: Drive modem/codec pins over GPIO API
by Liam Girdwood 22 Dec '11

22 Dec '11

On Sun, 2011-12-11 at 21:12 +0100, Janusz Krzysztofik wrote: > Don't use Amstrad Delta custom I/O functions any longer, replace them > with GPIO. Old pin definitions, no longer used by the modem bits either, > can be dropped. > > Depends on patch 2/10 "ARM: OMAP1: Convert Amstrad E3 latches to > basic_mmio_gpio". > > Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> Applied. Thanks Liam

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[alsa-devel] [PATCH 00/21] ALSA: asihpi - update to 4.10
by linux＠audioscience.com 22 Dec '11

22 Dec '11

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[alsa-devel] [PATCH] ALSA: asihpi - fix pcm dma pointer tracking
by linux＠audioscience.com 22 Dec '11

22 Dec '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Elapsed counter should only count data committed to snd_pcm_period_elapsed, rather than all data available Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/asihpi.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e9de799..322ddfe 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -888,8 +888,8 @@ static void snd_card_asihpi_timer_function(unsigned long data) pd, xfer2)); } } - ds->pcm_buf_host_rw_ofs = ds->pcm_buf_host_rw_ofs + xfercount; - ds->pcm_buf_elapsed_dma_ofs = pcm_buf_dma_ofs; + ds->pcm_buf_host_rw_ofs = += xfercount; + ds->pcm_buf_elapsed_dma_ofs += xfercount; snd_pcm_period_elapsed(s); } } -- 1.7.0.4

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[alsa-devel] how to enumerate useful ALSA devices?
by Joerg-Cyril.Hoehle＠t-systems.com 22 Dec '11

22 Dec '11

Hi, The 2008 audio guide recommends http://0pointer.de/blog/projects/guide-to-sound-apis.html "Do not use the snd_card_xxx() APIs. For enumerating use snd_device_name_hint() (and related functions). snd_card_xxx() is obsolete... Bluetooth audio are not included" Is that correct advice? Wine still uses snd_card, so it knows only about "default" and "plughw:x,y". I believe that's wrong but don't know what should be used instead in order not to come up with a long list of artificial names mentioned in /usr/share/alsa/alsa.conf such as surround51 when the poor on-board card knows nothing but 16 bit stereo. I'm looking for a list of working devices on this machine, which likely should contain "default" (usually leading to PA on a recent Ubuntu machine), "plug:dmix" (PA getting out of the way when trying to use other devices), "hw:0,0" and presumably not much more. Thank you for your help, Jörg Höhle

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