diff options
Diffstat (limited to 'sound')
90 files changed, 7695 insertions, 2057 deletions
diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1ed61c5df2c..4f913876f33 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o +snd-soc-core-objs += soc-pcm.o soc-io.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index d0e75323ec1..f81d4c3f895 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -364,9 +364,11 @@ static struct snd_pcm_ops atmel_pcm_ops = { \*--------------------------------------------------------------------------*/ static u64 atmel_pcm_dmamask = 0xffffffff; -static int atmel_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) @@ -382,7 +384,7 @@ static int atmel_pcm_new(struct snd_card *card, } if (dai->driver->capture.channels_min) { - pr_debug("at32-pcm:" + pr_debug("atmel-pcm:" "Allocating PCM capture DMA buffer\n"); ret = atmel_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 2597329302e..5e0a95e6432 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -60,7 +60,7 @@ struct atmel_ssc_mask { * This structure, shared between the PCM driver and the interface, * contains all information required by the PCM driver to perform the * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM + * by the interface. The dma_intr_handler() pointer is set by the PCM * driver and called by the interface SSC interrupt handler if it is * non-NULL. */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index eda955b1583..71225090c49 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -402,7 +402,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S && bits > 16) { printk(KERN_WARNING - "atmel_ssc_dai: sample size %d" + "atmel_ssc_dai: sample size %d " "is too large for I2S\n", bits); return -EINVAL; } @@ -838,10 +838,8 @@ int atmel_ssc_set_audio(int ssc_id) } ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id); - if (!ssc_pdev) { - ssc_free(ssc); + if (!ssc_pdev) return -ENOMEM; - } /* If we can grab the SSC briefly to parent the DAI device off it */ ssc = ssc_request(ssc_id); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 95572d290c2..bad3aa14d5b 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -92,6 +92,7 @@ static struct snd_soc_ops at91sam9g20ek_ops = { }; static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { static int mclk_on; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 10fdd2854e5..20bb53a837b 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -319,10 +319,11 @@ static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int au1xpsc_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index ae403597fd3..fe9d548a683 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -10,13 +10,36 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) select SND_BF5XX_SOC_I2S select SND_SOC_SSM2602 - select I2C help Say Y if you want to add support for SoC audio on BF527-EZKIT. +config SND_SOC_BFIN_EVAL_ADAU1701 + tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_ADAU1701 + select I2C + help + Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ + board connected to one of the Blackfin evaluation boards like the + BF5XX-STAMP or BF5XX-EZKIT. + +config SND_SOC_BFIN_EVAL_ADAV80X + tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" + depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + select SND_BF5XX_SOC_I2S + select SND_SOC_ADAV80X + help + Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or + EVAL-ADAV803 board connected to one of the Blackfin evaluation boards + like the BF5XX-STAMP or BF5XX-EZKIT. + + Note: This driver assumes that the ADAV80X digital record and playback + interfaces are connected to the first SPORT port on the BF5XX board. + config SND_BF5XX_SOC_AD73311 tristate "SoC AD73311 Audio support for Blackfin" depends on SND_BF5XX_I2S diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 49af3f32aec..6018bf52a23 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -21,9 +21,13 @@ snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad193x-objs := bf5xx-ad193x.o +snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o +snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o +obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o +obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 98b44b316e7..9e59f680bc1 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -418,9 +418,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("%s enter\n", __func__); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index f1fd95bb641..61ddf942fd4 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -168,7 +168,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - ret = snd_pcm_hw_constraint_integer(runtime, \ + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) goto out; @@ -257,9 +257,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("%s enter\n", __func__); @@ -304,8 +306,8 @@ static int __devexit bfin_i2s_soc_platform_remove(struct platform_device *pdev) static struct platform_driver bfin_i2s_pcm_driver = { .driver = { - .name = "bfin-i2s-pcm-audio", - .owner = THIS_MODULE, + .name = "bfin-i2s-pcm-audio", + .owner = THIS_MODULE, }, .probe = bfin_i2s_soc_platform_probe, diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 07cfc7a9e49..c95cc03d583 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -283,9 +283,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c new file mode 100644 index 00000000000..e5550acba2c --- /dev/null +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -0,0 +1,139 @@ +/* + * Machine driver for EVAL-ADAU1701MINIZ on Analog Devices bfin + * evaluation boards. + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "../codecs/adau1701.h" + +static const struct snd_soc_dapm_widget bfin_eval_adau1701_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route bfin_eval_adau1701_dapm_routes[] = { + { "Speaker", NULL, "OUT0" }, + { "Speaker", NULL, "OUT1" }, + { "Line Out", NULL, "OUT2" }, + { "Line Out", NULL, "OUT3" }, + + { "IN0", NULL, "Line In" }, + { "IN1", NULL, "Line In" }, +}; + +static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000, + SND_SOC_CLOCK_IN); + + return ret; +} + +static struct snd_soc_ops bfin_eval_adau1701_ops = { + .hw_params = bfin_eval_adau1701_hw_params, +}; + +static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { + { + .name = "adau1701", + .stream_name = "adau1701", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "adau1701", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "adau1701.0-0034", + .ops = &bfin_eval_adau1701_ops, + }, + { + .name = "adau1701", + .stream_name = "adau1701", + .cpu_dai_name = "bfin-i2s.1", + .codec_dai_name = "adau1701", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "adau1701.0-0034", + .ops = &bfin_eval_adau1701_ops, + }, +}; + +static struct snd_soc_card bfin_eval_adau1701 = { + .name = "bfin-eval-adau1701", + .dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM], + .num_links = 1, + + .dapm_widgets = bfin_eval_adau1701_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1701_dapm_widgets), + .dapm_routes = bfin_eval_adau1701_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1701_dapm_routes), +}; + +static int bfin_eval_adau1701_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &bfin_eval_adau1701; + + card->dev = &pdev->dev; + + return snd_soc_register_card(&bfin_eval_adau1701); +} + +static int __devexit bfin_eval_adau1701_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver bfin_eval_adau1701_driver = { + .driver = { + .name = "bfin-eval-adau1701", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bfin_eval_adau1701_probe, + .remove = __devexit_p(bfin_eval_adau1701_remove), +}; + +static int __init bfin_eval_adau1701_init(void) +{ + return platform_driver_register(&bfin_eval_adau1701_driver); +} +module_init(bfin_eval_adau1701_init); + +static void __exit bfin_eval_adau1701_exit(void) +{ + platform_driver_unregister(&bfin_eval_adau1701_driver); +} +module_exit(bfin_eval_adau1701_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bfin-eval-adau1701"); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c new file mode 100644 index 00000000000..8d014d01626 --- /dev/null +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -0,0 +1,173 @@ +/* + * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin + * evaluation boards. + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#include "../codecs/adav80x.h" + +static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = { + { "Line Out", NULL, "VOUTL" }, + { "Line Out", NULL, "VOUTR" }, + + { "VINL", NULL, "Line In" }, + { "VINR", NULL, "Line In" }, +}; + +static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL, + 27000000, params_rate(params) * 256); + if (ret) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + + return ret; +} + +static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0, + SND_SOC_CLOCK_OUT); + snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0, + SND_SOC_CLOCK_OUT); + snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0, + SND_SOC_CLOCK_OUT); + + snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0); + + return 0; +} + +static struct snd_soc_ops bfin_eval_adav80x_ops = { + .hw_params = bfin_eval_adav80x_hw_params, +}; + +static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { + { + .name = "adav80x", + .stream_name = "ADAV80x HiFi", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "adav80x-hifi", + .platform_name = "bfin-i2s-pcm-audio", + .init = bfin_eval_adav80x_codec_init, + .ops = &bfin_eval_adav80x_ops, + }, +}; + +static struct snd_soc_card bfin_eval_adav80x = { + .name = "bfin-eval-adav80x", + .dai_link = bfin_eval_adav80x_dais, + .num_links = ARRAY_SIZE(bfin_eval_adav80x_dais), + + .dapm_widgets = bfin_eval_adav80x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets), + .dapm_routes = bfin_eval_adav80x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(bfin_eval_adav80x_dapm_routes), +}; + +enum bfin_eval_adav80x_type { + BFIN_EVAL_ADAV801, + BFIN_EVAL_ADAV803, +}; + +static int bfin_eval_adav80x_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &bfin_eval_adav80x; + const char *codec_name; + + switch (platform_get_device_id(pdev)->driver_data) { + case BFIN_EVAL_ADAV801: + codec_name = "spi0.1"; + break; + case BFIN_EVAL_ADAV803: + codec_name = "adav803.0-0034"; + break; + default: + return -EINVAL; + } + + bfin_eval_adav80x_dais[0].codec_name = codec_name; + + card->dev = &pdev->dev; + + return snd_soc_register_card(&bfin_eval_adav80x); +} + +static int __devexit bfin_eval_adav80x_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct platform_device_id bfin_eval_adav80x_ids[] = { + { "bfin-eval-adav801", BFIN_EVAL_ADAV801 }, + { "bfin-eval-adav803", BFIN_EVAL_ADAV803 }, + { }, +}; +MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids); + +static struct platform_driver bfin_eval_adav80x_driver = { + .driver = { + .name = "bfin-eval-adav80x", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bfin_eval_adav80x_probe, + .remove = __devexit_p(bfin_eval_adav80x_remove), + .id_table = bfin_eval_adav80x_ids, +}; + +static int __init bfin_eval_adav80x_init(void) +{ + return platform_driver_register(&bfin_eval_adav80x_driver); +} +module_init(bfin_eval_adav80x_init); + +static void __exit bfin_eval_adav80x_exit(void) +{ + platform_driver_unregister(&bfin_eval_adav80x_driver); +} +module_exit(bfin_eval_adav80x_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a096df..ff43405752a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -42,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_STA32X if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -71,6 +73,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C @@ -130,7 +133,14 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate - + +config SND_SOC_ADAU1701 + select SIGMA + tristate + +config SND_SOC_ADAV80X + tristate + config SND_SOC_ADS117X tristate @@ -216,6 +226,9 @@ config SND_SOC_SPDIF config SND_SOC_SSM2602 tristate +config SND_SOC_STA32X + tristate + config SND_SOC_STAC9766 tristate @@ -299,6 +312,9 @@ config SND_SOC_WM8770 config SND_SOC_WM8776 tristate +config SND_SOC_WM8782 + tristate + config SND_SOC_WM8804 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd8558406ef..4957431e23f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,8 @@ snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-adau1701-objs := adau1701.o +snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -28,6 +30,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -55,6 +58,7 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o +snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o @@ -95,6 +99,8 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -120,6 +126,7 @@ obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o @@ -147,6 +154,7 @@ obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o +obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 754c496412b..4e5c5726366 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -1,19 +1,10 @@ -/* - * File: sound/soc/codecs/ad1836.c - * Author: Barry Song <Barry.Song@analog.com> - * - * Created: Aug 04 2009 - * Description: Driver for AD1836 sound chip - * - * Modified: - * Copyright 2009 Analog Devices Inc. + /* + * Audio Codec driver supporting: + * AD1835A, AD1836, AD1837A, AD1838A, AD1839A * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * Copyright 2009-2011 Analog Devices Inc. * - * 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. + * Licensed under the GPL-2 or later. */ #include <linux/init.h> @@ -30,10 +21,15 @@ #include <linux/spi/spi.h> #include "ad1836.h" +enum ad1836_type { + AD1835, + AD1836, + AD1838, +}; + /* codec private data */ struct ad1836_priv { - enum snd_soc_control_type control_type; - void *control_data; + enum ad1836_type type; }; /* @@ -44,29 +40,60 @@ static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; static const struct soc_enum ad1836_deemp_enum = SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); -static const struct snd_kcontrol_new ad1836_snd_controls[] = { - /* DAC volume control */ - SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL, - AD1836_DAC_R1_VOL, 0, 0x3FF, 0), - SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL, - AD1836_DAC_R2_VOL, 0, 0x3FF, 0), - SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL, - AD1836_DAC_R3_VOL, 0, 0x3FF, 0), - - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE, - AD1836_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE, - AD1836_ADCR2_MUTE, 1, 1), - - /* DAC switch control */ - SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE, - AD1836_DACR1_MUTE, 1, 1), - SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE, - AD1836_DACR2_MUTE, 1, 1), - SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE, - AD1836_DACR3_MUTE, 1, 1), +#define AD1836_DAC_VOLUME(x) \ + SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \ + AD1836_DAC_R_VOL(x), 0, 0x3FF, 0) + +#define AD1836_DAC_SWITCH(x) \ + SOC_DOUBLE("DAC" #x " Playback Switch", AD1836_DAC_CTRL2, \ + AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1) + +#define AD1836_ADC_SWITCH(x) \ + SOC_DOUBLE("ADC" #x " Capture Switch", AD1836_ADC_CTRL2, \ + AD1836_MUTE_LEFT(x), AD1836_MUTE_RIGHT(x), 1, 1) + +static const struct snd_kcontrol_new ad183x_dac_controls[] = { + AD1836_DAC_VOLUME(1), + AD1836_DAC_SWITCH(1), + AD1836_DAC_VOLUME(2), + AD1836_DAC_SWITCH(2), + AD1836_DAC_VOLUME(3), + AD1836_DAC_SWITCH(3), + AD1836_DAC_VOLUME(4), + AD1836_DAC_SWITCH(4), +}; + +static const struct snd_soc_dapm_widget ad183x_dac_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_route ad183x_dac_routes[] = { + { "DAC1OUT", NULL, "DAC" }, + { "DAC2OUT", NULL, "DAC" }, + { "DAC3OUT", NULL, "DAC" }, + { "DAC4OUT", NULL, "DAC" }, +}; + +static const struct snd_kcontrol_new ad183x_adc_controls[] = { + AD1836_ADC_SWITCH(1), + AD1836_ADC_SWITCH(2), + AD1836_ADC_SWITCH(3), +}; + +static const struct snd_soc_dapm_widget ad183x_adc_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route ad183x_adc_routes[] = { + { "ADC", NULL, "ADC1IN" }, + { "ADC", NULL, "ADC2IN" }, +}; +static const struct snd_kcontrol_new ad183x_controls[] = { /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1, AD1836_ADC_HIGHPASS_FILTER, 1, 0), @@ -75,27 +102,24 @@ static const struct snd_kcontrol_new ad1836_snd_controls[] = { SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum), }; -static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { +static const struct snd_soc_dapm_widget ad183x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1, AD1836_DAC_POWERDOWN, 1), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1, AD1836_ADC_POWERDOWN, 1, NULL, 0), - SND_SOC_DAPM_OUTPUT("DAC1OUT"), - SND_SOC_DAPM_OUTPUT("DAC2OUT"), - SND_SOC_DAPM_OUTPUT("DAC3OUT"), - SND_SOC_DAPM_INPUT("ADC1IN"), - SND_SOC_DAPM_INPUT("ADC2IN"), }; -static const struct snd_soc_dapm_route audio_paths[] = { +static const struct snd_soc_dapm_route ad183x_dapm_routes[] = { { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", "DAC1 Switch", "DAC" }, - { "DAC2OUT", "DAC2 Switch", "DAC" }, - { "DAC3OUT", "DAC3 Switch", "DAC" }, - { "ADC", "ADC1 Switch", "ADC1IN" }, - { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +static const DECLARE_TLV_DB_SCALE(ad1836_in_tlv, 0, 300, 0); + +static const struct snd_kcontrol_new ad1836_controls[] = { + SOC_DOUBLE_TLV("ADC2 Capture Volume", AD1836_ADC_CTRL1, 3, 0, 4, 0, + ad1836_in_tlv), }; /* @@ -165,64 +189,69 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, return 0; } +static struct snd_soc_dai_ops ad1836_dai_ops = { + .hw_params = ad1836_hw_params, + .set_fmt = ad1836_set_dai_fmt, +}; + +#define AD183X_DAI(_name, num_dacs, num_adcs) \ +{ \ + .name = _name "-hifi", \ + .playback = { \ + .stream_name = "Playback", \ + .channels_min = 2, \ + .channels_max = (num_dacs) * 2, \ + .rates = SNDRV_PCM_RATE_48000, \ + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \ + }, \ + .capture = { \ + .stream_name = "Capture", \ + .channels_min = 2, \ + .channels_max = (num_adcs) * 2, \ + .rates = SNDRV_PCM_RATE_48000, \ + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, \ + }, \ + .ops = &ad1836_dai_ops, \ +} + +static struct snd_soc_dai_driver ad183x_dais[] = { + [AD1835] = AD183X_DAI("ad1835", 4, 1), + [AD1836] = AD183X_DAI("ad1836", 3, 2), + [AD1838] = AD183X_DAI("ad1838", 3, 1), +}; + #ifdef CONFIG_PM -static int ad1836_soc_suspend(struct snd_soc_codec *codec, - pm_message_t state) +static int ad1836_suspend(struct snd_soc_codec *codec, pm_message_t state) { /* reset clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + AD1836_ADC_SERFMT_MASK, 0); } -static int ad1836_soc_resume(struct snd_soc_codec *codec) +static int ad1836_resume(struct snd_soc_codec *codec) { /* restore clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 |= AD1836_ADC_AUX; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX); } #else -#define ad1836_soc_suspend NULL -#define ad1836_soc_resume NULL +#define ad1836_suspend NULL +#define ad1836_resume NULL #endif -static struct snd_soc_dai_ops ad1836_dai_ops = { - .hw_params = ad1836_hw_params, - .set_fmt = ad1836_set_dai_fmt, -}; - -/* codec DAI instance */ -static struct snd_soc_dai_driver ad1836_dai = { - .name = "ad1836-hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 6, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .ops = &ad1836_dai_ops, -}; - static int ad1836_probe(struct snd_soc_codec *codec) { struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + int num_dacs, num_adcs; int ret = 0; + int i; + + num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2; + num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2; - codec->control_data = ad1836->control_data; ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O: %d\n", @@ -239,21 +268,46 @@ static int ad1836_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); /* unmute adc channles, adc aux mode */ snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); - /* left/right diff:PGA/MUX */ - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); /* volume */ - snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF); - snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF); - - snd_soc_add_controls(codec, ad1836_snd_controls, - ARRAY_SIZE(ad1836_snd_controls)); - snd_soc_dapm_new_controls(dapm, ad1836_dapm_widgets, - ARRAY_SIZE(ad1836_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); + for (i = 1; i <= num_dacs; ++i) { + snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF); + snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF); + } + + if (ad1836->type == AD1836) { + /* left/right diff:PGA/MUX */ + snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); + ret = snd_soc_add_controls(codec, ad1836_controls, + ARRAY_SIZE(ad1836_controls)); + if (ret) + return ret; + } else { + snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00); + } + + ret = snd_soc_add_controls(codec, ad183x_dac_controls, num_dacs * 2); + if (ret) + return ret; + + ret = snd_soc_add_controls(codec, ad183x_adc_controls, num_adcs); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(dapm, ad183x_dac_dapm_widgets, num_dacs); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(dapm, ad183x_adc_dapm_widgets, num_adcs); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, ad183x_dac_routes, num_dacs); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, ad183x_adc_routes, num_adcs); + if (ret) + return ret; return ret; } @@ -262,19 +316,24 @@ static int ad1836_probe(struct snd_soc_codec *codec) static int ad1836_remove(struct snd_soc_codec *codec) { /* reset clock control mode */ - u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); - adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - - return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + AD1836_ADC_SERFMT_MASK, 0); } static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { - .probe = ad1836_probe, - .remove = ad1836_remove, - .suspend = ad1836_soc_suspend, - .resume = ad1836_soc_resume, + .probe = ad1836_probe, + .remove = ad1836_remove, + .suspend = ad1836_suspend, + .resume = ad1836_resume, .reg_cache_size = AD1836_NUM_REGS, .reg_word_size = sizeof(u16), + + .controls = ad183x_controls, + .num_controls = ARRAY_SIZE(ad183x_controls), + .dapm_widgets = ad183x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad183x_dapm_widgets), + .dapm_routes = ad183x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes), }; static int __devinit ad1836_spi_probe(struct spi_device *spi) @@ -286,12 +345,12 @@ static int __devinit ad1836_spi_probe(struct spi_device *spi) if (ad1836 == NULL) return -ENOMEM; + ad1836->type = spi_get_device_id(spi)->driver_data; + spi_set_drvdata(spi, ad1836); - ad1836->control_data = spi; - ad1836->control_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ad1836, &ad1836_dai, 1); + &soc_codec_dev_ad1836, &ad183x_dais[ad1836->type], 1); if (ret < 0) kfree(ad1836); return ret; @@ -303,27 +362,29 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi) kfree(spi_get_drvdata(spi)); return 0; } +static const struct spi_device_id ad1836_ids[] = { + { "ad1835", AD1835 }, + { "ad1836", AD1836 }, + { "ad1837", AD1835 }, + { "ad1838", AD1838 }, + { "ad1839", AD1838 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, ad1836_ids); static struct spi_driver ad1836_spi_driver = { .driver = { - .name = "ad1836-codec", + .name = "ad1836", .owner = THIS_MODULE, }, .probe = ad1836_spi_probe, .remove = __devexit_p(ad1836_spi_remove), + .id_table = ad1836_ids, }; static int __init ad1836_init(void) { - int ret; - - ret = spi_register_driver(&ad1836_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n", - ret); - } - - return ret; + return spi_register_driver(&ad1836_spi_driver); } module_init(ad1836_init); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 9d6a3f8f8aa..444747f0db2 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -1,19 +1,10 @@ /* - * File: sound/soc/codecs/ad1836.h - * Based on: - * Author: Barry Song <Barry.Song@analog.com> + * Audio Codec driver supporting: + * AD1835A, AD1836, AD1837A, AD1838A, AD1839A * - * Created: Aug 04, 2009 - * Description: definitions for AD1836 registers + * Copyright 2009-2011 Analog Devices Inc. * - * Modified: - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * 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. + * Licensed under the GPL-2 or later. */ #ifndef __AD1836_H__ @@ -21,39 +12,30 @@ #define AD1836_DAC_CTRL1 0 #define AD1836_DAC_POWERDOWN 2 -#define AD1836_DAC_SERFMT_MASK 0xE0 +#define AD1836_DAC_SERFMT_MASK 0xE0 #define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) #define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) #define AD1836_DAC_WORD_LEN_MASK 0x18 #define AD1836_DAC_WORD_LEN_OFFSET 3 #define AD1836_DAC_CTRL2 1 -#define AD1836_DACL1_MUTE 0 -#define AD1836_DACR1_MUTE 1 -#define AD1836_DACL2_MUTE 2 -#define AD1836_DACR2_MUTE 3 -#define AD1836_DACL3_MUTE 4 -#define AD1836_DACR3_MUTE 5 -#define AD1836_DAC_L1_VOL 2 -#define AD1836_DAC_R1_VOL 3 -#define AD1836_DAC_L2_VOL 4 -#define AD1836_DAC_R2_VOL 5 -#define AD1836_DAC_L3_VOL 6 -#define AD1836_DAC_R3_VOL 7 +/* These macros are one-based. So AD183X_MUTE_LEFT(1) will return the mute bit + * for the first ADC/DAC */ +#define AD1836_MUTE_LEFT(x) (((x) * 2) - 2) +#define AD1836_MUTE_RIGHT(x) (((x) * 2) - 1) + +#define AD1836_DAC_L_VOL(x) ((x) * 2) +#define AD1836_DAC_R_VOL(x) (1 + ((x) * 2)) #define AD1836_ADC_CTRL1 12 #define AD1836_ADC_POWERDOWN 7 #define AD1836_ADC_HIGHPASS_FILTER 8 #define AD1836_ADC_CTRL2 13 -#define AD1836_ADCL1_MUTE 0 -#define AD1836_ADCR1_MUTE 1 -#define AD1836_ADCL2_MUTE 2 -#define AD1836_ADCR2_MUTE 3 #define AD1836_ADC_WORD_LEN_MASK 0x30 #define AD1836_ADC_WORD_OFFSET 5 -#define AD1836_ADC_SERFMT_MASK (7 << 6) +#define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) #define AD1836_ADC_AUX (0x6 << 6) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c new file mode 100644 index 00000000000..2758d5fc60d --- /dev/null +++ b/sound/soc/codecs/adau1701.c @@ -0,0 +1,549 @@ +/* + * Driver for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * based on an inital version by Cliff Cai <cliff.cai@analog.com> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/sigma.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau1701.h" + +#define ADAU1701_DSPCTRL 0x1c +#define ADAU1701_SEROCTL 0x1e +#define ADAU1701_SERICTL 0x1f + +#define ADAU1701_AUXNPOW 0x22 + +#define ADAU1701_OSCIPOW 0x26 +#define ADAU1701_DACSET 0x27 + +#define ADAU1701_NUM_REGS 0x28 + +#define ADAU1701_DSPCTRL_CR (1 << 2) +#define ADAU1701_DSPCTRL_DAM (1 << 3) +#define ADAU1701_DSPCTRL_ADM (1 << 4) +#define ADAU1701_DSPCTRL_SR_48 0x00 +#define ADAU1701_DSPCTRL_SR_96 0x01 +#define ADAU1701_DSPCTRL_SR_192 0x02 +#define ADAU1701_DSPCTRL_SR_MASK 0x03 + +#define ADAU1701_SEROCTL_INV_LRCLK 0x2000 +#define ADAU1701_SEROCTL_INV_BCLK 0x1000 +#define ADAU1701_SEROCTL_MASTER 0x0800 + +#define ADAU1701_SEROCTL_OBF16 0x0000 +#define ADAU1701_SEROCTL_OBF8 0x0200 +#define ADAU1701_SEROCTL_OBF4 0x0400 +#define ADAU1701_SEROCTL_OBF2 0x0600 +#define ADAU1701_SEROCTL_OBF_MASK 0x0600 + +#define ADAU1701_SEROCTL_OLF1024 0x0000 +#define ADAU1701_SEROCTL_OLF512 0x0080 +#define ADAU1701_SEROCTL_OLF256 0x0100 +#define ADAU1701_SEROCTL_OLF_MASK 0x0180 + +#define ADAU1701_SEROCTL_MSB_DEALY1 0x0000 +#define ADAU1701_SEROCTL_MSB_DEALY0 0x0004 +#define ADAU1701_SEROCTL_MSB_DEALY8 0x0008 +#define ADAU1701_SEROCTL_MSB_DEALY12 0x000c +#define ADAU1701_SEROCTL_MSB_DEALY16 0x0010 +#define ADAU1701_SEROCTL_MSB_DEALY_MASK 0x001c + +#define ADAU1701_SEROCTL_WORD_LEN_24 0x0000 +#define ADAU1701_SEROCTL_WORD_LEN_20 0x0001 +#define ADAU1701_SEROCTL_WORD_LEN_16 0x0010 +#define ADAU1701_SEROCTL_WORD_LEN_MASK 0x0003 + +#define ADAU1701_AUXNPOW_VBPD 0x40 +#define ADAU1701_AUXNPOW_VRPD 0x20 + +#define ADAU1701_SERICTL_I2S 0 +#define ADAU1701_SERICTL_LEFTJ 1 +#define ADAU1701_SERICTL_TDM 2 +#define ADAU1701_SERICTL_RIGHTJ_24 3 +#define ADAU1701_SERICTL_RIGHTJ_20 4 +#define ADAU1701_SERICTL_RIGHTJ_18 5 +#define ADAU1701_SERICTL_RIGHTJ_16 6 +#define ADAU1701_SERICTL_MODE_MASK 7 +#define ADAU1701_SERICTL_INV_BCLK BIT(3) +#define ADAU1701_SERICTL_INV_LRCLK BIT(4) + +#define ADAU1701_OSCIPOW_OPD 0x04 +#define ADAU1701_DACSET_DACINIT 1 + +#define ADAU1701_FIRMWARE "adau1701.bin" + +struct adau1701 { + unsigned int dai_fmt; +}; + +static const struct snd_kcontrol_new adau1701_controls[] = { + SOC_SINGLE("Master Capture Switch", ADAU1701_DSPCTRL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau1701_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC0", "Playback", ADAU1701_AUXNPOW, 3, 1), + SND_SOC_DAPM_DAC("DAC1", "Playback", ADAU1701_AUXNPOW, 2, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", ADAU1701_AUXNPOW, 1, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", ADAU1701_AUXNPOW, 0, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", ADAU1701_AUXNPOW, 7, 1), + + SND_SOC_DAPM_OUTPUT("OUT0"), + SND_SOC_DAPM_OUTPUT("OUT1"), + SND_SOC_DAPM_OUTPUT("OUT2"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_INPUT("IN0"), + SND_SOC_DAPM_INPUT("IN1"), +}; + +static const struct snd_soc_dapm_route adau1701_dapm_routes[] = { + { "OUT0", NULL, "DAC0" }, + { "OUT1", NULL, "DAC1" }, + { "OUT2", NULL, "DAC2" }, + { "OUT3", NULL, "DAC3" }, + + { "ADC", NULL, "IN0" }, + { "ADC", NULL, "IN1" }, +}; + +static unsigned int adau1701_register_size(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ADAU1701_DSPCTRL: + case ADAU1701_SEROCTL: + case ADAU1701_AUXNPOW: + case ADAU1701_OSCIPOW: + case ADAU1701_DACSET: + return 2; + case ADAU1701_SERICTL: + return 1; + } + + dev_err(codec->dev, "Unsupported register address: %d\n", reg); + return 0; +} + +static int adau1701_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + unsigned int i; + unsigned int size; + uint8_t buf[4]; + int ret; + + size = adau1701_register_size(codec, reg); + if (size == 0) + return -EINVAL; + + snd_soc_cache_write(codec, reg, value); + + buf[0] = 0x08; + buf[1] = reg; + + for (i = size + 1; i >= 2; --i) { + buf[i] = value; + value >>= 8; + } + + ret = i2c_master_send(to_i2c_client(codec->dev), buf, size + 2); + if (ret == size + 2) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int adau1701_read(struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value; + unsigned int ret; + + ret = snd_soc_cache_read(codec, reg, &value); + if (ret) + return ret; + + return value; +} + +static int adau1701_load_firmware(struct snd_soc_codec *codec) +{ + return process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE); +} + +static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = ADAU1701_SEROCTL_WORD_LEN_MASK; + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SEROCTL_WORD_LEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SEROCTL_WORD_LEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SEROCTL_WORD_LEN_24; + break; + default: + return -EINVAL; + } + + if (adau1701->dai_fmt == SND_SOC_DAIFMT_RIGHT_J) { + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val |= ADAU1701_SEROCTL_MSB_DEALY12; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val |= ADAU1701_SEROCTL_MSB_DEALY8; + break; + } + mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; + } + + snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + + return 0; +} + +static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, + snd_pcm_format_t format) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adau1701->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAU1701_SERICTL_RIGHTJ_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAU1701_SERICTL_RIGHTJ_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAU1701_SERICTL_RIGHTJ_24; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_SERICTL, + ADAU1701_SERICTL_MODE_MASK, val); + + return 0; +} + +static int adau1701_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + snd_pcm_format_t format; + unsigned int val; + + switch (params_rate(params)) { + case 192000: + val = ADAU1701_DSPCTRL_SR_192; + break; + case 96000: + val = ADAU1701_DSPCTRL_SR_96; + break; + case 48000: + val = ADAU1701_DSPCTRL_SR_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + ADAU1701_DSPCTRL_SR_MASK, val); + + format = params_format(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return adau1701_set_playback_pcm_format(codec, format); + else + return adau1701_set_capture_pcm_format(codec, format); +} + +static int adau1701_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + unsigned int serictl = 0x00, seroctl = 0x00; + bool invert_lrclk; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* master, 64-bits per sample, 1 frame per sample */ + seroctl |= ADAU1701_SEROCTL_MASTER | ADAU1701_SEROCTL_OBF16 + | ADAU1701_SEROCTL_OLF1024; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_NF: + invert_lrclk = false; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert_lrclk = true; + serictl |= ADAU1701_SERICTL_INV_BCLK; + seroctl |= ADAU1701_SEROCTL_INV_BCLK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + serictl |= ADAU1701_SERICTL_LEFTJ; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY0; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serictl |= ADAU1701_SERICTL_RIGHTJ_24; + seroctl |= ADAU1701_SEROCTL_MSB_DEALY8; + invert_lrclk = !invert_lrclk; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) { + seroctl |= ADAU1701_SEROCTL_INV_LRCLK; + serictl |= ADAU1701_SERICTL_INV_LRCLK; + } + + adau1701->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + snd_soc_write(codec, ADAU1701_SERICTL, serictl); + snd_soc_update_bits(codec, ADAU1701_SEROCTL, + ~ADAU1701_SEROCTL_WORD_LEN_MASK, seroctl); + + return 0; +} + +static int adau1701_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* Enable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + /* Disable VREF and VREF buffer */ + snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int mask = ADAU1701_DSPCTRL_DAM; + unsigned int val; + + if (mute) + val = 0; + else + val = mask; + + snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + + return 0; +} + +static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + unsigned int val; + + switch (clk_id) { + case ADAU1701_CLK_SRC_OSC: + val = 0x0; + break; + case ADAU1701_CLK_SRC_MCLK: + val = ADAU1701_OSCIPOW_OPD; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + + return 0; +} + +#define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) + +#define ADAU1701_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops adau1701_dai_ops = { + .set_fmt = adau1701_set_dai_fmt, + .hw_params = adau1701_hw_params, + .digital_mute = adau1701_digital_mute, +}; + +static struct snd_soc_dai_driver adau1701_dai = { + .name = "adau1701", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 8, + .rates = ADAU1701_RATES, + .formats = ADAU1701_FORMATS, + }, + .ops = &adau1701_dai_ops, + .symmetric_rates = 1, +}; + +static int adau1701_probe(struct snd_soc_codec *codec) +{ + int ret; + + codec->dapm.idle_bias_off = 1; + + ret = adau1701_load_firmware(codec); + if (ret) + dev_warn(codec->dev, "Failed to load firmware\n"); + + snd_soc_write(codec, ADAU1701_DACSET, ADAU1701_DACSET_DACINIT); + snd_soc_write(codec, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_CR); + + return 0; +} + +static struct snd_soc_codec_driver adau1701_codec_drv = { + .probe = adau1701_probe, + .set_bias_level = adau1701_set_bias_level, + + .reg_cache_size = ADAU1701_NUM_REGS, + .reg_word_size = sizeof(u16), + + .controls = adau1701_controls, + .num_controls = ARRAY_SIZE(adau1701_controls), + .dapm_widgets = adau1701_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1701_dapm_widgets), + .dapm_routes = adau1701_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1701_dapm_routes), + + .write = adau1701_write, + .read = adau1701_read, + + .set_sysclk = adau1701_set_sysclk, +}; + +static __devinit int adau1701_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1701 *adau1701; + int ret; + + adau1701 = kzalloc(sizeof(*adau1701), GFP_KERNEL); + if (!adau1701) + return -ENOMEM; + + i2c_set_clientdata(client, adau1701); + ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, + &adau1701_dai, 1); + if (ret < 0) + kfree(adau1701); + + return ret; +} + +static __devexit int adau1701_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id adau1701_i2c_id[] = { + { "adau1701", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); + +static struct i2c_driver adau1701_i2c_driver = { + .driver = { + .name = "adau1701", + .owner = THIS_MODULE, + }, + .probe = adau1701_i2c_probe, + .remove = __devexit_p(adau1701_i2c_remove), + .id_table = adau1701_i2c_id, +}; + +static int __init adau1701_init(void) +{ + return i2c_add_driver(&adau1701_i2c_driver); +} +module_init(adau1701_init); + +static void __exit adau1701_exit(void) +{ + i2c_del_driver(&adau1701_i2c_driver); +} +module_exit(adau1701_exit); + +MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver"); +MODULE_AUTHOR("Cliff Cai <cliff.cai@analog.com>"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1701.h b/sound/soc/codecs/adau1701.h new file mode 100644 index 00000000000..8d0949a2aec --- /dev/null +++ b/sound/soc/codecs/adau1701.h @@ -0,0 +1,17 @@ +/* + * header file for ADAU1701 SigmaDSP processor + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAU1701_H +#define _ADAU1701_H + +enum adau1701_clk_src { + ADAU1701_CLK_SRC_OSC, + ADAU1701_CLK_SRC_MCLK, +}; + +#endif diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c new file mode 100644 index 00000000000..e30fba62392 --- /dev/null +++ b/sound/soc/codecs/adav80x.c @@ -0,0 +1,951 @@ +/* + * ADAV80X Audio Codec driver supporting ADAV801, ADAV803 + * + * Copyright 2011 Analog Devices Inc. + * Author: Yi Li <yi.li@analog.com> + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> + +#include "adav80x.h" + +#define ADAV80X_PLAYBACK_CTRL 0x04 +#define ADAV80X_AUX_IN_CTRL 0x05 +#define ADAV80X_REC_CTRL 0x06 +#define ADAV80X_AUX_OUT_CTRL 0x07 +#define ADAV80X_DPATH_CTRL1 0x62 +#define ADAV80X_DPATH_CTRL2 0x63 +#define ADAV80X_DAC_CTRL1 0x64 +#define ADAV80X_DAC_CTRL2 0x65 +#define ADAV80X_DAC_CTRL3 0x66 +#define ADAV80X_DAC_L_VOL 0x68 +#define ADAV80X_DAC_R_VOL 0x69 +#define ADAV80X_PGA_L_VOL 0x6c +#define ADAV80X_PGA_R_VOL 0x6d +#define ADAV80X_ADC_CTRL1 0x6e +#define ADAV80X_ADC_CTRL2 0x6f +#define ADAV80X_ADC_L_VOL 0x70 +#define ADAV80X_ADC_R_VOL 0x71 +#define ADAV80X_PLL_CTRL1 0x74 +#define ADAV80X_PLL_CTRL2 0x75 +#define ADAV80X_ICLK_CTRL1 0x76 +#define ADAV80X_ICLK_CTRL2 0x77 +#define ADAV80X_PLL_CLK_SRC 0x78 +#define ADAV80X_PLL_OUTE 0x7a + +#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00 +#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll)) +#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll)) + +#define ADAV80X_ICLK_CTRL1_DAC_SRC(src) ((src) << 5) +#define ADAV80X_ICLK_CTRL1_ADC_SRC(src) ((src) << 2) +#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src) (src) +#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src) ((src) << 3) + +#define ADAV80X_PLL_CTRL1_PLLDIV 0x10 +#define ADAV80X_PLL_CTRL1_PLLPD(pll) (0x04 << (pll)) +#define ADAV80X_PLL_CTRL1_XTLPD 0x02 + +#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4)) + +#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00) +#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08) +#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c) + +#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02) +#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01) +#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f) + +#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80 +#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00 +#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80 + +#define ADAV80X_DAC_CTRL1_PD 0x80 + +#define ADAV80X_DAC_CTRL2_DIV1 0x00 +#define ADAV80X_DAC_CTRL2_DIV1_5 0x10 +#define ADAV80X_DAC_CTRL2_DIV2 0x20 +#define ADAV80X_DAC_CTRL2_DIV3 0x30 +#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30 + +#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00 +#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40 +#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80 +#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0 + +#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00 +#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01 +#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02 +#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03 +#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01 + +#define ADAV80X_CAPTURE_MODE_MASTER 0x20 +#define ADAV80X_CAPTURE_WORD_LEN24 0x00 +#define ADAV80X_CAPTURE_WORD_LEN20 0x04 +#define ADAV80X_CAPTRUE_WORD_LEN18 0x08 +#define ADAV80X_CAPTURE_WORD_LEN16 0x0c +#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c + +#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00 +#define ADAV80X_CAPTURE_MODE_I2S 0x01 +#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03 +#define ADAV80X_CAPTURE_MODE_MASK 0x03 + +#define ADAV80X_PLAYBACK_MODE_MASTER 0x10 +#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00 +#define ADAV80X_PLAYBACK_MODE_I2S 0x01 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07 +#define ADAV80X_PLAYBACK_MODE_MASK 0x07 + +#define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) + +static u8 adav80x_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, + 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, + 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, + 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, + 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, + 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, + 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +}; + +struct adav80x { + enum snd_soc_control_type control_type; + + enum adav80x_clk_src clk_src; + unsigned int sysclk; + enum adav80x_pll_src pll_src; + + unsigned int dai_fmt[2]; + unsigned int rate; + bool deemph; + bool sysclk_pd[3]; +}; + +static const char *adav80x_mux_text[] = { + "ADC", + "Playback", + "Aux Playback", +}; + +static const unsigned int adav80x_mux_values[] = { + 0, 2, 3, +}; + +#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \ + ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \ + adav80x_mux_values) + +static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0); +static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3); +static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3); + +static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum); +static const struct snd_kcontrol_new adav80x_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum); +static const struct snd_kcontrol_new adav80x_dac_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum); + +#define ADAV80X_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1), + SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1), + + SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0), + + ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl), + ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl), + ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl), + + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + + SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0), +}; + +static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + const char *clk; + + switch (adav80x->clk_src) { + case ADAV80X_CLK_PLL1: + clk = "PLL1"; + break; + case ADAV80X_CLK_PLL2: + clk = "PLL2"; + break; + case ADAV80X_CLK_XTAL: + clk = "OSC"; + break; + default: + return 0; + } + + return strcmp(source->name, clk) == 0; +} + +static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; +} + + +static const struct snd_soc_dapm_route adav80x_dapm_routes[] = { + { "DAC Select", "ADC", "ADC" }, + { "DAC Select", "Playback", "AIFIN" }, + { "DAC Select", "Aux Playback", "AIFAUXIN" }, + { "DAC", NULL, "DAC Select" }, + + { "Capture Select", "ADC", "ADC" }, + { "Capture Select", "Playback", "AIFIN" }, + { "Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFOUT", NULL, "Capture Select" }, + + { "Aux Capture Select", "ADC", "ADC" }, + { "Aux Capture Select", "Playback", "AIFIN" }, + { "Aux Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFAUXOUT", NULL, "Aux Capture Select" }, + + { "VOUTR", NULL, "DAC" }, + { "VOUTL", NULL, "DAC" }, + + { "Left PGA", NULL, "VINL" }, + { "Right PGA", NULL, "VINR" }, + { "ADC", NULL, "Left PGA" }, + { "ADC", NULL, "Right PGA" }, + + { "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check }, + { "PLL1", NULL, "OSC", adav80x_dapm_pll_check }, + { "PLL2", NULL, "OSC", adav80x_dapm_pll_check }, + + { "ADC", NULL, "SYSCLK" }, + { "DAC", NULL, "SYSCLK" }, + { "AIFOUT", NULL, "SYSCLK" }, + { "AIFAUXOUT", NULL, "SYSCLK" }, + { "AIFIN", NULL, "SYSCLK" }, + { "AIFAUXIN", NULL, "SYSCLK" }, +}; + +static int adav80x_set_deemph(struct snd_soc_codec *codec) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->deemph) { + switch (adav80x->rate) { + case 32000: + val = ADAV80X_DAC_CTRL2_DEEMPH_32; + break; + case 44100: + val = ADAV80X_DAC_CTRL2_DEEMPH_44; + break; + case 48000: + case 64000: + case 88200: + case 96000: + val = ADAV80X_DAC_CTRL2_DEEMPH_48; + break; + default: + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + break; + } + } else { + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + } + + return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); +} + +static int adav80x_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + adav80x->deemph = deemph; + + return adav80x_set_deemph(codec); +} + +static int adav80x_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = adav80x->deemph; + return 0; +}; + +static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adav80x_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL, + ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL, + ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL, + ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv), + + SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0), + SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0, + adav80x_get_deemph, adav80x_put_deemph), +}; + +static unsigned int adav80x_port_ctrl_regs[2][2] = { + { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, }, + { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL }, +}; + +static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int capture = 0x00; + unsigned int playback = 0x00; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + capture |= ADAV80X_CAPTURE_MODE_MASTER; + playback |= ADAV80X_PLAYBACK_MODE_MASTER; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + capture |= ADAV80X_CAPTURE_MODE_I2S; + playback |= ADAV80X_PLAYBACK_MODE_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + capture |= ADAV80X_CAPTURE_MODE_LEFT_J; + playback |= ADAV80X_PLAYBACK_MODE_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + capture |= ADAV80X_CAPTURE_MODE_RIGHT_J; + playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, + capture); + snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + + adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adav80x_set_adc_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_ADC_CTRL1_MODULATOR_128FS; + else + val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; + + snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); + + return 0; +} + +static int adav80x_set_dac_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS; + else + val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; + + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, + val); + + return 0; +} + +static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_CAPTURE_WORD_LEN16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_CAPTRUE_WORD_LEN18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_CAPTURE_WORD_LEN20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_CAPTURE_WORD_LEN24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_WORD_LEN_MASK, val); + + return 0; +} + +static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + ADAV80X_PLAYBACK_MODE_MASK, val); + + return 0; +} + +static int adav80x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + + if (rate * 256 != adav80x->sysclk) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + adav80x_set_playback_pcm_format(codec, dai, + params_format(params)); + adav80x_set_dac_clock(codec, rate); + } else { + adav80x_set_capture_pcm_format(codec, dai, + params_format(params)); + adav80x_set_adc_clock(codec, rate); + } + adav80x->rate = rate; + adav80x_set_deemph(codec); + + return 0; +} + +static int adav80x_set_sysclk(struct snd_soc_codec *codec, + int clk_id, unsigned int freq, int dir) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (dir == SND_SOC_CLOCK_IN) { + switch (clk_id) { + case ADAV80X_CLK_XIN: + case ADAV80X_CLK_XTAL: + case ADAV80X_CLK_MCLKI: + case ADAV80X_CLK_PLL1: + case ADAV80X_CLK_PLL2: + break; + default: + return -EINVAL; + } + + adav80x->sysclk = freq; + + if (adav80x->clk_src != clk_id) { + unsigned int iclk_ctrl1, iclk_ctrl2; + + adav80x->clk_src = clk_id; + if (clk_id == ADAV80X_CLK_XTAL) + clk_id = ADAV80X_CLK_XIN; + + iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); + iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); + + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + + snd_soc_dapm_sync(&codec->dapm); + } + } else { + unsigned int mask; + + switch (clk_id) { + case ADAV80X_CLK_SYSCLK1: + case ADAV80X_CLK_SYSCLK2: + case ADAV80X_CLK_SYSCLK3: + break; + default: + return -EINVAL; + } + + clk_id -= ADAV80X_CLK_SYSCLK1; + mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id); + + if (freq == 0) { + snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask); + adav80x->sysclk_pd[clk_id] = true; + } else { + snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0); + adav80x->sysclk_pd[clk_id] = false; + } + + if (adav80x->sysclk_pd[0]) + snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + else + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + + if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2]) + snd_soc_dapm_disable_pin(&codec->dapm, "PLL2"); + else + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int pll_ctrl1 = 0; + unsigned int pll_ctrl2 = 0; + unsigned int pll_src; + + switch (source) { + case ADAV80X_PLL_SRC_XTAL: + case ADAV80X_PLL_SRC_XIN: + case ADAV80X_PLL_SRC_MCLKI: + break; + default: + return -EINVAL; + } + + if (!freq_out) + return 0; + + switch (freq_in) { + case 27000000: + break; + case 54000000: + if (source == ADAV80X_PLL_SRC_XIN) { + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; + break; + } + default: + return -EINVAL; + } + + if (freq_out > 12288000) { + pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id); + freq_out /= 2; + } + + /* freq_out = sample_rate * 256 */ + switch (freq_out) { + case 8192000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id); + break; + case 11289600: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id); + break; + case 12288000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, + pll_ctrl1); + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); + + if (source != adav80x->pll_src) { + if (source == ADAV80X_PLL_SRC_MCLKI) + pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id); + else + pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); + + snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); + + adav80x->pll_src = source; + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAV80X_DAC_CTRL1_PD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +/* Enforce the same sample rate on all audio interfaces */ +static int adav80x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active || !adav80x->rate) + return 0; + + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate); +} + +static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active) + adav80x->rate = 0; +} + +static const struct snd_soc_dai_ops adav80x_dai_ops = { + .set_fmt = adav80x_set_dai_fmt, + .hw_params = adav80x_hw_params, + .startup = adav80x_dai_startup, + .shutdown = adav80x_dai_shutdown, +}; + +#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver adav80x_dais[] = { + { + .name = "adav80x-hifi", + .id = 0, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, + { + .name = "adav80x-aux", + .id = 1, + .playback = { + .stream_name = "Aux Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "Aux Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, +}; + +static int adav80x_probe(struct snd_soc_codec *codec) +{ + int ret; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Force PLLs on for SYSCLK output */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); + + /* Power down S/PDIF receiver, since it is currently not supported */ + snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20); + /* Disable DAC zero flag */ + snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adav80x_resume(struct snd_soc_codec *codec) +{ + adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->cache_sync = 1; + snd_soc_cache_sync(codec); + + return 0; +} + +static int adav80x_remove(struct snd_soc_codec *codec) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static struct snd_soc_codec_driver adav80x_codec_driver = { + .probe = adav80x_probe, + .remove = adav80x_remove, + .suspend = adav80x_suspend, + .resume = adav80x_resume, + .set_bias_level = adav80x_set_bias_level, + + .set_pll = adav80x_set_pll, + .set_sysclk = adav80x_set_sysclk, + + .reg_word_size = sizeof(u8), + .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), + .reg_cache_default = adav80x_default_regs, + + .controls = adav80x_controls, + .num_controls = ARRAY_SIZE(adav80x_controls), + .dapm_widgets = adav80x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets), + .dapm_routes = adav80x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), +}; + +static int __devinit adav80x_bus_probe(struct device *dev, + enum snd_soc_control_type control_type) +{ + struct adav80x *adav80x; + int ret; + + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + dev_set_drvdata(dev, adav80x); + adav80x->control_type = control_type; + + ret = snd_soc_register_codec(dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); + if (ret) + kfree(adav80x); + + return ret; +} + +static int __devexit adav80x_bus_remove(struct device *dev) +{ + snd_soc_unregister_codec(dev); + kfree(dev_get_drvdata(dev)); + return 0; +} + +#if defined(CONFIG_SPI_MASTER) +static int __devinit adav80x_spi_probe(struct spi_device *spi) +{ + return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); +} + +static int __devexit adav80x_spi_remove(struct spi_device *spi) +{ + return adav80x_bus_remove(&spi->dev); +} + +static struct spi_driver adav80x_spi_driver = { + .driver = { + .name = "adav801", + .owner = THIS_MODULE, + }, + .probe = adav80x_spi_probe, + .remove = __devexit_p(adav80x_spi_remove), +}; +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id adav80x_id[] = { + { "adav803", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adav80x_id); + +static int __devinit adav80x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return adav80x_bus_probe(&client->dev, SND_SOC_I2C); +} + +static int __devexit adav80x_i2c_remove(struct i2c_client *client) +{ + return adav80x_bus_remove(&client->dev); +} + +static struct i2c_driver adav80x_i2c_driver = { + .driver = { + .name = "adav803", + .owner = THIS_MODULE, + }, + .probe = adav80x_i2c_probe, + .remove = __devexit_p(adav80x_i2c_remove), + .id_table = adav80x_id, +}; +#endif + +static int __init adav80x_init(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&adav80x_i2c_driver); + if (ret) + return ret; +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&adav80x_spi_driver); +#endif + + return ret; +} +module_init(adav80x_init); + +static void __exit adav80x_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&adav80x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&adav80x_spi_driver); +#endif +} +module_exit(adav80x_exit); + +MODULE_DESCRIPTION("ASoC ADAV80x driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_AUTHOR("Yi Li <yi.li@analog.com>>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h new file mode 100644 index 00000000000..adb0fc76d4e --- /dev/null +++ b/sound/soc/codecs/adav80x.h @@ -0,0 +1,35 @@ +/* + * header file for ADAV80X parts + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAV80X_H +#define _ADAV80X_H + +enum adav80x_pll_src { + ADAV80X_PLL_SRC_XIN, + ADAV80X_PLL_SRC_XTAL, + ADAV80X_PLL_SRC_MCLKI, +}; + +enum adav80x_pll { + ADAV80X_PLL1 = 0, + ADAV80X_PLL2 = 1, +}; + +enum adav80x_clk_src { + ADAV80X_CLK_XIN = 0, + ADAV80X_CLK_MCLKI = 1, + ADAV80X_CLK_PLL1 = 2, + ADAV80X_CLK_PLL2 = 3, + ADAV80X_CLK_XTAL = 6, + + ADAV80X_CLK_SYSCLK1 = 6, + ADAV80X_CLK_SYSCLK2 = 7, + ADAV80X_CLK_SYSCLK3 = 8, +}; + +#endif diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index ed96f247c2d..7a64e58cddc 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -457,7 +457,7 @@ static struct snd_soc_dai_ops ak4641_pcm_dai_ops = { .set_sysclk = ak4641_set_dai_sysclk, }; -struct snd_soc_dai_driver ak4641_dai[] = { +static struct snd_soc_dai_driver ak4641_dai[] = { { .name = "ak4641-hifi", .id = 1, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0206a17d728..6cc8678f49f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -636,10 +636,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) #endif /* CONFIG_PM */ /* - * ASoC codec device structure - * - * Assign this variable to the codec_dev field of the machine driver's - * snd_soc_device structure. + * ASoC codec driver structure */ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .probe = cs4270_probe, diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4173b67c94d..ac65a2d3640 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1397,8 +1397,6 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, if (freq == max98088->sysclk) return 0; - max98088->sysclk = freq; /* remember current sysclk */ - /* Setup clocks for slave mode, and using the PLL * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) * 0x02 (when master clk is 20MHz to 30MHz).. diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index e1d282d477d..668434d4430 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1517,8 +1517,6 @@ static int max98095_dai_set_sysclk(struct snd_soc_dai *dai, if (freq == max98095->sysclk) return 0; - max98095->sysclk = freq; /* remember current sysclk */ - /* Setup clocks for slave mode, and using the PLL * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) * 0x02 (when master clk is 20MHz to 40MHz).. @@ -2261,11 +2259,11 @@ static int max98095_probe(struct snd_soc_codec *codec) ret = snd_soc_read(codec, M98095_0FF_REV_ID); if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", + dev_err(codec->dev, "Failure reading hardware revision: %d\n", ret); goto err_access; } - dev_info(codec->dev, "revision %c\n", ret + 'A'); + dev_info(codec->dev, "Hardware revision: %c\n", ret - 0x40 + 'A'); snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV); @@ -2342,8 +2340,8 @@ static int max98095_i2c_probe(struct i2c_client *i2c, max98095->control_data = i2c; max98095->pdata = i2c->dev.platform_data; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_max98095, &max98095_dai[0], 3); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095, + max98095_dai, ARRAY_SIZE(max98095_dai)); if (ret < 0) kfree(max98095); return ret; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c new file mode 100644 index 00000000000..409d89d1f34 --- /dev/null +++ b/sound/soc/codecs/sta32x.c @@ -0,0 +1,917 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur@freescale.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. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#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/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "sta32x.h" + +#define STA32X_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA32X_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = { + 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60, + 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69, + 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d, + 0xc0, 0xf3, 0x33, 0x00, 0x0c, +}; + +/* regulator power supply names */ +static const char *sta32x_supply_names[] = { + "Vdda", /* analog supply, 3.3VV */ + "Vdd3", /* digital supply, 3.3V */ + "Vcc" /* power amp spply, 10V - 36V */ +}; + +/* codec private data */ +struct sta32x_priv { + struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; + struct snd_soc_codec *codec; + + unsigned int mclk; + unsigned int format; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -120, 200, 0); + +static const char *sta32x_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" }; +static const char *sta32x_auto_eq_mode[] = { + "User", "Preset", "Loudness" }; +static const char *sta32x_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" }; +static const char *sta32x_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", "200Hz", + "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", "340Hz", "360Hz" }; +static const char *sta32x_preset_eq_mode[] = { + "Flat", "Rock", "Soft Rock", "Jazz", "Classical", "Dance", "Pop", "Soft", + "Hard", "Party", "Vocal", "Hip-Hop", "Dialog", "Bass-boost #1", + "Bass-boost #2", "Bass-boost #3", "Loudness 1", "Loudness 2", + "Loudness 3", "Loudness 4", "Loudness 5", "Loudness 6", "Loudness 7", + "Loudness 8", "Loudness 9", "Loudness 10", "Loudness 11", "Loudness 12", + "Loudness 13", "Loudness 14", "Loudness 15", "Loudness 16" }; +static const char *sta32x_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" }; +static const char *sta32x_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" }; +static const char *sta32x_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" }; + +static const unsigned int sta32x_limiter_ac_attack_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +}; + +static const unsigned int sta32x_limiter_ac_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const unsigned int sta32x_limiter_drc_attack_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +}; + +static const unsigned int sta32x_limiter_drc_release_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +}; + +static const struct soc_enum sta32x_drc_ac_enum = + SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + 2, sta32x_drc_ac); +static const struct soc_enum sta32x_auto_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + 3, sta32x_auto_eq_mode); +static const struct soc_enum sta32x_auto_gc_enum = + SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + 4, sta32x_auto_gc_mode); +static const struct soc_enum sta32x_auto_xo_enum = + SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + 16, sta32x_auto_xo_mode); +static const struct soc_enum sta32x_preset_eq_enum = + SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + 32, sta32x_preset_eq_mode); +static const struct soc_enum sta32x_limiter_ch1_enum = + SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch2_enum = + SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter_ch3_enum = + SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + 3, sta32x_limiter_select); +static const struct soc_enum sta32x_limiter1_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter2_attack_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, + 16, sta32x_limiter_attack_rate); +static const struct soc_enum sta32x_limiter1_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); +static const struct soc_enum sta32x_limiter2_release_rate_enum = + SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, + 16, sta32x_limiter_release_rate); + +/* byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA32x data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta32x_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta32x_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA32X_CFUD */ + cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0; + /* chip documentation does not say if the bits are self clearing, + * so do it explicitly */ + snd_soc_write(codec, STA32X_CFUD, cfud); + + snd_soc_write(codec, STA32X_CFADDR2, index); + if (numcoef == 1) + snd_soc_write(codec, STA32X_CFUD, cfud | 0x04); + else if (numcoef == 5) + snd_soc_write(codec, STA32X_CFUD, cfud | 0x08); + else + return -EINVAL; + for (i = 0; i < 3 * numcoef; i++) + ucontrol->value.bytes.data[i] = + snd_soc_read(codec, STA32X_B1CF1 + i); + + return 0; +} + +static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA32X_CFUD */ + cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0; + /* chip documentation does not say if the bits are self clearing, + * so do it explicitly */ + snd_soc_write(codec, STA32X_CFUD, cfud); + + snd_soc_write(codec, STA32X_CFADDR2, index); + for (i = 0; i < 3 * numcoef; i++) + snd_soc_write(codec, STA32X_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + snd_soc_write(codec, STA32X_CFUD, cfud | 0x01); + else if (numcoef == 5) + snd_soc_write(codec, STA32X_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta32x_coefficient_info, \ + .get = sta32x_coefficient_get,\ + .put = sta32x_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta32x_coefficient_info, \ + .get = sta32x_coefficient_get,\ + .put = sta32x_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta32x_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA32X_MVOL, 0, 0xff, 1, mvol_tlv), +SOC_SINGLE("Master Switch", STA32X_MMUTE, 0, 1, 1), +SOC_SINGLE("Ch1 Switch", STA32X_MMUTE, 1, 1, 1), +SOC_SINGLE("Ch2 Switch", STA32X_MMUTE, 2, 1, 1), +SOC_SINGLE("Ch3 Switch", STA32X_MMUTE, 3, 1, 1), +SOC_SINGLE_TLV("Ch1 Volume", STA32X_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA32X_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA32X_C3VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE("De-emphasis Filter Switch", STA32X_CONFD, STA32X_CONFD_DEMP_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta32x_drc_ac_enum), +SOC_SINGLE("Miami Mode Switch", STA32X_CONFD, STA32X_CONFD_MME_SHIFT, 1, 0), +SOC_SINGLE("Zero Cross Switch", STA32X_CONFE, STA32X_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA32X_CONFE, STA32X_CONFE_SVE_SHIFT, 1, 0), +SOC_SINGLE("Auto-Mute Switch", STA32X_CONFF, STA32X_CONFF_IDE_SHIFT, 1, 0), +SOC_ENUM("Automode EQ", sta32x_auto_eq_enum), +SOC_ENUM("Automode GC", sta32x_auto_gc_enum), +SOC_ENUM("Automode XO", sta32x_auto_xo_enum), +SOC_ENUM("Preset EQ", sta32x_preset_eq_enum), +SOC_SINGLE("Ch1 Tone Control Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", STA32X_C2CFG, STA32X_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", STA32X_C1CFG, STA32X_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Limiter Select", sta32x_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta32x_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta32x_limiter_ch3_enum), +SOC_SINGLE_TLV("Bass Tone Control", STA32X_TONE, STA32X_TONE_BTC_SHIFT, 15, 0, tone_tlv), +SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), + +/* depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxA_SHIFT, + 16, 0, sta32x_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", STA32X_L1ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", STA32X_L2ATRT, STA32X_LxR_SHIFT, + 16, 0, sta32x_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta32x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta32x_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static struct { + int ratio; + int mcs; +} mclk_ratios[3][7] = { + { { 768, 0 }, { 512, 1 }, { 384, 2 }, { 256, 3 }, + { 128, 4 }, { 576, 5 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, + { { 384, 2 }, { 256, 3 }, { 192, 4 }, { 128, 5 }, {64, 0 }, { 0, 0 } }, +}; + + +/** + * sta32x_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA32X, based on the mclk_ratios table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + * + * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause + * theoretically possible sample rates to be enabled. Call it again with a + * proper value set one the external clock is set (most probably you would do + * that from a machine's driver 'hw_param' hook. + */ +static int sta32x_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 sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, j, ir, fs; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + + pr_debug("mclk=%u\n", freq); + sta32x->mclk = freq; + + if (sta32x->mclk) { + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + ir = interpolation_ratios[i].ir; + fs = interpolation_ratios[i].fs; + for (j = 0; mclk_ratios[ir][j].ratio; j++) { + if (mclk_ratios[ir][j].ratio * fs == freq) { + rates |= snd_pcm_rate_to_rate_bit(fs); + if (fs < rate_min) + rate_min = fs; + if (fs > rate_max) + rate_max = fs; + } + } + } + /* FIXME: soc should support a rate list */ + rates &= ~SNDRV_PCM_RATE_KNOT; + + if (!rates) { + dev_err(codec->dev, "could not find a valid sample rate\n"); + return -EINVAL; + } + } else { + /* enable all possible rates */ + rates = STA32X_RATES; + rate_min = 32000; + rate_max = 192000; + } + + codec_dai->driver->playback.rates = rates; + codec_dai->driver->playback.rate_min = rate_min; + codec_dai->driver->playback.rate_max = rate_max; + return 0; +} + +/** + * sta32x_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta32x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + u8 confb = snd_soc_read(codec, STA32X_CONFB); + + pr_debug("\n"); + confb &= ~(STA32X_CONFB_C1IM | STA32X_CONFB_C2IM); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta32x->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA32X_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA32X_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_hw_params - program the STA32X with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int rate; + int i, mcs = -1, ir = -1; + u8 confa, confb; + + rate = params_rate(params); + pr_debug("rate: %u\n", rate); + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) + if (interpolation_ratios[i].fs == rate) + ir = interpolation_ratios[i].ir; + if (ir < 0) + return -EINVAL; + for (i = 0; mclk_ratios[ir][i].ratio; i++) + if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) + mcs = mclk_ratios[ir][i].mcs; + if (mcs < 0) + return -EINVAL; + + confa = snd_soc_read(codec, STA32X_CONFA); + confa &= ~(STA32X_CONFA_MCS_MASK | STA32X_CONFA_IR_MASK); + confa |= (ir << STA32X_CONFA_IR_SHIFT) | (mcs << STA32X_CONFA_MCS_SHIFT); + + confb = snd_soc_read(codec, STA32X_CONFB); + confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + case SNDRV_PCM_FORMAT_S24_3LE: + case SNDRV_PCM_FORMAT_S24_3BE: + pr_debug("24bit\n"); + /* fall through */ + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + pr_debug("24bit or 32bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + pr_debug("20bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S18_3BE: + pr_debug("18bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + pr_debug("16bit\n"); + switch (sta32x->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, STA32X_CONFA, confa); + snd_soc_write(codec, STA32X_CONFB, confb); + return 0; +} + +/** + * sta32x_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta32x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + pr_debug("level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + snd_soc_cache_sync(codec); + } + + /* Power up to mute */ + /* FIXME */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_PWDN | STA32X_CONFF_EAPD, + STA32X_CONFF_PWDN); + msleep(300); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops sta32x_dai_ops = { + .hw_params = sta32x_hw_params, + .set_sysclk = sta32x_set_dai_sysclk, + .set_fmt = sta32x_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta32x_dai = { + .name = "STA32X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA32X_RATES, + .formats = STA32X_FORMATS, + }, + .ops = &sta32x_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta32x_resume(struct snd_soc_codec *codec) +{ + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta32x_suspend NULL +#define sta32x_resume NULL +#endif + +static int sta32x_probe(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + sta32x->codec = codec; + + /* 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; + } + + /* Tell ASoC what kind of I/O to use to read the registers. ASoC will + * then do the I2C transactions itself. + */ + 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; + } + + /* read reg reset values into cache */ + for (i = 0; i < STA32X_REGISTER_COUNT; i++) + snd_soc_cache_write(codec, i, sta32x_regs[i]); + + /* preserve reset values of reserved register bits */ + snd_soc_cache_write(codec, STA32X_CONFC, + codec->hw_read(codec, STA32X_CONFC)); + snd_soc_cache_write(codec, STA32X_CONFE, + codec->hw_read(codec, STA32X_CONFE)); + snd_soc_cache_write(codec, STA32X_CONFF, + codec->hw_read(codec, STA32X_CONFF)); + snd_soc_cache_write(codec, STA32X_MMUTE, + codec->hw_read(codec, STA32X_MMUTE)); + snd_soc_cache_write(codec, STA32X_AUTO1, + codec->hw_read(codec, STA32X_AUTO1)); + snd_soc_cache_write(codec, STA32X_AUTO3, + codec->hw_read(codec, STA32X_AUTO3)); + snd_soc_cache_write(codec, STA32X_C3CFG, + codec->hw_read(codec, STA32X_C3CFG)); + + /* FIXME enable thermal warning adjustment and recovery */ + snd_soc_update_bits(codec, STA32X_CONFA, + STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0); + + /* FIXME select 2.1 mode */ + snd_soc_update_bits(codec, STA32X_CONFF, + STA32X_CONFF_OCFG_MASK, + 1 << STA32X_CONFF_OCFG_SHIFT); + + /* FIXME channel to output mapping */ + snd_soc_update_bits(codec, STA32X_C1CFG, + STA32X_CxCFG_OM_MASK, + 0 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C2CFG, + STA32X_CxCFG_OM_MASK, + 1 << STA32X_CxCFG_OM_SHIFT); + snd_soc_update_bits(codec, STA32X_C3CFG, + STA32X_CxCFG_OM_MASK, + 2 << STA32X_CxCFG_OM_SHIFT); + + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; + +err_get: + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); +err: + return ret; +} + +static int sta32x_remove(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + return 0; +} + +static int sta32x_reg_is_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case STA32X_CONFA ... STA32X_L2ATRT: + case STA32X_MPCC1 ... STA32X_FDRC2: + return 0; + } + return 1; +} + +static const struct snd_soc_codec_driver sta32x_codec = { + .probe = sta32x_probe, + .remove = sta32x_remove, + .suspend = sta32x_suspend, + .resume = sta32x_resume, + .reg_cache_size = STA32X_REGISTER_COUNT, + .reg_word_size = sizeof(u8), + .volatile_register = sta32x_reg_is_volatile, + .set_bias_level = sta32x_set_bias_level, + .controls = sta32x_snd_controls, + .num_controls = ARRAY_SIZE(sta32x_snd_controls), + .dapm_widgets = sta32x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta32x_dapm_widgets), + .dapm_routes = sta32x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes), +}; + +static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta32x_priv *sta32x; + int ret; + + sta32x = kzalloc(sizeof(struct sta32x_priv), GFP_KERNEL); + if (!sta32x) + return -ENOMEM; + + i2c_set_clientdata(i2c, sta32x); + + ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + return ret; + } + + return 0; +} + +static __devexit int sta32x_i2c_remove(struct i2c_client *client) +{ + struct sta32x_priv *sta32x = i2c_get_clientdata(client); + struct snd_soc_codec *codec = sta32x->codec; + + if (codec) + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (codec) { + snd_soc_unregister_codec(&client->dev); + snd_soc_codec_set_drvdata(codec, NULL); + } + + kfree(sta32x); + return 0; +} + +static const struct i2c_device_id sta32x_i2c_id[] = { + { "sta326", 0 }, + { "sta328", 0 }, + { "sta329", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); + +static struct i2c_driver sta32x_i2c_driver = { + .driver = { + .name = "sta32x", + .owner = THIS_MODULE, + }, + .probe = sta32x_i2c_probe, + .remove = __devexit_p(sta32x_i2c_remove), + .id_table = sta32x_i2c_id, +}; + +static int __init sta32x_init(void) +{ + return i2c_add_driver(&sta32x_i2c_driver); +} +module_init(sta32x_init); + +static void __exit sta32x_exit(void) +{ + i2c_del_driver(&sta32x_i2c_driver); +} +module_exit(sta32x_exit); + +MODULE_DESCRIPTION("ASoC STA32X driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h new file mode 100644 index 00000000000..b97ee5a7566 --- /dev/null +++ b/sound/soc/codecs/sta32x.h @@ -0,0 +1,210 @@ +/* + * Codec driver for ST STA32x 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on code from: + * Wolfson Microelectronics PLC. + * Mark Brown <broonie@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 _ASOC_STA_32X_H +#define _ASOC_STA_32X_H + +/* STA326 register addresses */ + +#define STA32X_REGISTER_COUNT 0x2d + +#define STA32X_CONFA 0x00 +#define STA32X_CONFB 0x01 +#define STA32X_CONFC 0x02 +#define STA32X_CONFD 0x03 +#define STA32X_CONFE 0x04 +#define STA32X_CONFF 0x05 +#define STA32X_MMUTE 0x06 +#define STA32X_MVOL 0x07 +#define STA32X_C1VOL 0x08 +#define STA32X_C2VOL 0x09 +#define STA32X_C3VOL 0x0a +#define STA32X_AUTO1 0x0b +#define STA32X_AUTO2 0x0c +#define STA32X_AUTO3 0x0d +#define STA32X_C1CFG 0x0e +#define STA32X_C2CFG 0x0f +#define STA32X_C3CFG 0x10 +#define STA32X_TONE 0x11 +#define STA32X_L1AR 0x12 +#define STA32X_L1ATRT 0x13 +#define STA32X_L2AR 0x14 +#define STA32X_L2ATRT 0x15 +#define STA32X_CFADDR2 0x16 +#define STA32X_B1CF1 0x17 +#define STA32X_B1CF2 0x18 +#define STA32X_B1CF3 0x19 +#define STA32X_B2CF1 0x1a +#define STA32X_B2CF2 0x1b +#define STA32X_B2CF3 0x1c +#define STA32X_A1CF1 0x1d +#define STA32X_A1CF2 0x1e +#define STA32X_A1CF3 0x1f +#define STA32X_A2CF1 0x20 +#define STA32X_A2CF2 0x21 +#define STA32X_A2CF3 0x22 +#define STA32X_B0CF1 0x23 +#define STA32X_B0CF2 0x24 +#define STA32X_B0CF3 0x25 +#define STA32X_CFUD 0x26 +#define STA32X_MPCC1 0x27 +#define STA32X_MPCC2 0x28 +/* Reserved 0x29 */ +/* Reserved 0x2a */ +#define STA32X_Reserved 0x2a +#define STA32X_FDRC1 0x2b +#define STA32X_FDRC2 0x2c +/* Reserved 0x2d */ + + +/* STA326 register field definitions */ + +/* 0x00 CONFA */ +#define STA32X_CONFA_MCS_MASK 0x03 +#define STA32X_CONFA_MCS_SHIFT 0 +#define STA32X_CONFA_IR_MASK 0x18 +#define STA32X_CONFA_IR_SHIFT 3 +#define STA32X_CONFA_TWRB 0x20 +#define STA32X_CONFA_TWAB 0x40 +#define STA32X_CONFA_FDRB 0x80 + +/* 0x01 CONFB */ +#define STA32X_CONFB_SAI_MASK 0x0f +#define STA32X_CONFB_SAI_SHIFT 0 +#define STA32X_CONFB_SAIFB 0x10 +#define STA32X_CONFB_DSCKE 0x20 +#define STA32X_CONFB_C1IM 0x40 +#define STA32X_CONFB_C2IM 0x80 + +/* 0x02 CONFC */ +#define STA32X_CONFC_OM_MASK 0x03 +#define STA32X_CONFC_OM_SHIFT 0 +#define STA32X_CONFC_CSZ_MASK 0x7c +#define STA32X_CONFC_CSZ_SHIFT 2 + +/* 0x03 CONFD */ +#define STA32X_CONFD_HPB 0x01 +#define STA32X_CONFD_HPB_SHIFT 0 +#define STA32X_CONFD_DEMP 0x02 +#define STA32X_CONFD_DEMP_SHIFT 1 +#define STA32X_CONFD_DSPB 0x04 +#define STA32X_CONFD_DSPB_SHIFT 2 +#define STA32X_CONFD_PSL 0x08 +#define STA32X_CONFD_PSL_SHIFT 3 +#define STA32X_CONFD_BQL 0x10 +#define STA32X_CONFD_BQL_SHIFT 4 +#define STA32X_CONFD_DRC 0x20 +#define STA32X_CONFD_DRC_SHIFT 5 +#define STA32X_CONFD_ZDE 0x40 +#define STA32X_CONFD_ZDE_SHIFT 6 +#define STA32X_CONFD_MME 0x80 +#define STA32X_CONFD_MME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA32X_CONFE_MPCV 0x01 +#define STA32X_CONFE_MPCV_SHIFT 0 +#define STA32X_CONFE_MPC 0x02 +#define STA32X_CONFE_MPC_SHIFT 1 +#define STA32X_CONFE_AME 0x08 +#define STA32X_CONFE_AME_SHIFT 3 +#define STA32X_CONFE_PWMS 0x10 +#define STA32X_CONFE_PWMS_SHIFT 4 +#define STA32X_CONFE_ZCE 0x40 +#define STA32X_CONFE_ZCE_SHIFT 6 +#define STA32X_CONFE_SVE 0x80 +#define STA32X_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA32X_CONFF_OCFG_MASK 0x03 +#define STA32X_CONFF_OCFG_SHIFT 0 +#define STA32X_CONFF_IDE 0x04 +#define STA32X_CONFF_IDE_SHIFT 3 +#define STA32X_CONFF_BCLE 0x08 +#define STA32X_CONFF_ECLE 0x20 +#define STA32X_CONFF_PWDN 0x40 +#define STA32X_CONFF_EAPD 0x80 + +/* 0x06 MMUTE */ +#define STA32X_MMUTE_MMUTE 0x01 + +/* 0x0b AUTO1 */ +#define STA32X_AUTO1_AMEQ_MASK 0x03 +#define STA32X_AUTO1_AMEQ_SHIFT 0 +#define STA32X_AUTO1_AMV_MASK 0xc0 +#define STA32X_AUTO1_AMV_SHIFT 2 +#define STA32X_AUTO1_AMGC_MASK 0x30 +#define STA32X_AUTO1_AMGC_SHIFT 4 +#define STA32X_AUTO1_AMPS 0x80 + +/* 0x0c AUTO2 */ +#define STA32X_AUTO2_AMAME 0x01 +#define STA32X_AUTO2_AMAM_MASK 0x0e +#define STA32X_AUTO2_AMAM_SHIFT 1 +#define STA32X_AUTO2_XO_MASK 0xf0 +#define STA32X_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA32X_AUTO3_PEQ_MASK 0x1f +#define STA32X_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA32X_CxCFG_TCB 0x01 /* only C1 and C2 */ +#define STA32X_CxCFG_TCB_SHIFT 0 +#define STA32X_CxCFG_EQBP 0x02 /* only C1 and C2 */ +#define STA32X_CxCFG_EQBP_SHIFT 1 +#define STA32X_CxCFG_VBP 0x03 +#define STA32X_CxCFG_VBP_SHIFT 2 +#define STA32X_CxCFG_BO 0x04 +#define STA32X_CxCFG_LS_MASK 0x30 +#define STA32X_CxCFG_LS_SHIFT 4 +#define STA32X_CxCFG_OM_MASK 0xc0 +#define STA32X_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA32X_TONE_BTC_SHIFT 0 +#define STA32X_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA32X_LxA_SHIFT 0 +#define STA32X_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA32X_CFUD_W1 0x01 +#define STA32X_CFUD_WA 0x02 +#define STA32X_CFUD_R1 0x04 +#define STA32X_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA32X_C1_BQ_BASE 0 +#define STA32X_C2_BQ_BASE 20 +#define STA32X_CH_BQ_NUM 4 +#define STA32X_BQ_NUM_COEF 5 +#define STA32X_XO_HP_BQ_BASE 40 +#define STA32X_XO_LP_BQ_BASE 45 +#define STA32X_C1_PRESCALE 50 +#define STA32X_C2_PRESCALE 51 +#define STA32X_C1_POSTSCALE 52 +#define STA32X_C2_POSTSCALE 53 +#define STA32X_C3_POSTSCALE 54 +#define STA32X_TW_POSTSCALE 55 +#define STA32X_C1_MIX1 56 +#define STA32X_C1_MIX2 57 +#define STA32X_C2_MIX1 58 +#define STA32X_C2_MIX2 59 +#define STA32X_C3_MIX1 60 +#define STA32X_C3_MIX2 61 + +#endif /* _ASOC_STA_32X_H */ diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 789453d44ec..0963c4c7a83 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -226,11 +226,13 @@ static const char *aic3x_adc_hpf[] = #define RDAC_ENUM 1 #define LHPCOM_ENUM 2 #define RHPCOM_ENUM 3 -#define LINE1L_ENUM 4 -#define LINE1R_ENUM 5 -#define LINE2L_ENUM 6 -#define LINE2R_ENUM 7 -#define ADC_HPF_ENUM 8 +#define LINE1L_2_L_ENUM 4 +#define LINE1L_2_R_ENUM 5 +#define LINE1R_2_L_ENUM 6 +#define LINE1R_2_R_ENUM 7 +#define LINE2L_ENUM 8 +#define LINE2R_ENUM 9 +#define ADC_HPF_ENUM 10 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -238,6 +240,8 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux), SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux), SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), @@ -490,12 +494,16 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { }; /* Left Line1 Mux */ -static const struct snd_kcontrol_new aic3x_left_line1_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_ENUM]); +static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]); +static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]); /* Right Line1 Mux */ -static const struct snd_kcontrol_new aic3x_right_line1_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_ENUM]); +static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]); +static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls = +SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]); /* Left Line2 Mux */ static const struct snd_kcontrol_new aic3x_left_line2_mux_controls = @@ -535,9 +543,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_left_pga_mixer_controls[0], ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line1_mux_controls), + &aic3x_left_line1l_mux_controls), SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line1_mux_controls), + &aic3x_left_line1r_mux_controls), SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line2_mux_controls), @@ -548,9 +556,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_pga_mixer_controls[0], ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_right_line1_mux_controls), + &aic3x_right_line1l_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, - &aic3x_right_line1_mux_controls), + &aic3x_right_line1r_mux_controls), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4c336636d4f..cd63bba623d 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -954,9 +954,9 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); /* * MICGAIN volume control: - * from -6 to 30 dB in 6 dB steps + * from 6 to 30 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0); +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); /* * AFMGAIN volume control: diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c new file mode 100644 index 00000000000..a2a09f85ea9 --- /dev/null +++ b/sound/soc/codecs/wm8782.c @@ -0,0 +1,80 @@ +/* + * sound/soc/codecs/wm8782.c + * simple, strap-pin configured 24bit 2ch ADC + * + * Copyright: 2011 Raumfeld GmbH + * Author: Johannes Stezenbach <js@sig21.net> + * + * based on ad73311.c + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.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. + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +static struct snd_soc_dai_driver wm8782_dai = { + .name = "wm8782", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + /* For configurations with FSAMPEN=0 */ + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8782; + +static __devinit int wm8782_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_wm8782, &wm8782_dai, 1); +} + +static int __devexit wm8782_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver wm8782_codec_driver = { + .driver = { + .name = "wm8782", + .owner = THIS_MODULE, + }, + .probe = wm8782_probe, + .remove = wm8782_remove, +}; + +static int __init wm8782_init(void) +{ + return platform_driver_register(&wm8782_codec_driver); +} +module_init(wm8782_init); + +static void __exit wm8782_exit(void) +{ + platform_driver_unregister(&wm8782_codec_driver); +} +module_exit(wm8782_exit); + +MODULE_DESCRIPTION("ASoC WM8782 driver"); +MODULE_AUTHOR("Johannes Stezenbach <js@sig21.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 449ea09a193..082040eda8a 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1167,6 +1167,7 @@ static int wm8900_resume(struct snd_soc_codec *codec) ret = wm8900_set_fll(codec, 0, fll_in, fll_out); if (ret != 0) { dev_err(codec->dev, "Failed to restart FLL\n"); + kfree(cache); return ret; } } diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9b3bba4df5b..b085575d4aa 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2560,6 +2560,7 @@ static __devexit int wm8904_i2c_remove(struct i2c_client *client) static const struct i2c_device_id wm8904_i2c_id[] = { { "wm8904", WM8904 }, { "wm8912", WM8912 }, + { "wm8918", WM8904 }, /* Actually a subset, updates to follow */ { } }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index e2ab4fac281..423baa9be24 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -41,14 +41,12 @@ #define HPOUT2L 4 #define HPOUT2R 8 -#define WM8915_NUM_SUPPLIES 6 +#define WM8915_NUM_SUPPLIES 4 static const char *wm8915_supply_names[WM8915_NUM_SUPPLIES] = { - "DCVDD", "DBVDD", "AVDD1", "AVDD2", "CPVDD", - "MICVDD", }; struct wm8915_priv { @@ -57,6 +55,7 @@ struct wm8915_priv { int ldo1ena; int sysclk; + int sysclk_src; int fll_src; int fll_fref; @@ -76,6 +75,7 @@ struct wm8915_priv { struct wm8915_pdata pdata; int rx_rate[WM8915_AIFS]; + int bclk_rate[WM8915_AIFS]; /* Platform dependant ReTune mobile configuration */ int num_retune_mobile_texts; @@ -113,8 +113,6 @@ WM8915_REGULATOR_EVENT(0) WM8915_REGULATOR_EVENT(1) WM8915_REGULATOR_EVENT(2) WM8915_REGULATOR_EVENT(3) -WM8915_REGULATOR_EVENT(4) -WM8915_REGULATOR_EVENT(5) static const u16 wm8915_reg[WM8915_MAX_REGISTER] = { [WM8915_SOFTWARE_RESET] = 0x8915, @@ -1565,6 +1563,50 @@ static int wm8915_reset(struct snd_soc_codec *codec) return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915); } +static const int bclk_divs[] = { + 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 +}; + +static void wm8915_update_bclk(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int aif, best, cur_val, bclk_rate, bclk_reg, i; + + /* Don't bother if we're in a low frequency idle mode that + * can't support audio. + */ + if (wm8915->sysclk < 64000) + return; + + for (aif = 0; aif < WM8915_AIFS; aif++) { + switch (aif) { + case 0: + bclk_reg = WM8915_AIF1_BCLK; + break; + case 1: + bclk_reg = WM8915_AIF2_BCLK; + break; + } + + bclk_rate = wm8915->bclk_rate[aif]; + + /* Pick a divisor for BCLK as close as we can get to ideal */ + best = 0; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate; + if (cur_val < 0) /* BCLK table is sorted */ + break; + best = i; + } + bclk_rate = wm8915->sysclk / bclk_divs[best]; + dev_dbg(codec->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", + bclk_divs[best], bclk_rate); + + snd_soc_update_bits(codec, bclk_reg, + WM8915_AIF1_BCLK_DIV_MASK, best); + } +} + static int wm8915_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1717,10 +1759,6 @@ static int wm8915_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static const int bclk_divs[] = { - 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 -}; - static const int dsp_divs[] = { 48000, 32000, 16000, 8000 }; @@ -1731,17 +1769,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); - int bits, i, bclk_rate, best, cur_val; + int bits, i, bclk_rate; int aifdata = 0; - int bclk = 0; int lrclk = 0; int dsp = 0; - int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift; - - if (!wm8915->sysclk) { - dev_err(codec->dev, "SYSCLK not configured\n"); - return -EINVAL; - } + int aifdata_reg, lrclk_reg, dsp_shift; switch (dai->id) { case 0: @@ -1753,7 +1785,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1; lrclk_reg = WM8915_AIF1_TX_LRCLK_1; } - bclk_reg = WM8915_AIF1_BCLK; dsp_shift = 0; break; case 1: @@ -1765,7 +1796,6 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1; lrclk_reg = WM8915_AIF2_TX_LRCLK_1; } - bclk_reg = WM8915_AIF2_BCLK; dsp_shift = WM8915_DSP2_DIV_SHIFT; break; default: @@ -1779,6 +1809,9 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, return bclk_rate; } + wm8915->bclk_rate[dai->id] = bclk_rate; + wm8915->rx_rate[dai->id] = params_rate(params); + /* Needs looking at for TDM */ bits = snd_pcm_format_width(params_format(params)); if (bits < 0) @@ -1796,18 +1829,7 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, } dsp |= i << dsp_shift; - /* Pick a divisor for BCLK as close as we can get to ideal */ - best = 0; - for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { - cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate; - if (cur_val < 0) /* BCLK table is sorted */ - break; - best = i; - } - bclk_rate = wm8915->sysclk / bclk_divs[best]; - dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", - bclk_divs[best], bclk_rate); - bclk |= best; + wm8915_update_bclk(codec); lrclk = bclk_rate / params_rate(params); dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", @@ -1817,14 +1839,11 @@ static int wm8915_hw_params(struct snd_pcm_substream *substream, WM8915_AIF1TX_WL_MASK | WM8915_AIF1TX_SLOT_LEN_MASK, aifdata); - snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk); snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK, lrclk); snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2, WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp); - wm8915->rx_rate[dai->id] = params_rate(params); - return 0; } @@ -1838,6 +1857,9 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, int src; int old; + if (freq == wm8915->sysclk && clk_id == wm8915->sysclk_src) + return 0; + /* Disable SYSCLK while we reconfigure */ old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA; snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, @@ -1882,6 +1904,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, return -EINVAL; } + wm8915_update_bclk(codec); + snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK, src << WM8915_SYSCLK_SRC_SHIFT | ratediv); @@ -1889,6 +1913,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_ENA, old); + wm8915->sysclk_src = clk_id; + return 0; } @@ -2007,6 +2033,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source, unsigned int Fref, unsigned int Fout) { struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = to_i2c_client(codec->dev); struct _fll_div fll_div; unsigned long timeout; int ret, reg; @@ -2093,7 +2120,18 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source, else timeout = msecs_to_jiffies(2); - wait_for_completion_timeout(&wm8915->fll_lock, timeout); + /* Allow substantially longer if we've actually got the IRQ */ + if (i2c->irq) + timeout *= 1000; + + ret = wait_for_completion_timeout(&wm8915->fll_lock, timeout); + + if (ret == 0 && i2c->irq) { + dev_err(codec->dev, "Timed out waiting for FLL\n"); + ret = -ETIMEDOUT; + } else { + ret = 0; + } dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); @@ -2101,7 +2139,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source, wm8915->fll_fout = Fout; wm8915->fll_src = source; - return 0; + return ret; } #ifdef CONFIG_GPIOLIB @@ -2293,6 +2331,12 @@ static void wm8915_micd(struct snd_soc_codec *codec) SND_JACK_HEADSET | SND_JACK_BTN_0); wm8915->jack_mic = true; wm8915->detecting = false; + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_RATE_MASK, + 5 << WM8915_MICD_RATE_SHIFT); } /* If we detected a lower impedence during initial startup @@ -2333,15 +2377,17 @@ static void wm8915_micd(struct snd_soc_codec *codec) SND_JACK_HEADPHONE, SND_JACK_HEADSET | SND_JACK_BTN_0); + + /* Increase the detection rate a bit for + * responsiveness. + */ + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_RATE_MASK, + 7 << WM8915_MICD_RATE_SHIFT); + wm8915->detecting = false; } } - - /* Increase poll rate to give better responsiveness for buttons */ - if (!wm8915->detecting) - snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, - WM8915_MICD_RATE_MASK, - 5 << WM8915_MICD_RATE_SHIFT); } static irqreturn_t wm8915_irq(int irq, void *data) @@ -2383,6 +2429,20 @@ static irqreturn_t wm8915_irq(int irq, void *data) } } +static irqreturn_t wm8915_edge_irq(int irq, void *data) +{ + irqreturn_t ret = IRQ_NONE; + irqreturn_t val; + + do { + val = wm8915_irq(irq, data); + if (val != IRQ_NONE) + ret = val; + } while (val != IRQ_NONE); + + return ret; +} + static void wm8915_retune_mobile_pdata(struct snd_soc_codec *codec) { struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); @@ -2482,8 +2542,6 @@ static int wm8915_probe(struct snd_soc_codec *codec) wm8915->disable_nb[1].notifier_call = wm8915_regulator_event_1; wm8915->disable_nb[2].notifier_call = wm8915_regulator_event_2; wm8915->disable_nb[3].notifier_call = wm8915_regulator_event_3; - wm8915->disable_nb[4].notifier_call = wm8915_regulator_event_4; - wm8915->disable_nb[5].notifier_call = wm8915_regulator_event_5; /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) { @@ -2709,8 +2767,14 @@ static int wm8915_probe(struct snd_soc_codec *codec) irq_flags |= IRQF_ONESHOT; - ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq, - irq_flags, "wm8915", codec); + if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) + ret = request_threaded_irq(i2c->irq, NULL, + wm8915_edge_irq, + irq_flags, "wm8915", codec); + else + ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq, + irq_flags, "wm8915", codec); + if (ret == 0) { /* Unmask the interrupt */ snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 25580e3ee7c..056daa0010f 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -297,8 +297,6 @@ static int wm8940_add_widgets(struct snd_soc_codec *codec) if (ret) goto error_ret; ret = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - if (ret) - goto error_ret; error_ret: return ret; @@ -683,8 +681,6 @@ static int wm8940_resume(struct snd_soc_codec *codec) } } ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret) - goto error_ret; error_ret: return ret; @@ -730,9 +726,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) if (ret) return ret; ret = wm8940_add_widgets(codec); - if (ret) - return ret; - return ret; } diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5e05eed96c3..8499c563a9b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -78,6 +78,8 @@ struct wm8962_priv { #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif + + int irq; }; /* We can't use the same notifier block for more than one supply and @@ -1982,6 +1984,7 @@ static const unsigned int classd_tlv[] = { 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); /* The VU bits for the headphones are in a different register to the mute * bits and only take effect on the PGA if it is actually powered. @@ -2119,6 +2122,18 @@ SOC_SINGLE_TLV("HPMIXR MIXINR Volume", WM8962_HEADPHONE_MIXER_4, SOC_SINGLE_TLV("Speaker Boost Volume", WM8962_CLASS_D_CONTROL_2, 0, 7, 0, classd_tlv), + +SOC_SINGLE("EQ Switch", WM8962_EQ1, WM8962_EQ_ENA_SHIFT, 1, 0), +SOC_DOUBLE_R_TLV("EQ1 Volume", WM8962_EQ2, WM8962_EQ22, + WM8962_EQL_B1_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_DOUBLE_R_TLV("EQ2 Volume", WM8962_EQ2, WM8962_EQ22, + WM8962_EQL_B2_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_DOUBLE_R_TLV("EQ3 Volume", WM8962_EQ2, WM8962_EQ22, + WM8962_EQL_B3_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23, + WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv), +SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23, + WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv), }; static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = { @@ -2184,6 +2199,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + unsigned long timeout; int src; int fll; @@ -2203,9 +2220,19 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (fll) + if (fll) { snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, WM8962_FLL_ENA); + if (wm8962->irq) { + timeout = msecs_to_jiffies(5); + timeout = wait_for_completion_timeout(&wm8962->fll_lock, + timeout); + + if (timeout == 0) + dev_err(codec->dev, + "Timed out starting FLL\n"); + } + } break; case SND_SOC_DAPM_POST_PMD: @@ -2763,18 +2790,44 @@ static const int bclk_divs[] = { 1, -1, 2, 3, 4, -1, 6, 8, -1, 12, 16, 24, -1, 32, 32, 32 }; +static const int sysclk_rates[] = { + 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536, +}; + static void wm8962_configure_bclk(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int dspclk, i; int clocking2 = 0; + int clocking4 = 0; int aif2 = 0; - if (!wm8962->bclk) { - dev_dbg(codec->dev, "No BCLK rate configured\n"); + if (!wm8962->sysclk_rate) { + dev_dbg(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!wm8962->bclk || !wm8962->lrclk) { + dev_dbg(codec->dev, "No audio clocks configured\n"); return; } + for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) { + if (sysclk_rates[i] == wm8962->sysclk_rate / wm8962->lrclk) { + clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT; + break; + } + } + + if (i == ARRAY_SIZE(sysclk_rates)) { + dev_err(codec->dev, "Unsupported sysclk ratio %d\n", + wm8962->sysclk_rate / wm8962->lrclk); + return; + } + + snd_soc_update_bits(codec, WM8962_CLOCKING_4, + WM8962_SYSCLK_RATE_MASK, clocking4); + dspclk = snd_soc_read(codec, WM8962_CLOCKING1); if (dspclk < 0) { dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); @@ -2844,6 +2897,8 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*50k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x80); + + wm8962_configure_bclk(codec); break; case SND_SOC_BIAS_STANDBY: @@ -2876,8 +2931,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); - - wm8962_configure_bclk(codec); } /* VMID 2*250k */ @@ -2918,10 +2971,6 @@ static const struct { { 96000, 6 }, }; -static const int sysclk_rates[] = { - 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536, -}; - static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2929,41 +2978,27 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int rate = params_rate(params); int i; int aif0 = 0; int adctl3 = 0; - int clocking4 = 0; wm8962->bclk = snd_soc_params_to_bclk(params); wm8962->lrclk = params_rate(params); for (i = 0; i < ARRAY_SIZE(sr_vals); i++) { - if (sr_vals[i].rate == rate) { + if (sr_vals[i].rate == wm8962->lrclk) { adctl3 |= sr_vals[i].reg; break; } } if (i == ARRAY_SIZE(sr_vals)) { - dev_err(codec->dev, "Unsupported rate %dHz\n", rate); + dev_err(codec->dev, "Unsupported rate %dHz\n", wm8962->lrclk); return -EINVAL; } - if (rate % 8000 == 0) + if (wm8962->lrclk % 8000 == 0) adctl3 |= WM8962_SAMPLE_RATE_INT_MODE; - for (i = 0; i < ARRAY_SIZE(sysclk_rates); i++) { - if (sysclk_rates[i] == wm8962->sysclk_rate / rate) { - clocking4 |= i << WM8962_SYSCLK_RATE_SHIFT; - break; - } - } - if (i == ARRAY_SIZE(sysclk_rates)) { - dev_err(codec->dev, "Unsupported sysclk ratio %d\n", - wm8962->sysclk_rate / rate); - return -EINVAL; - } - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -2985,8 +3020,6 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, WM8962_ADDITIONAL_CONTROL_3, WM8962_SAMPLE_RATE_INT_MODE | WM8962_SAMPLE_RATE_MASK, adctl3); - snd_soc_update_bits(codec, WM8962_CLOCKING_4, - WM8962_SYSCLK_RATE_MASK, clocking4); wm8962_configure_bclk(codec); @@ -3261,16 +3294,31 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); - /* This should be a massive overestimate */ - timeout = msecs_to_jiffies(1); + ret = 0; + + if (fll1 & WM8962_FLL_ENA) { + /* This should be a massive overestimate but go even + * higher if we'll error out + */ + if (wm8962->irq) + timeout = msecs_to_jiffies(5); + else + timeout = msecs_to_jiffies(1); + + timeout = wait_for_completion_timeout(&wm8962->fll_lock, + timeout); - wait_for_completion_timeout(&wm8962->fll_lock, timeout); + if (timeout == 0 && wm8962->irq) { + dev_err(codec->dev, "FLL lock timed out"); + ret = -ETIMEDOUT; + } + } wm8962->fll_fref = Fref; wm8962->fll_fout = Fout; wm8962->fll_src = source; - return 0; + return ret; } static int wm8962_mute(struct snd_soc_dai *dai, int mute) @@ -3731,8 +3779,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); - struct i2c_client *i2c = container_of(codec->dev, struct i2c_client, - dev); u16 *reg_cache = codec->reg_cache; int i, trigger, irq_pol; bool dmicclk, dmicdat; @@ -3871,6 +3917,9 @@ static int wm8962_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, WM8962_HPOUT_VU, WM8962_HPOUT_VU); + /* Stereo control for EQ */ + snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0); + wm8962_add_widgets(codec); /* Save boards having to disable DMIC when not in use */ @@ -3899,7 +3948,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_init_beep(codec); wm8962_init_gpio(codec); - if (i2c->irq) { + if (wm8962->irq) { if (pdata && pdata->irq_active_low) { trigger = IRQF_TRIGGER_LOW; irq_pol = WM8962_IRQ_POL; @@ -3911,12 +3960,13 @@ static int wm8962_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_INTERRUPT_CONTROL, WM8962_IRQ_POL, irq_pol); - ret = request_threaded_irq(i2c->irq, NULL, wm8962_irq, + ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq, trigger | IRQF_ONESHOT, "wm8962", codec); if (ret != 0) { dev_err(codec->dev, "Failed to request IRQ %d: %d\n", - i2c->irq, ret); + wm8962->irq, ret); + wm8962->irq = 0; /* Non-fatal */ } else { /* Enable some IRQs by default */ @@ -3941,12 +3991,10 @@ err: static int wm8962_remove(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = container_of(codec->dev, struct i2c_client, - dev); int i; - if (i2c->irq) - free_irq(i2c->irq, codec); + if (wm8962->irq) + free_irq(wm8962->irq, codec); cancel_delayed_work_sync(&wm8962->mic_work); @@ -3986,6 +4034,8 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8962); + wm8962->irq = i2c->irq; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8962, &wm8962_dai, 1); if (ret < 0) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 9e5ff789b80..6e85b8869af 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -876,7 +876,7 @@ SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { @@ -1434,6 +1434,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.hp_startup_mode = 1; wm8993->hubs_data.dcs_codes = -2; + wm8993->hubs_data.series_startup = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret != 0) { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5f0c238e178..ee64be2d994 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -195,10 +195,6 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) aif + 1, rate); } - if (rate && rate < 3000000) - dev_warn(codec->dev, "AIF%dCLK is %dHz, should be >=3MHz for optimal performance\n", - aif + 1, rate); - wm8994->aifclk[aif] = rate; snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset, @@ -1146,13 +1142,33 @@ SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), +SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), }; static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { @@ -1283,14 +1299,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), - -SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, - left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), -SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, - right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - SND_SOC_DAPM_POST("Debug log", post_ev), }; @@ -1623,6 +1631,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, int reg_offset, ret; struct fll_div fll; u16 reg, aif1, aif2; + unsigned long timeout; aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA; @@ -1714,7 +1723,15 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_ENA | WM8994_FLL1_FRAC, reg); - msleep(5); + if (wm8994->fll_locked_irq) { + timeout = wait_for_completion_timeout(&wm8994->fll_locked[id], + msecs_to_jiffies(10)); + if (timeout == 0) + dev_warn(codec->dev, + "Timed out waiting for FLL lock\n"); + } else { + msleep(5); + } } wm8994->fll[id].in = freq_in; @@ -1732,6 +1749,14 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, return 0; } +static irqreturn_t wm8994_fll_locked_irq(int irq, void *data) +{ + struct completion *completion = data; + + complete(completion); + + return IRQ_HANDLED; +} static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 }; @@ -2849,6 +2874,15 @@ out: return IRQ_HANDLED; } +static irqreturn_t wm8994_fifo_error(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "FIFO error\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -2867,6 +2901,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + init_completion(&wm8994->fll_locked[i]); + if (wm8994->pdata && wm8994->pdata->micdet_irq) wm8994->micdet_irq = wm8994->pdata->micdet_irq; else if (wm8994->pdata && wm8994->pdata->irq_base) @@ -2905,6 +2942,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_codes = -5; wm8994->hubs.hp_startup_mode = 1; wm8994->hubs.dcs_readback_mode = 1; + wm8994->hubs.series_startup = 1; break; default: wm8994->hubs.dcs_readback_mode = 1; @@ -2919,6 +2957,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, + wm8994_fifo_error, "FIFO error", codec); + + ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, + wm_hubs_dcs_done, "DC servo done", + &wm8994->hubs); + if (ret == 0) + wm8994->hubs.dcs_done_irq = true; + switch (control->type) { case WM8994: if (wm8994->micdet_irq) { @@ -2975,6 +3022,16 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } } + wm8994->fll_locked_irq = true; + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) { + ret = wm8994_request_irq(codec->control_data, + WM8994_IRQ_FLL1_LOCK + i, + wm8994_fll_locked_irq, "FLL lock", + &wm8994->fll_locked[i]); + if (ret != 0) + wm8994->fll_locked_irq = false; + } + /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically * at runtime we can deal with that then. @@ -3050,10 +3107,18 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); - /* Unconditionally enable AIF1 ADC TDM mode; it only affects - * behaviour on idle TDM clock cycles. */ - snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, - WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + /* Unconditionally enable AIF1 ADC TDM mode on chips which can + * use this; it only affects behaviour on idle TDM clock + * cycles. */ + switch (control->type) { + case WM8994: + case WM8958: + snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, + WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + break; + default: + break; + } wm8994_update_class_w(codec); @@ -3152,6 +3217,12 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); if (wm8994->micdet_irq) free_irq(wm8994->micdet_irq, wm8994); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + &wm8994->fll_locked[i]); + wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, + &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); err: kfree(wm8994); return ret; @@ -3161,11 +3232,20 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; + int i; wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); pm_runtime_disable(codec->dev); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + &wm8994->fll_locked[i]); + + wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, + &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); + switch (control->type) { case WM8994: if (wm8994->micdet_irq) diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 0a1db04b73b..1ab2266039f 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -11,6 +11,7 @@ #include <sound/soc.h> #include <linux/firmware.h> +#include <linux/completion.h> #include "wm_hubs.h" @@ -79,6 +80,8 @@ struct wm8994_priv { int mclk[2]; int aifclk[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; + struct completion fll_locked[2]; + bool fll_locked_irq; int dac_rates[2]; int lrclk_shared[2]; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 91c6b39de50..a4691321f9b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -727,7 +727,7 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), +SND_SOC_DAPM_OUT_DRV("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LINEOUT"), SND_SOC_DAPM_OUTPUT("SPKN"), diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 9e370d14ad8..5c2d5657b47 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -63,9 +63,11 @@ static const struct soc_enum speaker_mode = static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; int count = 0; unsigned int val; + unsigned long timeout; val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; @@ -74,18 +76,37 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) dev_dbg(codec->dev, "Waiting for DC servo...\n"); - do { - count++; - msleep(1); + if (hubs->dcs_done_irq) { + timeout = wait_for_completion_timeout(&hubs->dcs_done, + msecs_to_jiffies(500)); + if (timeout == 0) + dev_warn(codec->dev, "No DC servo interrupt\n"); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < 400); + } else { + do { + count++; + msleep(1); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & op && count < 400); + } if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", op); } +irqreturn_t wm_hubs_dcs_done(int irq, void *data) +{ + struct wm_hubs_data *hubs = data; + + complete(&hubs->dcs_done); + + return IRQ_HANDLED; +} +EXPORT_SYMBOL_GPL(wm_hubs_dcs_done); + /* * Startup calibration of the DC servo */ @@ -107,8 +128,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) return; } - /* Devices not using a DCS code correction have startup mode */ - if (hubs->dcs_codes) { + if (hubs->series_startup) { /* Set for 32 series updates */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_SERIES_NO_01_MASK, @@ -134,9 +154,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) break; case 1: reg = snd_soc_read(codec, WM8993_DC_SERVO_3); - reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; - reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; break; default: WARN(1, "Unknown DCS readback method\n"); @@ -150,13 +170,13 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Applying %d code DC servo correction\n", hubs->dcs_codes); - /* HPOUT1L */ - offset = reg_l; + /* HPOUT1R */ + offset = reg_r; offset += hubs->dcs_codes; dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; - /* HPOUT1R */ - offset = reg_r; + /* HPOUT1L */ + offset = reg_l; offset += hubs->dcs_codes; dcs_cfg |= (u8)offset; @@ -168,8 +188,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); } else { - dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; - dcs_cfg |= reg_r; + dcs_cfg = reg_r << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + dcs_cfg |= reg_l; } /* Save the callibrated offset if we're in class W mode and @@ -195,7 +215,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ - if (hubs->dcs_codes) + if (hubs->dcs_codes || hubs->no_series_update) return ret; /* Only need to do this if the outputs are active */ @@ -599,9 +619,6 @@ SND_SOC_DAPM_MIXER("IN2L PGA", WM8993_POWER_MANAGEMENT_2, 7, 0, SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0, in2r_pga, ARRAY_SIZE(in2r_pga)), -/* Dummy widgets to represent differential paths */ -SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0, mixinl, ARRAY_SIZE(mixinl)), SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0, @@ -867,8 +884,11 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls); int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + init_completion(&hubs->dcs_done); + snd_soc_dapm_add_routes(dapm, analogue_routes, ARRAY_SIZE(analogue_routes)); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index f8a5e976b5e..676b1252ab9 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -14,6 +14,9 @@ #ifndef _WM_HUBS_H #define _WM_HUBS_H +#include <linux/completion.h> +#include <linux/interrupt.h> + struct snd_soc_codec; extern const unsigned int wm_hubs_spkmix_tlv[]; @@ -23,9 +26,14 @@ struct wm_hubs_data { int dcs_codes; int dcs_readback_mode; int hp_startup_mode; + int series_startup; + int no_series_update; bool class_w; u16 class_w_dcs; + + bool dcs_done_irq; + struct completion dcs_done; }; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); @@ -36,4 +44,6 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int jd_scthr, int jd_thr, int micbias1_lvl, int micbias2_lvl); +extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); + #endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 9d35b8c1a62..a49e667373b 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -46,11 +46,28 @@ static void print_buf_info(int slot, char *name) } #endif +#define DAVINCI_PCM_FMTBITS (\ + SNDRV_PCM_FMTBIT_S8 |\ + SNDRV_PCM_FMTBIT_U8 |\ + SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_U16_LE |\ + SNDRV_PCM_FMTBIT_U16_BE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S24_BE |\ + SNDRV_PCM_FMTBIT_U24_LE |\ + SNDRV_PCM_FMTBIT_U24_BE |\ + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_S32_BE |\ + SNDRV_PCM_FMTBIT_U32_LE |\ + SNDRV_PCM_FMTBIT_U32_BE) + static struct snd_pcm_hardware pcm_hardware_playback = { .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), - .formats = (SNDRV_PCM_FMTBIT_S16_LE), + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| + SNDRV_PCM_INFO_BATCH), + .formats = DAVINCI_PCM_FMTBITS, .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | @@ -59,7 +76,7 @@ static struct snd_pcm_hardware pcm_hardware_playback = { .rate_min = 8000, .rate_max = 96000, .channels_min = 2, - .channels_max = 2, + .channels_max = 384, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, @@ -71,8 +88,9 @@ static struct snd_pcm_hardware pcm_hardware_playback = { static struct snd_pcm_hardware pcm_hardware_capture = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE), - .formats = (SNDRV_PCM_FMTBIT_S16_LE), + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_BATCH), + .formats = DAVINCI_PCM_FMTBITS, .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | @@ -81,7 +99,7 @@ static struct snd_pcm_hardware pcm_hardware_capture = { .rate_min = 8000, .rate_max = 96000, .channels_min = 2, - .channels_max = 2, + .channels_max = 384, .buffer_bytes_max = 128 * 1024, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, @@ -139,6 +157,22 @@ struct davinci_runtime_data { struct edmacc_param ram_params; }; +static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static void davinci_pcm_period_reset(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + + prtd->period = 0; +} /* * Not used with ping/pong */ @@ -199,10 +233,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) else edma_set_transfer_params(link, acnt, fifo_level, count, fifo_level, ABSYNC); - - prtd->period++; - if (unlikely(prtd->period >= runtime->periods)) - prtd->period = 0; } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) @@ -217,12 +247,13 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) return; if (snd_pcm_running(substream)) { + spin_lock(&prtd->lock); if (prtd->ram_channel < 0) { /* No ping/pong must fix up link dma data*/ - spin_lock(&prtd->lock); davinci_pcm_enqueue_dma(substream); - spin_unlock(&prtd->lock); } + davinci_pcm_period_elapsed(substream); + spin_unlock(&prtd->lock); snd_pcm_period_elapsed(substream); } } @@ -425,7 +456,8 @@ static int request_ping_pong(struct snd_pcm_substream *substream, edma_read_slot(link, &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); - prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); + prtd->asp_params.opt |= TCCHEN | + EDMA_TCC(prtd->ram_channel & 0x3f); edma_write_slot(link, &prtd->asp_params); /* pong */ @@ -439,7 +471,7 @@ static int request_ping_pong(struct snd_pcm_substream *substream, prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); /* interrupt after every pong completion */ prtd->asp_params.opt |= TCINTEN | TCCHEN | - EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel)); + EDMA_TCC(prtd->ram_channel & 0x3f); edma_write_slot(link, &prtd->asp_params); /* ram */ @@ -527,6 +559,13 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: + edma_start(prtd->asp_channel); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + prtd->ram_channel >= 0) { + /* copy 1st iram buffer */ + edma_start(prtd->ram_channel); + } + break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: edma_resume(prtd->asp_channel); @@ -550,6 +589,7 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; + davinci_pcm_period_reset(substream); if (prtd->ram_channel >= 0) { int ret = ping_pong_dma_setup(substream); if (ret < 0) @@ -565,21 +605,31 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) print_buf_info(prtd->asp_link[0], "asp_link[0]"); print_buf_info(prtd->asp_link[1], "asp_link[1]"); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* copy 1st iram buffer */ - edma_start(prtd->ram_channel); - } - edma_start(prtd->asp_channel); + /* + * There is a phase offset of 2 periods between the position + * used by dma setup and the position reported in the pointer + * function. + * + * The phase offset, when not using ping-pong buffers, is due to + * the two consecutive calls to davinci_pcm_enqueue_dma() below. + * + * Whereas here, with ping-pong buffers, the phase is due to + * there being an entire buffer transfer complete before the + * first dma completion event triggers davinci_pcm_dma_irq(). + */ + davinci_pcm_period_elapsed(substream); + davinci_pcm_period_elapsed(substream); + return 0; } - prtd->period = 0; davinci_pcm_enqueue_dma(substream); + davinci_pcm_period_elapsed(substream); /* Copy self-linked parameter RAM entry into master channel */ edma_read_slot(prtd->asp_link[0], &prtd->asp_params); edma_write_slot(prtd->asp_channel, &prtd->asp_params); davinci_pcm_enqueue_dma(substream); - edma_start(prtd->asp_channel); + davinci_pcm_period_elapsed(substream); return 0; } @@ -591,51 +641,23 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = runtime->private_data; unsigned int offset; int asp_count; - dma_addr_t asp_src, asp_dst; - + unsigned int period_size = snd_pcm_lib_period_bytes(substream); + + /* + * There is a phase offset of 2 periods between the position used by dma + * setup and the position reported in the pointer function. Either +2 in + * the dma setup or -2 here in the pointer function (with wrapping, + * both) accounts for this offset -- choose the latter since it makes + * the first-time setup clearer. + */ spin_lock(&prtd->lock); - if (prtd->ram_channel >= 0) { - int ram_count; - int mod_ram; - dma_addr_t ram_src, ram_dst; - unsigned int period_size = snd_pcm_lib_period_bytes(substream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* reading ram before asp should be safe - * as long as the asp transfers less than a ping size - * of bytes between the 2 reads - */ - edma_get_position(prtd->ram_channel, - &ram_src, &ram_dst); - edma_get_position(prtd->asp_channel, - &asp_src, &asp_dst); - asp_count = asp_src - prtd->asp_params.src; - ram_count = ram_src - prtd->ram_params.src; - mod_ram = ram_count % period_size; - mod_ram -= asp_count; - if (mod_ram < 0) - mod_ram += period_size; - else if (mod_ram == 0) { - if (snd_pcm_running(substream)) - mod_ram += period_size; - } - ram_count -= mod_ram; - if (ram_count < 0) - ram_count += period_size * runtime->periods; - } else { - edma_get_position(prtd->ram_channel, - &ram_src, &ram_dst); - ram_count = ram_dst - prtd->ram_params.dst; - } - asp_count = ram_count; - } else { - edma_get_position(prtd->asp_channel, &asp_src, &asp_dst); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - asp_count = asp_src - runtime->dma_addr; - else - asp_count = asp_dst - runtime->dma_addr; - } + asp_count = prtd->period - 2; spin_unlock(&prtd->lock); + if (asp_count < 0) + asp_count += runtime->periods; + asp_count *= period_size; + offset = bytes_to_frames(runtime, asp_count); if (offset >= runtime->buffer_size) offset = 0; @@ -811,9 +833,11 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; -static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; if (!card->dev->dma_mask) diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index a456e491155..e27c417da43 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -266,9 +266,11 @@ static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 ep93xx_pcm_dmamask = 0xffffffff; -static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 6680c0b4d20..732208c8c0b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -294,9 +294,11 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * Regardless of where the memory is actually allocated, since the device can * technically DMA to any 36-bit address, we do need to set the DMA mask to 36. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; @@ -939,7 +941,7 @@ static int __devinit fsl_soc_dma_probe(struct platform_device *pdev) iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL); if (iprop) - dma->ssi_fifo_depth = *iprop; + dma->ssi_fifo_depth = be32_to_cpup(iprop); else /* Older 8610 DTs didn't have the fifo-depth property */ dma->ssi_fifo_depth = 8; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 313e0ccedd5..d48afea5d93 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -678,7 +678,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) kfree(ssi_private); return ret; } - ssi_private->ssi = ioremap(res.start, 1 + res.end - res.start); + ssi_private->ssi = of_iomap(np, 0); + if (!ssi_private->ssi) { + dev_err(&pdev->dev, "could not map device resources\n"); + kfree(ssi_private); + return -ENOMEM; + } ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); @@ -691,7 +696,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); if (iprop) - ssi_private->fifo_depth = *iprop; + ssi_private->fifo_depth = be32_to_cpup(iprop); else /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fff695ccdd3..19ad0c1be67 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -299,10 +299,11 @@ static struct snd_pcm_ops psc_dma_ops = { }; static u64 psc_dma_dmamask = 0xffffffff; -static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); size_t size = psc_dma_hardware.buffer_bytes_max; int rc = 0; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index c16c6b2eff9..a1929795958 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -233,7 +233,7 @@ static int get_parent_cell_index(struct device_node *np) if (!iprop) return -1; - return *iprop; + return be32_to_cpup(iprop); } /** @@ -258,7 +258,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (!iprop) return -EINVAL; - addr = *iprop; + addr = be32_to_cpup(iprop); bus = get_parent_cell_index(np); if (bus < 0) @@ -305,7 +305,7 @@ static int get_dma_channel(struct device_node *ssi_np, return -EINVAL; } - *dma_channel_id = *iprop; + *dma_channel_id = be32_to_cpup(iprop); *dma_id = get_parent_cell_index(dma_channel_np); of_node_put(dma_channel_np); @@ -379,7 +379,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - machine_data->ssi_id = *iprop; + machine_data->ssi_id = be32_to_cpup(iprop); /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -405,7 +405,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - machine_data->clk_frequency = *iprop; + machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { machine_data->dai_format = SND_SOC_DAIFMT_I2S; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 66e0b68af14..8fa4d5f8eda 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -232,7 +232,7 @@ static int get_parent_cell_index(struct device_node *np) iprop = of_get_property(parent, "cell-index", NULL); if (iprop) - ret = *iprop; + ret = be32_to_cpup(iprop); of_node_put(parent); @@ -261,7 +261,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (!iprop) return -EINVAL; - addr = *iprop; + addr = be32_to_cpup(iprop); bus = get_parent_cell_index(np); if (bus < 0) @@ -308,7 +308,7 @@ static int get_dma_channel(struct device_node *ssi_np, return -EINVAL; } - *dma_channel_id = *iprop; + *dma_channel_id = be32_to_cpup(iprop); *dma_id = get_parent_cell_index(dma_channel_np); of_node_put(dma_channel_np); @@ -379,7 +379,7 @@ static int p1022_ds_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - mdata->ssi_id = *iprop; + mdata->ssi_id = be32_to_cpup(iprop); /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -405,7 +405,7 @@ static int p1022_ds_probe(struct platform_device *pdev) ret = -EINVAL; goto error; } - mdata->clk_frequency = *iprop; + mdata->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { mdata->dai_format = SND_SOC_DAIFMT_I2S; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 413b78da248..309c59e6fb6 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -238,12 +238,14 @@ static struct snd_pcm_ops imx_pcm_ops = { static int ssi_irq = 0; -static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; - ret = imx_pcm_new(card, dai, pcm); + ret = imx_pcm_new(rtd); if (ret) return ret; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 61fceb09cdb..10a8e278375 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -388,10 +388,11 @@ static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); -int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) { - + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index dc8a87530e3..0a84cec3599 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -225,8 +225,7 @@ struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, struct imx_ssi *ssi); int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); -int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm); +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); void imx_pcm_free(struct snd_pcm *pcm); /* diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index fb1483f7c96..a7c9578be98 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -299,9 +299,11 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); -int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index e13c6ce4632..cd33de1c5b7 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -312,9 +312,11 @@ static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm, return 0; } -static int kirkwood_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; if (!card->dev->dma_mask) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 5a946b4115a..3e7826058ef 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -402,9 +402,10 @@ static void sst_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int retval = 0; pr_debug("sst_pcm_new called\n"); diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index dac6732da96..9c0edad90d8 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -356,7 +356,7 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) nuc900_audio->irq_num = platform_get_irq(pdev, 0); if (!nuc900_audio->irq_num) { ret = -EBUSY; - goto out2; + goto out3; } nuc900_ac97_data = nuc900_audio; diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index 8263f56dc66..d589ef14e91 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -315,9 +315,12 @@ static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) } static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); -static int nuc900_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + if (!card->dev->dma_mask) card->dev->dma_mask = &nuc900_pcm_dmamask; if (!card->dev->coherent_dma_mask) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 99054cf1f68..fe83d0d176b 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -9,6 +9,9 @@ config SND_OMAP_SOC_MCBSP config SND_OMAP_SOC_MCPDM tristate +config SND_OMAP_SOC_HDMI + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C @@ -100,6 +103,14 @@ config SND_OMAP_SOC_SDP4430 Say Y if you want to add support for SoC audio on Texas Instruments SDP4430. +config SND_OMAP_SOC_OMAP4_HDMI + tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" + depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 + select SND_OMAP_SOC_HDMI + help + Say Y if you want to add support for SoC HDMI audio on Texas Instruments + OMAP4 chips + config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 6c2c87eed5b..59e2c8d1e38 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -2,10 +2,12 @@ snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o +snd-soc-omap-hdmi-objs := omap-hdmi.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o +obj-$(CONFIG_SND_OMAP_SOC_HDMI) += snd-soc-omap-hdmi.o # OMAP Machine Support snd-soc-n810-objs := n810.o @@ -21,6 +23,7 @@ snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o snd-soc-igep0020-objs := igep0020.o +snd-soc-omap4-hdmi-objs := omap4-hdmi-card.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o @@ -36,3 +39,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP4_HDMI) += snd-soc-omap4-hdmi.o diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 462cbcbea74..b40095a1988 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -427,7 +427,8 @@ static struct snd_soc_ops ams_delta_ops = { /* Board specific codec bias level control */ static int ams_delta_set_bias_level(struct snd_soc_card *card, - enum snd_soc_bias_level level) + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) { struct snd_soc_codec *codec = card->rtd->codec; diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c new file mode 100644 index 00000000000..36c6eaeffb0 --- /dev/null +++ b/sound/soc/omap/omap-hdmi.c @@ -0,0 +1,158 @@ +/* + * omap-hdmi.c + * + * OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors. + * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/ + * Authors: Jorge Candelaria <jorge.candelaria@ti.com> + * Ricardo Neri <ricardo.neri@ti.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/dma.h> +#include "omap-pcm.h" +#include "omap-hdmi.h" + +#define DRV_NAME "hdmi-audio-dai" + +static struct omap_pcm_dma_data omap_hdmi_dai_dma_params = { + .name = "HDMI playback", + .sync_mode = OMAP_DMA_SYNC_PACKET, +}; + +static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int err; + /* + * Make sure that the period bytes are multiple of the DMA packet size. + * Largest packet size we use is 32 32-bit words = 128 bytes + */ + err = snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128); + if (err < 0) + return err; + + return 0; +} + +static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int err = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + omap_hdmi_dai_dma_params.packet_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + omap_hdmi_dai_dma_params.packet_size = 32; + break; + default: + err = -EINVAL; + } + + omap_hdmi_dai_dma_params.data_type = OMAP_DMA_DATA_TYPE_S32; + + snd_soc_dai_set_dma_data(dai, substream, + &omap_hdmi_dai_dma_params); + + return err; +} + +static struct snd_soc_dai_ops omap_hdmi_dai_ops = { + .startup = omap_hdmi_dai_startup, + .hw_params = omap_hdmi_dai_hw_params, +}; + +static struct snd_soc_dai_driver omap_hdmi_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = OMAP_HDMI_RATES, + .formats = OMAP_HDMI_FORMATS, + }, + .ops = &omap_hdmi_dai_ops, +}; + +static __devinit int omap_hdmi_probe(struct platform_device *pdev) +{ + int ret; + struct resource *hdmi_rsrc; + + hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!hdmi_rsrc) { + dev_err(&pdev->dev, "Cannot obtain IORESOURCE_MEM HDMI\n"); + return -EINVAL; + } + + omap_hdmi_dai_dma_params.port_addr = hdmi_rsrc->start + + OMAP_HDMI_AUDIO_DMA_PORT; + + hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!hdmi_rsrc) { + dev_err(&pdev->dev, "Cannot obtain IORESOURCE_DMA HDMI\n"); + return -EINVAL; + } + + omap_hdmi_dai_dma_params.dma_req = hdmi_rsrc->start; + + ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai); + return ret; +} + +static int __devexit omap_hdmi_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&pdev->dev); + return 0; +} + +static struct platform_driver hdmi_dai_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = omap_hdmi_probe, + .remove = __devexit_p(omap_hdmi_remove), +}; + +static int __init hdmi_dai_init(void) +{ + return platform_driver_register(&hdmi_dai_driver); +} +module_init(hdmi_dai_init); + +static void __exit hdmi_dai_exit(void) +{ + platform_driver_unregister(&hdmi_dai_driver); +} +module_exit(hdmi_dai_exit); + +MODULE_AUTHOR("Jorge Candelaria <jorge.candelaria@ti.com>"); +MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); +MODULE_DESCRIPTION("OMAP HDMI SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-hdmi.h b/sound/soc/omap/omap-hdmi.h new file mode 100644 index 00000000000..34c298d5057 --- /dev/null +++ b/sound/soc/omap/omap-hdmi.h @@ -0,0 +1,36 @@ +/* + * omap-hdmi.h + * + * Definitions for OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors. + * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/ + * Authors: Jorge Candelaria <jorge.candelaria@ti.com> + * Ricardo Neri <ricardo.neri@ti.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_HDMI_H__ +#define __OMAP_HDMI_H__ + +#define OMAP_HDMI_AUDIO_DMA_PORT 0x8c + +#define OMAP_HDMI_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define OMAP_HDMI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#endif diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index e6a6b991d05..b2f5751edae 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -366,9 +366,11 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c new file mode 100644 index 00000000000..9f32615b81f --- /dev/null +++ b/sound/soc/omap/omap4-hdmi-card.c @@ -0,0 +1,129 @@ +/* + * omap4-hdmi-card.c + * + * OMAP ALSA SoC machine driver for TI OMAP4 HDMI + * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Ricardo Neri <ricardo.neri@ti.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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/mach-types.h> +#include <video/omapdss.h> + +#define DRV_NAME "omap4-hdmi-audio" + +static int omap4_hdmi_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct omap_overlay_manager *mgr = NULL; + struct device *dev = substream->pcm->card->dev; + + /* Find DSS HDMI device */ + for (i = 0; i < omap_dss_get_num_overlay_managers(); i++) { + mgr = omap_dss_get_overlay_manager(i); + if (mgr && mgr->device + && mgr->device->type == OMAP_DISPLAY_TYPE_HDMI) + break; + } + + if (i == omap_dss_get_num_overlay_managers()) { + dev_err(dev, "HDMI display device not found!\n"); + return -ENODEV; + } + + /* Make sure HDMI is power-on to avoid L3 interconnect errors */ + if (mgr->device->state != OMAP_DSS_DISPLAY_ACTIVE) { + dev_err(dev, "HDMI display is not active!\n"); + return -EIO; + } + + return 0; +} + +static struct snd_soc_ops omap4_hdmi_dai_ops = { + .hw_params = omap4_hdmi_dai_hw_params, +}; + +static struct snd_soc_dai_link omap4_hdmi_dai = { + .name = "HDMI", + .stream_name = "HDMI", + .cpu_dai_name = "hdmi-audio-dai", + .platform_name = "omap-pcm-audio", + .codec_name = "omapdss_hdmi", + .codec_dai_name = "hdmi-audio-codec", + .ops = &omap4_hdmi_dai_ops, +}; + +static struct snd_soc_card snd_soc_omap4_hdmi = { + .name = "OMAP4HDMI", + .dai_link = &omap4_hdmi_dai, + .num_links = 1, +}; + +static __devinit int omap4_hdmi_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_omap4_hdmi; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + card->dev = NULL; + return ret; + } + return 0; +} + +static int __devexit omap4_hdmi_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + card->dev = NULL; + return 0; +} + +static struct platform_driver omap4_hdmi_driver = { + .driver = { + .name = "omap4-hdmi-audio", + .owner = THIS_MODULE, + }, + .probe = omap4_hdmi_probe, + .remove = __devexit_p(omap4_hdmi_remove), +}; + +static int __init omap4_hdmi_init(void) +{ + return platform_driver_register(&omap4_hdmi_driver); +} +module_init(omap4_hdmi_init); + +static void __exit omap4_hdmi_exit(void) +{ + platform_driver_unregister(&omap4_hdmi_driver); +} +module_exit(omap4_hdmi_exit); + +MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>"); +MODULE_DESCRIPTION("OMAP4 HDMI machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index fab20a54e86..c43060053dd 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -85,9 +85,10 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); -static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index ab3ccaec72d..80c85fd64e1 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -443,10 +443,11 @@ static void s6000_pcm_free(struct snd_pcm *pcm) static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); -static int s6000_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) { - struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct snd_card *card = runtime->card->snd_card; + struct snd_soc_dai *dai = runtime->cpu_dai; + struct snd_pcm *pcm = runtime->pcm; struct s6000_pcm_dma_params *params; int res; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index d155cbb58e1..54b0e4b7faf 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -158,7 +158,7 @@ config SND_SOC_GONI_AQUILA_WM8994 config SND_SOC_SAMSUNG_SMDK_SPDIF tristate "SoC S/PDIF Audio support for SMDK" - depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210) + depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310) select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. @@ -171,9 +171,23 @@ config SND_SOC_SMDK_WM8580_PCM help Say Y if you want to add support for SoC audio on the SMDK. +config SND_SOC_SMDK_WM8994_PCM + tristate "SoC PCM Audio support for WM8994 on SMDK" + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + select SND_SOC_WM8994 + select SND_SAMSUNG_PCM + help + Say Y if you want to add support for SoC audio on the SMDK + config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S select SND_SOC_WM8915 select SND_SOC_WM9081 + +config SND_SOC_SPEYSIDE_WM8962 + tristate "Audio support for Wolfson Speyside with WM8962" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM8962 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 683843a2744..9eb3b12eb72 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -35,7 +35,9 @@ snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o +snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o +snd-soc-speyside-wm8962-objs := speyside_wm8962.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -54,4 +56,6 @@ obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o +obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o +obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 5cb3b880f0d..9465588b02f 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -425,9 +425,11 @@ static void dma_free_dma_buffers(struct snd_pcm *pcm) static u64 dma_mask = DMA_BIT_MASK(32); -static int dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h new file mode 100644 index 00000000000..c0e6d9a19ef --- /dev/null +++ b/sound/soc/samsung/i2s-regs.h @@ -0,0 +1,143 @@ +/* + * linux/sound/soc/samsung/i2s-regs.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * Samsung I2S driver's register header + * + * 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 __SND_SOC_SAMSUNG_I2S_REGS_H +#define __SND_SOC_SAMSUNG_I2S_REGS_H + +#define I2SCON 0x0 +#define I2SMOD 0x4 +#define I2SFIC 0x8 +#define I2SPSR 0xc +#define I2STXD 0x10 +#define I2SRXD 0x14 +#define I2SFICS 0x18 +#define I2STXDS 0x1c +#define I2SAHB 0x20 +#define I2SSTR0 0x24 +#define I2SSIZE 0x28 +#define I2STRNCNT 0x2c +#define I2SLVL0ADDR 0x30 +#define I2SLVL1ADDR 0x34 +#define I2SLVL2ADDR 0x38 +#define I2SLVL3ADDR 0x3c + +#define CON_RSTCLR (1 << 31) +#define CON_FRXOFSTATUS (1 << 26) +#define CON_FRXORINTEN (1 << 25) +#define CON_FTXSURSTAT (1 << 24) +#define CON_FTXSURINTEN (1 << 23) +#define CON_TXSDMA_PAUSE (1 << 20) +#define CON_TXSDMA_ACTIVE (1 << 18) + +#define CON_FTXURSTATUS (1 << 17) +#define CON_FTXURINTEN (1 << 16) +#define CON_TXFIFO2_EMPTY (1 << 15) +#define CON_TXFIFO1_EMPTY (1 << 14) +#define CON_TXFIFO2_FULL (1 << 13) +#define CON_TXFIFO1_FULL (1 << 12) + +#define CON_LRINDEX (1 << 11) +#define CON_TXFIFO_EMPTY (1 << 10) +#define CON_RXFIFO_EMPTY (1 << 9) +#define CON_TXFIFO_FULL (1 << 8) +#define CON_RXFIFO_FULL (1 << 7) +#define CON_TXDMA_PAUSE (1 << 6) +#define CON_RXDMA_PAUSE (1 << 5) +#define CON_TXCH_PAUSE (1 << 4) +#define CON_RXCH_PAUSE (1 << 3) +#define CON_TXDMA_ACTIVE (1 << 2) +#define CON_RXDMA_ACTIVE (1 << 1) +#define CON_ACTIVE (1 << 0) + +#define MOD_OPCLK_CDCLK_OUT (0 << 30) +#define MOD_OPCLK_CDCLK_IN (1 << 30) +#define MOD_OPCLK_BCLK_OUT (2 << 30) +#define MOD_OPCLK_PCLK (3 << 30) +#define MOD_OPCLK_MASK (3 << 30) +#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ + +#define MOD_BLCS_SHIFT 26 +#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT) +#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT) +#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT) +#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT) +#define MOD_BLCP_SHIFT 24 +#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT) +#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT) +#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT) +#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT) + +#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ +#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ +#define MOD_C1DD_HHALF (1 << 19) +#define MOD_C1DD_LHALF (1 << 18) +#define MOD_DC2_EN (1 << 17) +#define MOD_DC1_EN (1 << 16) +#define MOD_BLC_16BIT (0 << 13) +#define MOD_BLC_8BIT (1 << 13) +#define MOD_BLC_24BIT (2 << 13) +#define MOD_BLC_MASK (3 << 13) + +#define MOD_IMS_SYSMUX (1 << 10) +#define MOD_SLAVE (1 << 11) +#define MOD_TXONLY (0 << 8) +#define MOD_RXONLY (1 << 8) +#define MOD_TXRX (2 << 8) +#define MOD_MASK (3 << 8) +#define MOD_LR_LLOW (0 << 7) +#define MOD_LR_RLOW (1 << 7) +#define MOD_SDF_IIS (0 << 5) +#define MOD_SDF_MSB (1 << 5) +#define MOD_SDF_LSB (2 << 5) +#define MOD_SDF_MASK (3 << 5) +#define MOD_RCLK_256FS (0 << 3) +#define MOD_RCLK_512FS (1 << 3) +#define MOD_RCLK_384FS (2 << 3) +#define MOD_RCLK_768FS (3 << 3) +#define MOD_RCLK_MASK (3 << 3) +#define MOD_BCLK_32FS (0 << 1) +#define MOD_BCLK_48FS (1 << 1) +#define MOD_BCLK_16FS (2 << 1) +#define MOD_BCLK_24FS (3 << 1) +#define MOD_BCLK_MASK (3 << 1) +#define MOD_8BIT (1 << 0) + +#define MOD_CDCLKCON (1 << 12) + +#define PSR_PSREN (1 << 15) + +#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf) +#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf) + +#define FIC_TXFLUSH (1 << 15) +#define FIC_RXFLUSH (1 << 7) + +#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf) +#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f) + +#define AHB_INTENLVL0 (1 << 24) +#define AHB_LVL0INT (1 << 20) +#define AHB_CLRLVL0INT (1 << 16) +#define AHB_DMARLD (1 << 5) +#define AHB_INTMASK (1 << 3) +#define AHB_DMAEN (1 << 0) +#define AHB_LVLINTMASK (0xf << 20) + +#define I2SSIZE_TRNMSK (0xffff) +#define I2SSIZE_SHIFT (16) + +#endif /* __SND_SOC_SAMSUNG_I2S_REGS_H */ + + diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 992a732b521..1568eea31f4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -22,109 +22,7 @@ #include "dma.h" #include "i2s.h" - -#define I2SCON 0x0 -#define I2SMOD 0x4 -#define I2SFIC 0x8 -#define I2SPSR 0xc -#define I2STXD 0x10 -#define I2SRXD 0x14 -#define I2SFICS 0x18 -#define I2STXDS 0x1c - -#define CON_RSTCLR (1 << 31) -#define CON_FRXOFSTATUS (1 << 26) -#define CON_FRXORINTEN (1 << 25) -#define CON_FTXSURSTAT (1 << 24) -#define CON_FTXSURINTEN (1 << 23) -#define CON_TXSDMA_PAUSE (1 << 20) -#define CON_TXSDMA_ACTIVE (1 << 18) - -#define CON_FTXURSTATUS (1 << 17) -#define CON_FTXURINTEN (1 << 16) -#define CON_TXFIFO2_EMPTY (1 << 15) -#define CON_TXFIFO1_EMPTY (1 << 14) -#define CON_TXFIFO2_FULL (1 << 13) -#define CON_TXFIFO1_FULL (1 << 12) - -#define CON_LRINDEX (1 << 11) -#define CON_TXFIFO_EMPTY (1 << 10) -#define CON_RXFIFO_EMPTY (1 << 9) -#define CON_TXFIFO_FULL (1 << 8) -#define CON_RXFIFO_FULL (1 << 7) -#define CON_TXDMA_PAUSE (1 << 6) -#define CON_RXDMA_PAUSE (1 << 5) -#define CON_TXCH_PAUSE (1 << 4) -#define CON_RXCH_PAUSE (1 << 3) -#define CON_TXDMA_ACTIVE (1 << 2) -#define CON_RXDMA_ACTIVE (1 << 1) -#define CON_ACTIVE (1 << 0) - -#define MOD_OPCLK_CDCLK_OUT (0 << 30) -#define MOD_OPCLK_CDCLK_IN (1 << 30) -#define MOD_OPCLK_BCLK_OUT (2 << 30) -#define MOD_OPCLK_PCLK (3 << 30) -#define MOD_OPCLK_MASK (3 << 30) -#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ - -#define MOD_BLCS_SHIFT 26 -#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT) -#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT) -#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT) -#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT) -#define MOD_BLCP_SHIFT 24 -#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT) -#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT) -#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT) -#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT) - -#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ -#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ -#define MOD_C1DD_HHALF (1 << 19) -#define MOD_C1DD_LHALF (1 << 18) -#define MOD_DC2_EN (1 << 17) -#define MOD_DC1_EN (1 << 16) -#define MOD_BLC_16BIT (0 << 13) -#define MOD_BLC_8BIT (1 << 13) -#define MOD_BLC_24BIT (2 << 13) -#define MOD_BLC_MASK (3 << 13) - -#define MOD_IMS_SYSMUX (1 << 10) -#define MOD_SLAVE (1 << 11) -#define MOD_TXONLY (0 << 8) -#define MOD_RXONLY (1 << 8) -#define MOD_TXRX (2 << 8) -#define MOD_MASK (3 << 8) -#define MOD_LR_LLOW (0 << 7) -#define MOD_LR_RLOW (1 << 7) -#define MOD_SDF_IIS (0 << 5) -#define MOD_SDF_MSB (1 << 5) -#define MOD_SDF_LSB (2 << 5) -#define MOD_SDF_MASK (3 << 5) -#define MOD_RCLK_256FS (0 << 3) -#define MOD_RCLK_512FS (1 << 3) -#define MOD_RCLK_384FS (2 << 3) -#define MOD_RCLK_768FS (3 << 3) -#define MOD_RCLK_MASK (3 << 3) -#define MOD_BCLK_32FS (0 << 1) -#define MOD_BCLK_48FS (1 << 1) -#define MOD_BCLK_16FS (2 << 1) -#define MOD_BCLK_24FS (3 << 1) -#define MOD_BCLK_MASK (3 << 1) -#define MOD_8BIT (1 << 0) - -#define MOD_CDCLKCON (1 << 12) - -#define PSR_PSREN (1 << 15) - -#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf) -#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf) - -#define FIC_TXFLUSH (1 << 15) -#define FIC_RXFLUSH (1 << 7) -#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf) -#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f) +#include "i2s-regs.h" #define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index e7c1009a1e1..45fbe2b3727 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -8,6 +8,7 @@ */ #include "../codecs/wm8994.h" +#include <sound/pcm_params.h> /* * Default CFG switch settings to use this driver: @@ -44,7 +45,9 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, int ret; /* AIF1CLK should be >=3MHz for optimal performance */ - if (params_rate(params) == 8000 || params_rate(params) == 11025) + if (params_format(params) == SNDRV_PCM_FORMAT_S24_LE) + pll_out = params_rate(params) * 384; + else if (params_rate(params) == 8000 || params_rate(params) == 11025) pll_out = params_rate(params) * 512; else pll_out = params_rate(params) * 256; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c new file mode 100644 index 00000000000..5f211168548 --- /dev/null +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -0,0 +1,176 @@ +/* + * sound/soc/samsung/smdk_wm8994pcm.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.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. + */ +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "../codecs/wm8994.h" +#include "dma.h" +#include "pcm.h" + +/* + * Board Settings: + * o '1' means 'ON' + * o '0' means 'OFF' + * o 'X' means 'Don't care' + * + * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111 + */ + +/* + * Configure audio route as :- + * $ amixer sset 'DAC1' on,on + * $ amixer sset 'Right Headphone Mux' 'DAC' + * $ amixer sset 'Left Headphone Mux' 'DAC' + * $ amixer sset 'DAC1R Mixer AIF1.1' on + * $ amixer sset 'DAC1L Mixer AIF1.1' on + * $ amixer sset 'IN2L' on + * $ amixer sset 'IN2L PGA IN2LN' on + * $ amixer sset 'MIXINL IN2L' on + * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on + * $ amixer sset 'IN2R' on + * $ amixer sset 'IN2R PGA IN2RN' on + * $ amixer sset 'MIXINR IN2R' on + * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on + */ + +/* SMDK has a 16.9344MHZ crystal attached to WM8994 */ +#define SMDK_WM8994_FREQ 16934400 + +static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + 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; + unsigned long mclk_freq; + int rfs, ret; + + switch(params_rate(params)) { + case 8000: + rfs = 512; + break; + default: + dev_err(cpu_dai->dev, "%s:%d Sampling Rate %u not supported!\n", + __func__, __LINE__, params_rate(params)); + return -EINVAL; + } + + mclk_freq = params_rate(params) * rfs; + + /* Set the codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* Set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + SMDK_WM8994_FREQ, mclk_freq); + if (ret < 0) + return ret; + + /* Set PCM source clock on CPU */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set SCLK_DIV for making bclk */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops smdk_wm8994_pcm_ops = { + .hw_params = smdk_wm8994_pcm_hw_params, +}; + +static struct snd_soc_dai_link smdk_dai[] = { + { + .name = "WM8994 PAIF PCM", + .stream_name = "Primary PCM", + .cpu_dai_name = "samsung-pcm.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm8994-codec", + .ops = &smdk_wm8994_pcm_ops, + }, +}; + +static struct snd_soc_card smdk_pcm = { + .name = "SMDK-PCM", + .dai_link = smdk_dai, + .num_links = 1, +}; + +static int __devinit snd_smdk_probe(struct platform_device *pdev) +{ + int ret = 0; + + smdk_pcm.dev = &pdev->dev; + ret = snd_soc_register_card(&smdk_pcm); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + return 0; +} + +static int __devexit snd_smdk_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&smdk_pcm); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_smdk_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "samsung-smdk-pcm", + }, + .probe = snd_smdk_probe, + .remove = __devexit_p(snd_smdk_remove), +}; + +static int __init smdk_audio_init(void) +{ + return platform_driver_register(&snd_smdk_driver); +} + +module_init(smdk_audio_init); + +static void __exit smdk_audio_exit(void) +{ + platform_driver_unregister(&snd_smdk_driver); +} + +module_exit(smdk_audio_exit); + +MODULE_AUTHOR("Sangbeom Kim, <sbkim73@samsung.com>"); +MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 360a333cb7c..d6dee4d0203 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -20,24 +20,29 @@ #define WM8915_HPSEL_GPIO 214 static int speyside_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_STANDBY: - ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK1, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK1, + ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK2, 0, 0, 0); if (ret < 0) { pr_err("Failed to stop FLL\n"); return ret; } + break; default: break; @@ -46,6 +51,45 @@ static int speyside_set_bias_level(struct snd_soc_card *card, return 0; } +static int speyside_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + WM8915_FLL_MCLK2, + 32768, 48000 * 256); + if (ret < 0) { + pr_err("Failed to start FLL\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8915_SYSCLK_FLL, + 48000 * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + } + break; + + default: + break; + } + + card->dapm.bias_level = level; + + return 0; +} + static int speyside_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -66,16 +110,6 @@ static int speyside_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK1, - 32768, 256 * 48000); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_FLL, - 256 * 48000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - return 0; } @@ -127,7 +161,7 @@ static int speyside_wm8915_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; int ret; - ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0); + ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK2, 32768, 0); if (ret < 0) return ret; @@ -267,6 +301,7 @@ static struct snd_soc_card speyside = { .num_configs = ARRAY_SIZE(speyside_codec_conf), .set_bias_level = speyside_set_bias_level, + .set_bias_level_post = speyside_set_bias_level_post, .controls = controls, .num_controls = ARRAY_SIZE(controls), diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c new file mode 100644 index 00000000000..8ac42bf8209 --- /dev/null +++ b/sound/soc/samsung/speyside_wm8962.c @@ -0,0 +1,264 @@ +/* + * Speyside with WM8962 audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * 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. + */ + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/gpio.h> + +#include "../codecs/wm8962.h" + +static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 32768, + 44100 * 256); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8962_SYSCLK_FLL, + 44100 * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n"); + return ret; + } + } + break; + + default: + break; + } + + return 0; +} + +static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops speyside_wm8962_ops = { + .hw_params = speyside_wm8962_hw_params, +}; + +static struct snd_soc_dai_link speyside_wm8962_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm8962", + .platform_name = "samsung-audio", + .codec_name = "wm8962.1-001a", + .ops = &speyside_wm8962_ops, + }, +}; + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Main Speaker"), +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_MIC("DMIC", NULL), + + SND_SOC_DAPM_SPK("Main Speaker", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Headphone", NULL, "HPOUTL" }, + { "Headphone", NULL, "HPOUTR" }, + + { "Main Speaker", NULL, "SPKOUTL" }, + { "Main Speaker", NULL, "SPKOUTR" }, + + { "MICBIAS", NULL, "Headset Mic" }, + { "IN4L", NULL, "MICBIAS" }, + { "IN4R", NULL, "MICBIAS" }, + + { "MICBIAS", NULL, "DMIC" }, + { "DMICDAT", NULL, "MICBIAS" }, +}; + +static struct snd_soc_jack speyside_wm8962_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin speyside_wm8962_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int speyside_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &speyside_wm8962_headset); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&speyside_wm8962_headset, + ARRAY_SIZE(speyside_wm8962_headset_pins), + speyside_wm8962_headset_pins); + if (ret) + return ret; + + wm8962_mic_detect(codec, &speyside_wm8962_headset); + + return 0; +} + +static struct snd_soc_card speyside_wm8962 = { + .name = "Speyside WM8962", + .dai_link = speyside_wm8962_dai, + .num_links = ARRAY_SIZE(speyside_wm8962_dai), + + .set_bias_level = speyside_wm8962_set_bias_level, + .set_bias_level_post = speyside_wm8962_set_bias_level_post, + + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), + + .late_probe = speyside_wm8962_late_probe, +}; + +static __devinit int speyside_wm8962_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &speyside_wm8962; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit speyside_wm8962_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver speyside_wm8962_driver = { + .driver = { + .name = "speyside-wm8962", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = speyside_wm8962_probe, + .remove = __devexit_p(speyside_wm8962_remove), +}; + +static int __init speyside_wm8962_audio_init(void) +{ + return platform_driver_register(&speyside_wm8962_driver); +} +module_init(speyside_wm8962_audio_init); + +static void __exit speyside_wm8962_audio_exit(void) +{ + platform_driver_unregister(&speyside_wm8962_driver); +} +module_exit(speyside_wm8962_audio_exit); + +MODULE_DESCRIPTION("Speyside WM8962 audio support"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:speyside-wm8962"); diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index c326d29992f..db74005f37c 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -327,10 +327,10 @@ static void camelot_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_pcm *pcm = rtd->pcm; + /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel * in MMAP mode (i.e. aplay -M) */ diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4a9da6b5f4e..8e112ccffb1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -118,10 +118,38 @@ typedef int (*set_rate_func)(struct device *dev, int is_porta, int rate, int ena /* * FSI driver use below type name for variable * - * xxx_len : data length - * xxx_width : data width - * xxx_offset : data offset * xxx_num : number of data + * xxx_pos : position of data + * xxx_capa : capacity of data + */ + +/* + * period/frame/sample image + * + * ex) PCM (2ch) + * + * period pos period pos + * [n] [n + 1] + * |<-------------------- period--------------------->| + * ==|============================================ ... =|== + * | | + * ||<----- frame ----->|<------ frame ----->| ... | + * |+--------------------+--------------------+- ... | + * ||[ sample ][ sample ]|[ sample ][ sample ]| ... | + * |+--------------------+--------------------+- ... | + * ==|============================================ ... =|== + */ + +/* + * FSI FIFO image + * + * | | + * | | + * | [ sample ] | + * | [ sample ] | + * | [ sample ] | + * | [ sample ] | + * --> go to codecs */ /* @@ -131,12 +159,11 @@ typedef int (*set_rate_func)(struct device *dev, int is_porta, int rate, int ena struct fsi_stream { struct snd_pcm_substream *substream; - int fifo_max_num; - - int buff_offset; - int buff_len; - int period_len; - int period_num; + int fifo_sample_capa; /* sample capacity of FSI FIFO */ + int buff_sample_capa; /* sample capacity of ALSA buffer */ + int buff_sample_pos; /* sample position of ALSA buffer */ + int period_samples; /* sample number / 1 period */ + int period_pos; /* current period position */ int uerr_num; int oerr_num; @@ -149,17 +176,14 @@ struct fsi_priv { struct fsi_stream playback; struct fsi_stream capture; + u32 do_fmt; + u32 di_fmt; + int chan_num:16; int clk_master:1; + int spdif:1; long rate; - - /* for suspend/resume */ - u32 saved_do_fmt; - u32 saved_di_fmt; - u32 saved_ckg1; - u32 saved_ckg2; - u32 saved_out_sel; }; struct fsi_core { @@ -180,14 +204,6 @@ struct fsi_master { struct fsi_core *core; struct sh_fsi_platform_info *info; spinlock_t lock; - - /* for suspend/resume */ - u32 saved_a_mclk; - u32 saved_b_mclk; - u32 saved_iemsk; - u32 saved_imsk; - u32 saved_clk_rst; - u32 saved_soft_rst; }; /* @@ -271,6 +287,11 @@ static int fsi_is_port_a(struct fsi_priv *fsi) return fsi->master->base == fsi->base; } +static int fsi_is_spdif(struct fsi_priv *fsi) +{ + return fsi->spdif; +} + static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -342,28 +363,59 @@ static u32 fsi_get_port_shift(struct fsi_priv *fsi, int is_play) return shift; } +static int fsi_frame2sample(struct fsi_priv *fsi, int frames) +{ + return frames * fsi->chan_num; +} + +static int fsi_sample2frame(struct fsi_priv *fsi, int samples) +{ + return samples / fsi->chan_num; +} + +static int fsi_stream_is_working(struct fsi_priv *fsi, + int is_play) +{ + struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct fsi_master *master = fsi_get_master(fsi); + unsigned long flags; + int ret; + + spin_lock_irqsave(&master->lock, flags); + ret = !!io->substream; + spin_unlock_irqrestore(&master->lock, flags); + + return ret; +} + static void fsi_stream_push(struct fsi_priv *fsi, int is_play, - struct snd_pcm_substream *substream, - u32 buffer_len, - u32 period_len) + struct snd_pcm_substream *substream) { struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsi_master *master = fsi_get_master(fsi); + unsigned long flags; + spin_lock_irqsave(&master->lock, flags); io->substream = substream; - io->buff_len = buffer_len; - io->buff_offset = 0; - io->period_len = period_len; - io->period_num = 0; + io->buff_sample_capa = fsi_frame2sample(fsi, runtime->buffer_size); + io->buff_sample_pos = 0; + io->period_samples = fsi_frame2sample(fsi, runtime->period_size); + io->period_pos = 0; io->oerr_num = -1; /* ignore 1st err */ io->uerr_num = -1; /* ignore 1st err */ + spin_unlock_irqrestore(&master->lock, flags); } static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) { struct fsi_stream *io = fsi_get_stream(fsi, is_play); struct snd_soc_dai *dai = fsi_get_dai(io->substream); + struct fsi_master *master = fsi_get_master(fsi); + unsigned long flags; + spin_lock_irqsave(&master->lock, flags); if (io->oerr_num > 0) dev_err(dai->dev, "over_run = %d\n", io->oerr_num); @@ -372,47 +424,27 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) dev_err(dai->dev, "under_run = %d\n", io->uerr_num); io->substream = NULL; - io->buff_len = 0; - io->buff_offset = 0; - io->period_len = 0; - io->period_num = 0; + io->buff_sample_capa = 0; + io->buff_sample_pos = 0; + io->period_samples = 0; + io->period_pos = 0; io->oerr_num = 0; io->uerr_num = 0; + spin_unlock_irqrestore(&master->lock, flags); } -static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play) +static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, int is_play) { u32 status; - int data_num; + int frames; status = is_play ? fsi_reg_read(fsi, DOFF_ST) : fsi_reg_read(fsi, DIFF_ST); - data_num = 0x1ff & (status >> 8); - data_num *= fsi->chan_num; - - return data_num; -} - -static int fsi_len2num(int len, int width) -{ - return len / width; -} - -#define fsi_num2offset(a, b) fsi_num2len(a, b) -static int fsi_num2len(int num, int width) -{ - return num * width; -} - -static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play) -{ - struct fsi_stream *io = fsi_get_stream(fsi, is_play); - struct snd_pcm_substream *substream = io->substream; - struct snd_pcm_runtime *runtime = substream->runtime; + frames = 0x1ff & (status >> 8); - return frames_to_bytes(runtime, 1) / fsi->chan_num; + return fsi_frame2sample(fsi, frames); } static void fsi_count_fifo_err(struct fsi_priv *fsi) @@ -444,8 +476,10 @@ static u8 *fsi_dma_get_area(struct fsi_priv *fsi, int stream) { int is_play = fsi_stream_is_play(stream); struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct snd_pcm_runtime *runtime = io->substream->runtime; - return io->substream->runtime->dma_area + io->buff_offset; + return runtime->dma_area + + samples_to_bytes(runtime, io->buff_sample_pos); } static void fsi_dma_soft_push16(struct fsi_priv *fsi, int num) @@ -559,37 +593,94 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable) /* * clock function */ -#define fsi_module_init(m, d) __fsi_module_clk_ctrl(m, d, 1) -#define fsi_module_kill(m, d) __fsi_module_clk_ctrl(m, d, 0) -static void __fsi_module_clk_ctrl(struct fsi_master *master, - struct device *dev, - int enable) +static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, + long rate, int enable) { - pm_runtime_get_sync(dev); + struct fsi_master *master = fsi_get_master(fsi); + set_rate_func set_rate = fsi_get_info_set_rate(master); + int fsi_ver = master->core->ver; + int ret; - if (enable) { - /* enable only SR */ - fsi_master_mask_set(master, SOFT_RST, FSISR, FSISR); - fsi_master_mask_set(master, SOFT_RST, PASR | PBSR, 0); - } else { - /* clear all registers */ - fsi_master_mask_set(master, SOFT_RST, FSISR, 0); + ret = set_rate(dev, fsi_is_port_a(fsi), rate, enable); + if (ret < 0) /* error */ + return ret; + + if (!enable) + return 0; + + if (ret > 0) { + u32 data = 0; + + switch (ret & SH_FSI_ACKMD_MASK) { + default: + /* FALL THROUGH */ + case SH_FSI_ACKMD_512: + data |= (0x0 << 12); + break; + case SH_FSI_ACKMD_256: + data |= (0x1 << 12); + break; + case SH_FSI_ACKMD_128: + data |= (0x2 << 12); + break; + case SH_FSI_ACKMD_64: + data |= (0x3 << 12); + break; + case SH_FSI_ACKMD_32: + if (fsi_ver < 2) + dev_err(dev, "unsupported ACKMD\n"); + else + data |= (0x4 << 12); + break; + } + + switch (ret & SH_FSI_BPFMD_MASK) { + default: + /* FALL THROUGH */ + case SH_FSI_BPFMD_32: + data |= (0x0 << 8); + break; + case SH_FSI_BPFMD_64: + data |= (0x1 << 8); + break; + case SH_FSI_BPFMD_128: + data |= (0x2 << 8); + break; + case SH_FSI_BPFMD_256: + data |= (0x3 << 8); + break; + case SH_FSI_BPFMD_512: + data |= (0x4 << 8); + break; + case SH_FSI_BPFMD_16: + if (fsi_ver < 2) + dev_err(dev, "unsupported ACKMD\n"); + else + data |= (0x7 << 8); + break; + } + + fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data); + udelay(10); + ret = 0; } - pm_runtime_put_sync(dev); + return ret; } -#define fsi_port_start(f) __fsi_port_clk_ctrl(f, 1) -#define fsi_port_stop(f) __fsi_port_clk_ctrl(f, 0) -static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int enable) +#define fsi_port_start(f, i) __fsi_port_clk_ctrl(f, i, 1) +#define fsi_port_stop(f, i) __fsi_port_clk_ctrl(f, i, 0) +static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable) { struct fsi_master *master = fsi_get_master(fsi); - u32 soft = fsi_is_port_a(fsi) ? PASR : PBSR; u32 clk = fsi_is_port_a(fsi) ? CRA : CRB; - int is_master = fsi_is_clk_master(fsi); - fsi_master_mask_set(master, SOFT_RST, soft, (enable) ? soft : 0); - if (is_master) + if (enable) + fsi_irq_enable(fsi, is_play); + else + fsi_irq_disable(fsi, is_play); + + if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } @@ -598,18 +689,19 @@ static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int enable) */ static void fsi_fifo_init(struct fsi_priv *fsi, int is_play, - struct snd_soc_dai *dai) + struct device *dev) { struct fsi_master *master = fsi_get_master(fsi); struct fsi_stream *io = fsi_get_stream(fsi, is_play); u32 shift, i; + int frame_capa; /* get on-chip RAM capacity */ shift = fsi_master_read(master, FIFO_SZ); shift >>= fsi_get_port_shift(fsi, is_play); shift &= FIFO_SZ_MASK; - io->fifo_max_num = 256 << shift; - dev_dbg(dai->dev, "fifo = %d words\n", io->fifo_max_num); + frame_capa = 256 << shift; + dev_dbg(dev, "fifo = %d words\n", frame_capa); /* * The maximum number of sample data varies depending @@ -631,9 +723,11 @@ static void fsi_fifo_init(struct fsi_priv *fsi, * 8 channels: 32 ( 32 x 8 = 256) */ for (i = 1; i < fsi->chan_num; i <<= 1) - io->fifo_max_num >>= 1; - dev_dbg(dai->dev, "%d channel %d store\n", - fsi->chan_num, io->fifo_max_num); + frame_capa >>= 1; + dev_dbg(dev, "%d channel %d store\n", + fsi->chan_num, frame_capa); + + io->fifo_sample_capa = fsi_frame2sample(fsi, frame_capa); /* * set interrupt generation factor @@ -654,10 +748,10 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) struct snd_pcm_substream *substream = NULL; int is_play = fsi_stream_is_play(stream); struct fsi_stream *io = fsi_get_stream(fsi, is_play); - int data_residue_num; - int data_num; - int data_num_max; - int ch_width; + int sample_residues; + int sample_width; + int samples; + int samples_max; int over_period; void (*fn)(struct fsi_priv *fsi, int size); @@ -673,36 +767,35 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) /* FSI FIFO has limit. * So, this driver can not send periods data at a time */ - if (io->buff_offset >= - fsi_num2offset(io->period_num + 1, io->period_len)) { + if (io->buff_sample_pos >= + io->period_samples * (io->period_pos + 1)) { over_period = 1; - io->period_num = (io->period_num + 1) % runtime->periods; + io->period_pos = (io->period_pos + 1) % runtime->periods; - if (0 == io->period_num) - io->buff_offset = 0; + if (0 == io->period_pos) + io->buff_sample_pos = 0; } - /* get 1 channel data width */ - ch_width = fsi_get_frame_width(fsi, is_play); + /* get 1 sample data width */ + sample_width = samples_to_bytes(runtime, 1); - /* get residue data number of alsa */ - data_residue_num = fsi_len2num(io->buff_len - io->buff_offset, - ch_width); + /* get number of residue samples */ + sample_residues = io->buff_sample_capa - io->buff_sample_pos; if (is_play) { /* * for play-back * - * data_num_max : number of FSI fifo free space - * data_num : number of ALSA residue data + * samples_max : number of FSI fifo free samples space + * samples : number of ALSA residue samples */ - data_num_max = io->fifo_max_num * fsi->chan_num; - data_num_max -= fsi_get_fifo_data_num(fsi, is_play); + samples_max = io->fifo_sample_capa; + samples_max -= fsi_get_current_fifo_samples(fsi, is_play); - data_num = data_residue_num; + samples = sample_residues; - switch (ch_width) { + switch (sample_width) { case 2: fn = fsi_dma_soft_push16; break; @@ -716,13 +809,13 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) /* * for capture * - * data_num_max : number of ALSA free space - * data_num : number of data in FSI fifo + * samples_max : number of ALSA free samples space + * samples : number of samples in FSI fifo */ - data_num_max = data_residue_num; - data_num = fsi_get_fifo_data_num(fsi, is_play); + samples_max = sample_residues; + samples = fsi_get_current_fifo_samples(fsi, is_play); - switch (ch_width) { + switch (sample_width) { case 2: fn = fsi_dma_soft_pop16; break; @@ -734,12 +827,12 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) } } - data_num = min(data_num, data_num_max); + samples = min(samples, samples_max); - fn(fsi, data_num); + fn(fsi, samples); - /* update buff_offset */ - io->buff_offset += fsi_num2offset(data_num, ch_width); + /* update buff_sample_pos */ + io->buff_sample_pos += samples; if (over_period) snd_pcm_period_elapsed(substream); @@ -788,16 +881,20 @@ static irqreturn_t fsi_interrupt(int irq, void *data) * dai ops */ -static int fsi_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int fsi_hw_startup(struct fsi_priv *fsi, + int is_play, + struct device *dev) { - struct fsi_priv *fsi = fsi_get_priv(substream); u32 flags = fsi_get_info_flags(fsi); - u32 data; - int is_play = fsi_is_play(substream); + u32 data = 0; - pm_runtime_get_sync(dai->dev); + pm_runtime_get_sync(dev); + /* clock setting */ + if (fsi_is_clk_master(fsi)) + data = DIMD | DOMD; + + fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data); /* clock inversion (CKG2) */ data = 0; @@ -812,54 +909,70 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, fsi_reg_write(fsi, CKG2, data); + /* set format */ + fsi_reg_write(fsi, DO_FMT, fsi->do_fmt); + fsi_reg_write(fsi, DI_FMT, fsi->di_fmt); + + /* spdif ? */ + if (fsi_is_spdif(fsi)) { + fsi_spdif_clk_ctrl(fsi, 1); + fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); + } + /* irq clear */ fsi_irq_disable(fsi, is_play); fsi_irq_clear_status(fsi); /* fifo init */ - fsi_fifo_init(fsi, is_play, dai); + fsi_fifo_init(fsi, is_play, dev); return 0; } -static void fsi_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static void fsi_hw_shutdown(struct fsi_priv *fsi, + int is_play, + struct device *dev) +{ + if (fsi_is_clk_master(fsi)) + fsi_set_master_clk(dev, fsi, fsi->rate, 0); + + pm_runtime_put_sync(dev); +} + +static int fsi_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); int is_play = fsi_is_play(substream); - struct fsi_master *master = fsi_get_master(fsi); - set_rate_func set_rate = fsi_get_info_set_rate(master); - fsi_irq_disable(fsi, is_play); + return fsi_hw_startup(fsi, is_play, dai->dev); +} - if (fsi_is_clk_master(fsi)) - set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0); +static void fsi_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get_priv(substream); + int is_play = fsi_is_play(substream); + fsi_hw_shutdown(fsi, is_play, dai->dev); fsi->rate = 0; - - pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - struct snd_pcm_runtime *runtime = substream->runtime; int is_play = fsi_is_play(substream); int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: - fsi_stream_push(fsi, is_play, substream, - frames_to_bytes(runtime, runtime->buffer_size), - frames_to_bytes(runtime, runtime->period_size)); + fsi_stream_push(fsi, is_play, substream); ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); - fsi_irq_enable(fsi, is_play); - fsi_port_start(fsi); + fsi_port_start(fsi, is_play); break; case SNDRV_PCM_TRIGGER_STOP: - fsi_port_stop(fsi); - fsi_irq_disable(fsi, is_play); + fsi_port_stop(fsi, is_play); fsi_stream_pop(fsi, is_play); break; } @@ -884,8 +997,8 @@ static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt) return -EINVAL; } - fsi_reg_write(fsi, DO_FMT, data); - fsi_reg_write(fsi, DI_FMT, data); + fsi->do_fmt = data; + fsi->di_fmt = data; return 0; } @@ -900,11 +1013,10 @@ static int fsi_set_fmt_spdif(struct fsi_priv *fsi) data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; fsi->chan_num = 2; - fsi_spdif_clk_ctrl(fsi, 1); - fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); + fsi->spdif = 1; - fsi_reg_write(fsi, DO_FMT, data); - fsi_reg_write(fsi, DI_FMT, data); + fsi->do_fmt = data; + fsi->di_fmt = data; return 0; } @@ -915,32 +1027,24 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct fsi_master *master = fsi_get_master(fsi); set_rate_func set_rate = fsi_get_info_set_rate(master); u32 flags = fsi_get_info_flags(fsi); - u32 data = 0; int ret; - pm_runtime_get_sync(dai->dev); - /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - data = DIMD | DOMD; fsi->clk_master = 1; break; case SND_SOC_DAIFMT_CBS_CFS: break; default: - ret = -EINVAL; - goto set_fmt_exit; + return -EINVAL; } if (fsi_is_clk_master(fsi) && !set_rate) { dev_err(dai->dev, "platform doesn't have set_rate\n"); - ret = -EINVAL; - goto set_fmt_exit; + return -EINVAL; } - fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data); - /* set format */ switch (flags & SH_FSI_FMT_MASK) { case SH_FSI_FMT_DAI: @@ -953,9 +1057,6 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = -EINVAL; } -set_fmt_exit: - pm_runtime_put_sync(dai->dev); - return ret; } @@ -964,79 +1065,19 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - struct fsi_master *master = fsi_get_master(fsi); - set_rate_func set_rate = fsi_get_info_set_rate(master); - int fsi_ver = master->core->ver; long rate = params_rate(params); int ret; if (!fsi_is_clk_master(fsi)) return 0; - ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1); - if (ret < 0) /* error */ + ret = fsi_set_master_clk(dai->dev, fsi, rate, 1); + if (ret < 0) return ret; fsi->rate = rate; - if (ret > 0) { - u32 data = 0; - - switch (ret & SH_FSI_ACKMD_MASK) { - default: - /* FALL THROUGH */ - case SH_FSI_ACKMD_512: - data |= (0x0 << 12); - break; - case SH_FSI_ACKMD_256: - data |= (0x1 << 12); - break; - case SH_FSI_ACKMD_128: - data |= (0x2 << 12); - break; - case SH_FSI_ACKMD_64: - data |= (0x3 << 12); - break; - case SH_FSI_ACKMD_32: - if (fsi_ver < 2) - dev_err(dai->dev, "unsupported ACKMD\n"); - else - data |= (0x4 << 12); - break; - } - - switch (ret & SH_FSI_BPFMD_MASK) { - default: - /* FALL THROUGH */ - case SH_FSI_BPFMD_32: - data |= (0x0 << 8); - break; - case SH_FSI_BPFMD_64: - data |= (0x1 << 8); - break; - case SH_FSI_BPFMD_128: - data |= (0x2 << 8); - break; - case SH_FSI_BPFMD_256: - data |= (0x3 << 8); - break; - case SH_FSI_BPFMD_512: - data |= (0x4 << 8); - break; - case SH_FSI_BPFMD_16: - if (fsi_ver < 2) - dev_err(dai->dev, "unsupported ACKMD\n"); - else - data |= (0x7 << 8); - break; - } - - fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data); - udelay(10); - ret = 0; - } return ret; - } static struct snd_soc_dai_ops fsi_dai_ops = { @@ -1097,16 +1138,14 @@ static int fsi_hw_free(struct snd_pcm_substream *substream) static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream)); - long location; + int samples_pos = io->buff_sample_pos - 1; - location = (io->buff_offset - 1); - if (location < 0) - location = 0; + if (samples_pos < 0) + samples_pos = 0; - return bytes_to_frames(runtime, location); + return fsi_sample2frame(fsi, samples_pos); } static struct snd_pcm_ops fsi_pcm_ops = { @@ -1129,10 +1168,10 @@ static void fsi_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int fsi_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_pcm *pcm = rtd->pcm; + /* * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel * in MMAP mode (i.e. aplay -M) @@ -1246,8 +1285,6 @@ static int fsi_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); - fsi_module_init(master, &pdev->dev); - ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, id_entry->name, master); if (ret) { @@ -1290,8 +1327,6 @@ static int fsi_remove(struct platform_device *pdev) master = dev_get_drvdata(&pdev->dev); - fsi_module_kill(master, &pdev->dev); - free_irq(master->irq, master); pm_runtime_disable(&pdev->dev); @@ -1305,53 +1340,43 @@ static int fsi_remove(struct platform_device *pdev) } static void __fsi_suspend(struct fsi_priv *fsi, - struct device *dev, - set_rate_func set_rate) + int is_play, + struct device *dev) { - fsi->saved_do_fmt = fsi_reg_read(fsi, DO_FMT); - fsi->saved_di_fmt = fsi_reg_read(fsi, DI_FMT); - fsi->saved_ckg1 = fsi_reg_read(fsi, CKG1); - fsi->saved_ckg2 = fsi_reg_read(fsi, CKG2); - fsi->saved_out_sel = fsi_reg_read(fsi, OUT_SEL); + if (!fsi_stream_is_working(fsi, is_play)) + return; - if (fsi_is_clk_master(fsi)) - set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 0); + fsi_port_stop(fsi, is_play); + fsi_hw_shutdown(fsi, is_play, dev); } static void __fsi_resume(struct fsi_priv *fsi, - struct device *dev, - set_rate_func set_rate) + int is_play, + struct device *dev) { - fsi_reg_write(fsi, DO_FMT, fsi->saved_do_fmt); - fsi_reg_write(fsi, DI_FMT, fsi->saved_di_fmt); - fsi_reg_write(fsi, CKG1, fsi->saved_ckg1); - fsi_reg_write(fsi, CKG2, fsi->saved_ckg2); - fsi_reg_write(fsi, OUT_SEL, fsi->saved_out_sel); + if (!fsi_stream_is_working(fsi, is_play)) + return; + + fsi_hw_startup(fsi, is_play, dev); + + if (fsi_is_clk_master(fsi) && fsi->rate) + fsi_set_master_clk(dev, fsi, fsi->rate, 1); + + fsi_port_start(fsi, is_play); - if (fsi_is_clk_master(fsi)) - set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 1); } static int fsi_suspend(struct device *dev) { struct fsi_master *master = dev_get_drvdata(dev); - set_rate_func set_rate = fsi_get_info_set_rate(master); - - pm_runtime_get_sync(dev); - - __fsi_suspend(&master->fsia, dev, set_rate); - __fsi_suspend(&master->fsib, dev, set_rate); + struct fsi_priv *fsia = &master->fsia; + struct fsi_priv *fsib = &master->fsib; - master->saved_a_mclk = fsi_core_read(master, a_mclk); - master->saved_b_mclk = fsi_core_read(master, b_mclk); - master->saved_iemsk = fsi_core_read(master, iemsk); - master->saved_imsk = fsi_core_read(master, imsk); - master->saved_clk_rst = fsi_master_read(master, CLK_RST); - master->saved_soft_rst = fsi_master_read(master, SOFT_RST); + __fsi_suspend(fsia, 1, dev); + __fsi_suspend(fsia, 0, dev); - fsi_module_kill(master, dev); - - pm_runtime_put_sync(dev); + __fsi_suspend(fsib, 1, dev); + __fsi_suspend(fsib, 0, dev); return 0; } @@ -1359,23 +1384,14 @@ static int fsi_suspend(struct device *dev) static int fsi_resume(struct device *dev) { struct fsi_master *master = dev_get_drvdata(dev); - set_rate_func set_rate = fsi_get_info_set_rate(master); - - pm_runtime_get_sync(dev); - - fsi_module_init(master, dev); + struct fsi_priv *fsia = &master->fsia; + struct fsi_priv *fsib = &master->fsib; - fsi_master_mask_set(master, SOFT_RST, 0xffff, master->saved_soft_rst); - fsi_master_mask_set(master, CLK_RST, 0xffff, master->saved_clk_rst); - fsi_core_mask_set(master, a_mclk, 0xffff, master->saved_a_mclk); - fsi_core_mask_set(master, b_mclk, 0xffff, master->saved_b_mclk); - fsi_core_mask_set(master, iemsk, 0xffff, master->saved_iemsk); - fsi_core_mask_set(master, imsk, 0xffff, master->saved_imsk); + __fsi_resume(fsia, 1, dev); + __fsi_resume(fsia, 0, dev); - __fsi_resume(&master->fsia, dev, set_rate); - __fsi_resume(&master->fsib, dev, set_rate); - - pm_runtime_put_sync(dev); + __fsi_resume(fsib, 1, dev); + __fsi_resume(fsib, 0, dev); return 0; } diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index a423babcf14..f8f681690a7 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -527,10 +527,11 @@ static snd_pcm_uframes_t siu_pcm_pointer_dma(struct snd_pcm_substream *ss) return bytes_to_frames(ss->runtime, ptr); } -static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int siu_pcm_new(struct snd_soc_pcm_runtime *rtd) { /* card->dev == socdev->dev, see snd_soc_new_pcms() */ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; struct siu_info *info = siu_i2s_data; struct platform_device *pdev = to_platform_device(card->dev); int ret; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 039b9532b27..d9f8aded51f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,422 +20,6 @@ #include <trace/events/asoc.h> -#ifdef CONFIG_SPI_MASTER -static int do_spi_write(void *control, const char *data, int len) -{ - struct spi_device *spi = control; - int ret; - - ret = spi_write(spi, data, len); - if (ret < 0) - return ret; - - return len; -} -#endif - -static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value, const void *data, int len) -{ - int ret; - - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - - ret = codec->hw_write(codec->control_data, data, len); - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; -} - -static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) -{ - int ret; - unsigned int val; - - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) - return -1; - return val; -} - -static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - -static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - -static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - return do_hw_write(codec, reg, value, data, 2); -} - -static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - reg &= 0xff; - data[0] = reg; - data[1] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 2); -} - -static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - -static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - - data[0] = reg; - data[1] = (value >> 8) & 0xff; - data[2] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 3); -} - -static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int do_i2c_read(struct snd_soc_codec *codec, - void *reg, int reglen, - void *data, int datalen) -{ - struct i2c_msg xfer[2]; - int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = reglen; - xfer[0].buf = reg; - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = datalen; - xfer[1].buf = data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret == 2) - return 0; - else if (ret < 0) - return ret; - else - return -EIO; -} -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_8_8_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 2); - if (ret < 0) - return 0; - return (data >> 8) | ((data & 0xff) << 8); -} -#else -#define snd_soc_8_16_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_i2c NULL -#endif - -static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - -static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - - data[0] = (reg >> 8) & 0xff; - data[1] = reg & 0xff; - data[2] = value; - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = cpu_to_be16(r); - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 2); - if (ret < 0) - return 0; - return be16_to_cpu(data); -} -#else -#define snd_soc_16_16_read_i2c NULL -#endif - -static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return do_hw_read(codec, reg); -} - -static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[4]; - - data[0] = (reg >> 8) & 0xff; - data[1] = reg & 0xff; - data[2] = (value >> 8) & 0xff; - data[3] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 4); -} - -/* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects - * including any leading register specific data. Any data written - * through this function will not go through the cache as it only - * handles writing to volatile or out of bounds registers. - */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, - const void *data, size_t len) -{ - int ret; - - /* To ensure that we don't get out of sync with the cache, check - * whether the base register is volatile or if we've directly asked - * to bypass the cache. Out of bounds registers are considered - * volatile. - */ - if (!codec->cache_bypass - && !snd_soc_codec_volatile_register(codec, reg) - && reg < codec->driver->reg_cache_size) - return -EINVAL; - - switch (codec->control_type) { -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - case SND_SOC_I2C: - ret = i2c_master_send(codec->control_data, data, len); - break; -#endif -#if defined(CONFIG_SPI_MASTER) - case SND_SOC_SPI: - ret = spi_write(codec->control_data, data, len); - break; -#endif - default: - BUG(); - } - - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; -} - -static struct { - int addr_bits; - int data_bits; - int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); -} io_types[] = { - { - .addr_bits = 4, .data_bits = 12, - .write = snd_soc_4_12_write, .read = snd_soc_4_12_read, - }, - { - .addr_bits = 7, .data_bits = 9, - .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, - }, - { - .addr_bits = 8, .data_bits = 8, - .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, - .i2c_read = snd_soc_8_8_read_i2c, - }, - { - .addr_bits = 8, .data_bits = 16, - .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, - .i2c_read = snd_soc_8_16_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 8, - .write = snd_soc_16_8_write, .read = snd_soc_16_8_read, - .i2c_read = snd_soc_16_8_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 16, - .write = snd_soc_16_16_write, .read = snd_soc_16_16_read, - .i2c_read = snd_soc_16_16_read_i2c, - }, -}; - -/** - * snd_soc_codec_set_cache_io: Set up standard I/O functions. - * - * @codec: CODEC to configure. - * @addr_bits: Number of bits of register address data. - * @data_bits: Number of bits of data per register. - * @control: Control bus used. - * - * Register formats are frequently shared between many I2C and SPI - * devices. In order to promote code reuse the ASoC core provides - * some standard implementations of CODEC read and write operations - * which can be set up using this function. - * - * The caller is responsible for allocating and initialising the - * actual cache. - * - * Note that at present this code cannot be used by CODECs with - * volatile registers. - */ -int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(io_types); i++) - if (io_types[i].addr_bits == addr_bits && - io_types[i].data_bits == data_bits) - break; - if (i == ARRAY_SIZE(io_types)) { - printk(KERN_ERR - "No I/O functions for %d bit address %d bit data\n", - addr_bits, data_bits); - return -EINVAL; - } - - codec->write = io_types[i].write; - codec->read = io_types[i].read; - codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; - - switch (control) { - case SND_SOC_I2C: -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - codec->hw_write = (hw_write_t)i2c_master_send; -#endif - if (io_types[i].i2c_read) - codec->hw_read = io_types[i].i2c_read; - - codec->control_data = container_of(codec->dev, - struct i2c_client, - dev); - break; - - case SND_SOC_SPI: -#ifdef CONFIG_SPI_MASTER - codec->hw_write = do_spi_write; -#endif - - codec->control_data = container_of(codec->dev, - struct spi_device, - dev); - break; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); - static bool snd_soc_set_cache_val(void *base, unsigned int idx, unsigned int val, unsigned int word_size) { @@ -483,31 +67,86 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, } struct snd_soc_rbtree_node { - struct rb_node node; - unsigned int reg; - unsigned int value; - unsigned int defval; + struct rb_node node; /* the actual rbtree node holding this block */ + unsigned int base_reg; /* base register handled by this block */ + unsigned int word_size; /* number of bytes needed to represent the register index */ + void *block; /* block of adjacent registers */ + unsigned int blklen; /* number of registers available in the block */ } __attribute__ ((packed)); struct snd_soc_rbtree_ctx { struct rb_root root; + struct snd_soc_rbtree_node *cached_rbnode; }; +static inline void snd_soc_rbtree_get_base_top_reg( + struct snd_soc_rbtree_node *rbnode, + unsigned int *base, unsigned int *top) +{ + *base = rbnode->base_reg; + *top = rbnode->base_reg + rbnode->blklen - 1; +} + +static unsigned int snd_soc_rbtree_get_register( + struct snd_soc_rbtree_node *rbnode, unsigned int idx) +{ + unsigned int val; + + switch (rbnode->word_size) { + case 1: { + u8 *p = rbnode->block; + val = p[idx]; + return val; + } + case 2: { + u16 *p = rbnode->block; + val = p[idx]; + return val; + } + default: + BUG(); + break; + } + return -1; +} + +static void snd_soc_rbtree_set_register(struct snd_soc_rbtree_node *rbnode, + unsigned int idx, unsigned int val) +{ + switch (rbnode->word_size) { + case 1: { + u8 *p = rbnode->block; + p[idx] = val; + break; + } + case 2: { + u16 *p = rbnode->block; + p[idx] = val; + break; + } + default: + BUG(); + break; + } +} + static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup( struct rb_root *root, unsigned int reg) { struct rb_node *node; struct snd_soc_rbtree_node *rbnode; + unsigned int base_reg, top_reg; node = root->rb_node; while (node) { rbnode = container_of(node, struct snd_soc_rbtree_node, node); - if (rbnode->reg < reg) - node = node->rb_left; - else if (rbnode->reg > reg) - node = node->rb_right; - else + snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); + if (reg >= base_reg && reg <= top_reg) return rbnode; + else if (reg > top_reg) + node = node->rb_right; + else if (reg < base_reg) + node = node->rb_left; } return NULL; @@ -518,19 +157,28 @@ static int snd_soc_rbtree_insert(struct rb_root *root, { struct rb_node **new, *parent; struct snd_soc_rbtree_node *rbnode_tmp; + unsigned int base_reg_tmp, top_reg_tmp; + unsigned int base_reg; parent = NULL; new = &root->rb_node; while (*new) { rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node, node); + /* base and top registers of the current rbnode */ + snd_soc_rbtree_get_base_top_reg(rbnode_tmp, &base_reg_tmp, + &top_reg_tmp); + /* base register of the rbnode to be added */ + base_reg = rbnode->base_reg; parent = *new; - if (rbnode_tmp->reg < rbnode->reg) - new = &((*new)->rb_left); - else if (rbnode_tmp->reg > rbnode->reg) - new = &((*new)->rb_right); - else + /* if this register has already been inserted, just return */ + if (base_reg >= base_reg_tmp && + base_reg <= top_reg_tmp) return 0; + else if (base_reg > top_reg_tmp) + new = &((*new)->rb_right); + else if (base_reg < base_reg_tmp) + new = &((*new)->rb_left); } /* insert the node into the rbtree */ @@ -545,58 +193,146 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) struct snd_soc_rbtree_ctx *rbtree_ctx; struct rb_node *node; struct snd_soc_rbtree_node *rbnode; - unsigned int val; + unsigned int regtmp; + unsigned int val, def; int ret; + int i; rbtree_ctx = codec->reg_cache; for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) { rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); - if (rbnode->value == rbnode->defval) - continue; - WARN_ON(codec->writable_register && - codec->writable_register(codec, rbnode->reg)); - ret = snd_soc_cache_read(codec, rbnode->reg, &val); - if (ret) - return ret; - codec->cache_bypass = 1; - ret = snd_soc_write(codec, rbnode->reg, val); - codec->cache_bypass = 0; - if (ret) - return ret; - dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - rbnode->reg, val); + for (i = 0; i < rbnode->blklen; ++i) { + regtmp = rbnode->base_reg + i; + WARN_ON(codec->writable_register && + codec->writable_register(codec, regtmp)); + val = snd_soc_rbtree_get_register(rbnode, i); + def = snd_soc_get_cache_val(codec->reg_def_copy, i, + rbnode->word_size); + if (val == def) + continue; + + codec->cache_bypass = 1; + ret = snd_soc_write(codec, regtmp, val); + codec->cache_bypass = 0; + if (ret) + return ret; + dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", + regtmp, val); + } } return 0; } +static int snd_soc_rbtree_insert_to_block(struct snd_soc_rbtree_node *rbnode, + unsigned int pos, unsigned int reg, + unsigned int value) +{ + u8 *blk; + + blk = krealloc(rbnode->block, + (rbnode->blklen + 1) * rbnode->word_size, GFP_KERNEL); + if (!blk) + return -ENOMEM; + + /* insert the register value in the correct place in the rbnode block */ + memmove(blk + (pos + 1) * rbnode->word_size, + blk + pos * rbnode->word_size, + (rbnode->blklen - pos) * rbnode->word_size); + + /* update the rbnode block, its size and the base register */ + rbnode->block = blk; + rbnode->blklen++; + if (!pos) + rbnode->base_reg = reg; + + snd_soc_rbtree_set_register(rbnode, pos, value); + return 0; +} + static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { struct snd_soc_rbtree_ctx *rbtree_ctx; - struct snd_soc_rbtree_node *rbnode; + struct snd_soc_rbtree_node *rbnode, *rbnode_tmp; + struct rb_node *node; + unsigned int val; + unsigned int reg_tmp; + unsigned int base_reg, top_reg; + unsigned int pos; + int i; + int ret; rbtree_ctx = codec->reg_cache; + /* look up the required register in the cached rbnode */ + rbnode = rbtree_ctx->cached_rbnode; + if (rbnode) { + snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); + if (reg >= base_reg && reg <= top_reg) { + reg_tmp = reg - base_reg; + val = snd_soc_rbtree_get_register(rbnode, reg_tmp); + if (val == value) + return 0; + snd_soc_rbtree_set_register(rbnode, reg_tmp, value); + return 0; + } + } + /* if we can't locate it in the cached rbnode we'll have + * to traverse the rbtree looking for it. + */ rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); if (rbnode) { - if (rbnode->value == value) + reg_tmp = reg - rbnode->base_reg; + val = snd_soc_rbtree_get_register(rbnode, reg_tmp); + if (val == value) return 0; - rbnode->value = value; + snd_soc_rbtree_set_register(rbnode, reg_tmp, value); + rbtree_ctx->cached_rbnode = rbnode; } else { /* bail out early, no need to create the rbnode yet */ if (!value) return 0; - /* - * for uninitialized registers whose value is changed - * from the default zero, create an rbnode and insert - * it into the tree. + /* look for an adjacent register to the one we are about to add */ + for (node = rb_first(&rbtree_ctx->root); node; + node = rb_next(node)) { + rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node); + for (i = 0; i < rbnode_tmp->blklen; ++i) { + reg_tmp = rbnode_tmp->base_reg + i; + if (abs(reg_tmp - reg) != 1) + continue; + /* decide where in the block to place our register */ + if (reg_tmp + 1 == reg) + pos = i + 1; + else + pos = i; + ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos, + reg, value); + if (ret) + return ret; + rbtree_ctx->cached_rbnode = rbnode_tmp; + return 0; + } + } + /* we did not manage to find a place to insert it in an existing + * block so create a new rbnode with a single register in its block. + * This block will get populated further if any other adjacent + * registers get modified in the future. */ rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL); if (!rbnode) return -ENOMEM; - rbnode->reg = reg; - rbnode->value = value; + rbnode->blklen = 1; + rbnode->base_reg = reg; + rbnode->word_size = codec->driver->reg_word_size; + rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size, + GFP_KERNEL); + if (!rbnode->block) { + kfree(rbnode); + return -ENOMEM; + } + snd_soc_rbtree_set_register(rbnode, 0, value); snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode); + rbtree_ctx->cached_rbnode = rbnode; } return 0; @@ -607,11 +343,28 @@ static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec, { struct snd_soc_rbtree_ctx *rbtree_ctx; struct snd_soc_rbtree_node *rbnode; + unsigned int base_reg, top_reg; + unsigned int reg_tmp; rbtree_ctx = codec->reg_cache; + /* look up the required register in the cached rbnode */ + rbnode = rbtree_ctx->cached_rbnode; + if (rbnode) { + snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg); + if (reg >= base_reg && reg <= top_reg) { + reg_tmp = reg - base_reg; + *value = snd_soc_rbtree_get_register(rbnode, reg_tmp); + return 0; + } + } + /* if we can't locate it in the cached rbnode we'll have + * to traverse the rbtree looking for it. + */ rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg); if (rbnode) { - *value = rbnode->value; + reg_tmp = reg - rbnode->base_reg; + *value = snd_soc_rbtree_get_register(rbnode, reg_tmp); + rbtree_ctx->cached_rbnode = rbnode; } else { /* uninitialized registers default to 0 */ *value = 0; @@ -637,6 +390,7 @@ static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node); next = rb_next(&rbtree_node->node); rb_erase(&rbtree_node->node, &rbtree_ctx->root); + kfree(rbtree_node->block); kfree(rbtree_node); } @@ -649,10 +403,9 @@ static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) { - struct snd_soc_rbtree_node *rbtree_node; struct snd_soc_rbtree_ctx *rbtree_ctx; - unsigned int val; unsigned int word_size; + unsigned int val; int i; int ret; @@ -662,32 +415,27 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) rbtree_ctx = codec->reg_cache; rbtree_ctx->root = RB_ROOT; + rbtree_ctx->cached_rbnode = NULL; if (!codec->reg_def_copy) return 0; - /* - * populate the rbtree with the initialized registers. All other - * registers will be inserted when they are first modified. - */ word_size = codec->driver->reg_word_size; for (i = 0; i < codec->driver->reg_cache_size; ++i) { - val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size); + val = snd_soc_get_cache_val(codec->reg_def_copy, i, + word_size); if (!val) continue; - rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL); - if (!rbtree_node) { - ret = -ENOMEM; - snd_soc_cache_exit(codec); - break; - } - rbtree_node->reg = i; - rbtree_node->value = val; - rbtree_node->defval = val; - snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node); + ret = snd_soc_rbtree_cache_write(codec, i, val); + if (ret) + goto err; } return 0; + +err: + snd_soc_cache_exit(codec); + return ret; } #ifdef CONFIG_SND_SOC_CACHE_LZO diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b194be09e74..e44267f6621 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -44,7 +44,6 @@ #define NAME_SIZE 32 -static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -58,7 +57,7 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -485,552 +484,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && - !rtd->dai_link->symmetric_rates) - return 0; - - /* This can happen if multiple streams are starting simultaneously - - * the second can need to get its constraints before the first has - * picked a rate. Complain and allow the application to carry on. - */ - if (!rtd->rate) { - dev_warn(&rtd->dev, - "Not enforcing symmetric_rates due to race\n"); - return 0; - } - - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } - - return 0; -} - -/* - * Called by ALSA when a PCM substream is opened, the runtime->hw record is - * then initialized and any private data can be allocated. This also calls - * startup for the cpu DAI, platform, machine and codec DAI. - */ -static int soc_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; - int ret = 0; - - mutex_lock(&pcm_mutex); - - /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->open) { - ret = platform->driver->ops->open(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); - goto platform_err; - } - } - - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); - goto codec_dai_err; - } - } - - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { - ret = rtd->dai_link->ops->startup(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); - goto machine_err; - } - } - - /* Check that the codec and cpu DAIs are compatible */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - runtime->hw.rate_min = - max(codec_dai_drv->playback.rate_min, - cpu_dai_drv->playback.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->playback.rate_max, - cpu_dai_drv->playback.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->playback.channels_min, - cpu_dai_drv->playback.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->playback.channels_max, - cpu_dai_drv->playback.channels_max); - runtime->hw.formats = - codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; - runtime->hw.rates = - codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; - if (codec_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->playback.rates; - if (cpu_dai_drv->playback.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->playback.rates; - } else { - runtime->hw.rate_min = - max(codec_dai_drv->capture.rate_min, - cpu_dai_drv->capture.rate_min); - runtime->hw.rate_max = - min(codec_dai_drv->capture.rate_max, - cpu_dai_drv->capture.rate_max); - runtime->hw.channels_min = - max(codec_dai_drv->capture.channels_min, - cpu_dai_drv->capture.channels_min); - runtime->hw.channels_max = - min(codec_dai_drv->capture.channels_max, - cpu_dai_drv->capture.channels_max); - runtime->hw.formats = - codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; - runtime->hw.rates = - codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; - if (codec_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai_drv->capture.rates; - if (cpu_dai_drv->capture.rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai_drv->capture.rates; - } - - ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); - if (!runtime->hw.rates) { - printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.formats) { - printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - if (!runtime->hw.channels_min || !runtime->hw.channels_max || - runtime->hw.channels_min > runtime->hw.channels_max) { - printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); - goto config_err; - } - - /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); - if (ret != 0) - goto config_err; - } - - pr_debug("asoc: %s <-> %s info:\n", - codec_dai->name, cpu_dai->name); - pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); - pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, - runtime->hw.channels_max); - pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, - runtime->hw.rate_max); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; - mutex_unlock(&pcm_mutex); - return 0; - -config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - -machine_err: - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - -codec_dai_err: - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - -platform_err: - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Power down the audio subsystem pmdown_time msecs after close is called. - * This is to ensure there are no pops or clicks in between any music tracks - * due to DAPM power cycling. - */ -static void close_delayed_work(struct work_struct *work) -{ - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - mutex_lock(&pcm_mutex); - - pr_debug("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); - - /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); -} - -/* - * Called by ALSA when a PCM substream is closed. Private data can be - * freed here. The cpu DAI, codec DAI, machine and platform are also - * shutdown. - */ -static int soc_codec_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - codec_dai->capture_active--; - } - - cpu_dai->active--; - codec_dai->active--; - codec->active--; - - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); - - if (cpu_dai->driver->ops->shutdown) - cpu_dai->driver->ops->shutdown(substream, cpu_dai); - - if (codec_dai->driver->ops->shutdown) - codec_dai->driver->ops->shutdown(substream, codec_dai); - - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) - rtd->dai_link->ops->shutdown(substream); - - if (platform->driver->ops && platform->driver->ops->close) - platform->driver->ops->close(substream); - cpu_dai->runtime = NULL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); - } else { - /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } - - mutex_unlock(&pcm_mutex); - return 0; -} - -/* - * Called by ALSA when the PCM substream is prepared, can set format, sample - * rate, etc. This function is non atomic and can be called multiple times, - * it can refer to the runtime info. - */ -static int soc_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { - ret = rtd->dai_link->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); - goto out; - } - } - - if (platform->driver->ops && platform->driver->ops->prepare) { - ret = platform->driver->ops->prepare(substream); - if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); - goto out; - } - } - - if (codec_dai->driver->ops->prepare) { - ret = codec_dai->driver->ops->prepare(substream, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); - goto out; - } - } - - if (cpu_dai->driver->ops->prepare) { - ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); - goto out; - } - } - - /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; - cancel_delayed_work(&rtd->delayed_work); - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - - snd_soc_dai_digital_mute(codec_dai, 0); - -out: - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Called by ALSA when the hardware params are set by application. This - * function can also be called multiple times and can allocate buffers - * (using snd_pcm_lib_* ). It's non-atomic. - */ -static int soc_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret = 0; - - mutex_lock(&pcm_mutex); - - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { - ret = rtd->dai_link->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); - goto out; - } - } - - if (codec_dai->driver->ops->hw_params) { - ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); - goto codec_err; - } - } - - if (cpu_dai->driver->ops->hw_params) { - ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); - if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); - goto interface_err; - } - } - - if (platform->driver->ops && platform->driver->ops->hw_params) { - ret = platform->driver->ops->hw_params(substream, params); - if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); - goto platform_err; - } - } - - rtd->rate = params_rate(params); - -out: - mutex_unlock(&pcm_mutex); - return ret; - -platform_err: - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - -interface_err: - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - -codec_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - mutex_unlock(&pcm_mutex); - return ret; -} - -/* - * Frees resources allocated by hw_params, can be called multiple times - */ -static int soc_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&pcm_mutex); - - /* apply codec digital mute */ - if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); - - /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) - rtd->dai_link->ops->hw_free(substream); - - /* free any DMA resources */ - if (platform->driver->ops && platform->driver->ops->hw_free) - platform->driver->ops->hw_free(substream); - - /* now free hw params for the DAIs */ - if (codec_dai->driver->ops->hw_free) - codec_dai->driver->ops->hw_free(substream, codec_dai); - - if (cpu_dai->driver->ops->hw_free) - cpu_dai->driver->ops->hw_free(substream, cpu_dai); - - mutex_unlock(&pcm_mutex); - return 0; -} - -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - if (codec_dai->driver->ops->trigger) { - ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); - if (ret < 0) - return ret; - } - - if (platform->driver->ops && platform->driver->ops->trigger) { - ret = platform->driver->ops->trigger(substream, cmd); - if (ret < 0) - return ret; - } - - if (cpu_dai->driver->ops->trigger) { - ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); - if (ret < 0) - return ret; - } - return 0; -} - -/* - * soc level wrapper for pointer callback - * If cpu_dai, codec_dai, platform driver has the delay callback, than - * the runtime->delay will be updated accordingly. - */ -static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t offset = 0; - snd_pcm_sframes_t delay = 0; - - if (platform->driver->ops && platform->driver->ops->pointer) - offset = platform->driver->ops->pointer(substream); - - if (cpu_dai->driver->ops->delay) - delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - - if (codec_dai->driver->ops->delay) - delay += codec_dai->driver->ops->delay(substream, codec_dai); - - if (platform->driver->delay) - delay += platform->driver->delay(substream, codec_dai); - - runtime->delay = delay; - - return offset; -} - -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_codec_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - #ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ int snd_soc_suspend(struct device *dev) @@ -1256,7 +709,7 @@ static void soc_resume_deferred(struct work_struct *work) int snd_soc_resume(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); - int i; + int i, ac97_control = 0; /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that @@ -1265,14 +718,15 @@ int snd_soc_resume(struct device *dev) */ for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - if (cpu_dai->driver->ac97_control) { - dev_dbg(dev, "Resuming AC97 immediately\n"); - soc_resume_deferred(&card->deferred_resume_work); - } else { - dev_dbg(dev, "Scheduling resume work\n"); - if (!schedule_work(&card->deferred_resume_work)) - dev_err(dev, "resume work item may be lost\n"); - } + ac97_control |= cpu_dai->driver->ac97_control; + } + if (ac97_control) { + dev_dbg(dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(dev, "resume work item may be lost\n"); } return 0; @@ -1393,7 +847,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num) +static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; @@ -1410,7 +864,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC DAI */ - if (codec_dai && codec_dai->probed) { + if (codec_dai && codec_dai->probed && + codec_dai->driver->remove_order == order) { if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) @@ -1421,7 +876,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the platform */ - if (platform && platform->probed) { + if (platform && platform->probed && + platform->driver->remove_order == order) { if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) @@ -1433,11 +889,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC */ - if (codec && codec->probed) + if (codec && codec->probed && + codec->driver->remove_order == order) soc_remove_codec(codec); /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed) { + if (cpu_dai && cpu_dai->probed && + cpu_dai->driver->remove_order == order) { if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) @@ -1451,11 +909,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) static void soc_remove_dai_links(struct snd_soc_card *card) { - int i; - - for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + int dai, order; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (dai = 0; dai < card->num_rtd; dai++) + soc_remove_dai_link(card, dai, order); + } card->num_rtd = 0; } @@ -1526,6 +986,52 @@ err_probe: return ret; } +static int soc_probe_platform(struct snd_soc_card *card, + struct snd_soc_platform *platform) +{ + int ret = 0; + const struct snd_soc_platform_driver *driver = platform->driver; + + platform->card = card; + platform->dapm.card = card; + + if (!try_module_get(platform->dev->driver->owner)) + return -ENODEV; + + if (driver->dapm_widgets) + snd_soc_dapm_new_controls(&platform->dapm, + driver->dapm_widgets, driver->num_dapm_widgets); + + if (driver->probe) { + ret = driver->probe(platform); + if (ret < 0) { + dev_err(platform->dev, + "asoc: failed to probe platform %s: %d\n", + platform->name, ret); + goto err_probe; + } + } + + if (driver->controls) + snd_soc_add_platform_controls(platform, driver->controls, + driver->num_controls); + if (driver->dapm_routes) + snd_soc_dapm_add_routes(&platform->dapm, driver->dapm_routes, + driver->num_dapm_routes); + + /* mark platform as probed and add to card platform list */ + platform->probed = 1; + list_add(&platform->card_list, &card->platform_dev_list); + list_add(&platform->dapm.list, &card->dapm_list); + + return 0; + +err_probe: + module_put(platform->dev->driver->owner); + + return ret; +} + static void rtd_release(struct device *dev) {} static int soc_post_component_init(struct snd_soc_card *card, @@ -1572,6 +1078,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; + mutex_init(&rtd->pcm_mutex); ret = device_register(&rtd->dev); if (ret < 0) { dev_err(card->dev, @@ -1596,7 +1103,7 @@ static int soc_post_component_init(struct snd_soc_card *card, return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num) +static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1605,7 +1112,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int ret; - dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); + dev_dbg(card->dev, "probe %s dai link %d late %d\n", + card->name, num, order); /* config components */ codec_dai->codec = codec; @@ -1617,7 +1125,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ - if (!cpu_dai->probed) { + if (!cpu_dai->probed && + cpu_dai->driver->probe_order == order) { if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; @@ -1636,33 +1145,23 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC */ - if (!codec->probed) { + if (!codec->probed && + codec->driver->probe_order == order) { ret = soc_probe_codec(card, codec); if (ret < 0) return ret; } /* probe the platform */ - if (!platform->probed) { - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - if (platform->driver->probe) { - ret = platform->driver->probe(platform); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe platform %s\n", - platform->name); - module_put(platform->dev->driver->owner); - return ret; - } - } - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->card_list, &card->platform_dev_list); + if (!platform->probed && + platform->driver->probe_order == order) { + ret = soc_probe_platform(card, platform); + if (ret < 0) + return ret; } /* probe the CODEC DAI */ - if (!codec_dai->probed) { + if (!codec_dai->probed && codec_dai->driver->probe_order == order) { if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { @@ -1677,8 +1176,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) list_add(&codec_dai->card_list, &card->dai_dev_list); } - /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + /* complete DAI probe during last probe */ + if (order != SND_SOC_COMP_ORDER_LAST) + return 0; ret = soc_post_component_init(card, codec, num, 0); if (ret) @@ -1817,7 +1317,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; - int ret, i; + int ret, i, order; mutex_lock(&card->mutex); @@ -1895,12 +1395,16 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } - for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i); - if (ret < 0) { - pr_err("asoc: failed to instantiate card %s: %d\n", + /* early DAI link probe */ + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_dai_link(card, i, order); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", card->name, ret); - goto probe_dai_err; + goto probe_dai_err; + } } } @@ -2096,67 +1600,6 @@ static struct platform_driver soc_driver = { .remove = soc_remove, }; -/* create a new pcm */ -static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_pcm *pcm; - char new_name[64]; - int ret = 0, playback = 0, capture = 0; - - /* check client and interface hw capabilities */ - snprintf(new_name, sizeof(new_name), "%s %s-%d", - rtd->dai_link->stream_name, codec_dai->name, num); - - if (codec_dai->driver->playback.channels_min) - playback = 1; - if (codec_dai->driver->capture.channels_min) - capture = 1; - - dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); - ret = snd_pcm_new(rtd->card->snd_card, new_name, - num, playback, capture, &pcm); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); - return ret; - } - - rtd->pcm = pcm; - pcm->private_data = rtd; - if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; - } - - if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); - - if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - - if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd->card->snd_card, - codec_dai, pcm); - if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); - return ret; - } - } - - pcm->private_free = platform->driver->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); - return ret; -} - /** * snd_soc_codec_volatile_register: Report if a register is volatile. * @@ -2211,6 +1654,38 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); +int snd_soc_platform_read(struct snd_soc_platform *platform, + unsigned int reg) +{ + unsigned int ret; + + if (!platform->driver->read) { + dev_err(platform->dev, "platform has no read back\n"); + return -1; + } + + ret = platform->driver->read(platform, reg); + dev_dbg(platform->dev, "read %x => %x\n", reg, ret); + trace_snd_soc_preg_read(platform, reg, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_platform_read); + +int snd_soc_platform_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val) +{ + if (!platform->driver->write) { + dev_err(platform->dev, "platform has no write back\n"); + return -1; + } + + dev_dbg(platform->dev, "write %x = %x\n", reg, val); + trace_snd_soc_preg_write(platform, reg, val); + return platform->driver->write(platform, reg, val); +} +EXPORT_SYMBOL_GPL(snd_soc_platform_write); + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -2323,7 +1798,7 @@ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, return ret; old = ret; - new = (old & ~mask) | value; + new = (old & ~mask) | (value & mask); change = old != new; if (change) { ret = snd_soc_write(codec, reg, new); @@ -2490,6 +1965,36 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_add_controls); /** + * snd_soc_add_platform_controls - add an array of controls to a platform. + * Convienience function to add a list of controls. + * + * @platform: platform to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_platform_controls(struct snd_soc_platform *platform, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = platform->card->snd_card; + int err, i; + + for (i = 0; i < num_controls; i++) { + const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, platform, + control->name, NULL)); + if (err < 0) { + dev_err(platform->dev, "Failed to add %s %d\n",control->name, err); + return err; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls); + +/** * snd_soc_info_enum_double - enumerated double mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -3633,6 +3138,8 @@ int snd_soc_register_platform(struct device *dev, platform->dev = dev; platform->driver = platform_drv; + platform->dapm.dev = dev; + platform->dapm.platform = platform; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 32ab7fc4579..54fa2e5e307 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,51 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) +{ + if (w->codec) + return snd_soc_read(w->codec, reg); + else if (w->platform) + return snd_soc_platform_read(w->platform, reg); + + dev_err(w->dapm->dev, "no valid widget read method\n"); + return -1; +} + +static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) +{ + if (w->codec) + return snd_soc_write(w->codec, reg, val); + else if (w->platform) + return snd_soc_platform_write(w->platform, reg, val); + + dev_err(w->dapm->dev, "no valid widget write method\n"); + return -1; +} + +static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, + unsigned short reg, unsigned int mask, unsigned int value) +{ + int change; + unsigned int old, new; + int ret; + + ret = soc_widget_read(w, reg); + if (ret < 0) + return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = soc_widget_write(w, reg, new); + if (ret < 0) + return ret; + } + + return change; +} + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -139,39 +184,26 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, struct snd_soc_card *card = dapm->card; int ret = 0; - switch (level) { - case SND_SOC_BIAS_ON: - dev_dbg(dapm->dev, "Setting full bias\n"); - break; - case SND_SOC_BIAS_PREPARE: - dev_dbg(dapm->dev, "Setting bias prepare\n"); - break; - case SND_SOC_BIAS_STANDBY: - dev_dbg(dapm->dev, "Setting standby bias\n"); - break; - case SND_SOC_BIAS_OFF: - dev_dbg(dapm->dev, "Setting bias off\n"); - break; - default: - dev_err(dapm->dev, "Setting invalid bias %d\n", level); - return -EINVAL; - } - trace_snd_soc_bias_level_start(card, level); if (card && card->set_bias_level) - ret = card->set_bias_level(card, level); - if (ret == 0) { - if (dapm->codec && dapm->codec->driver->set_bias_level) - ret = dapm->codec->driver->set_bias_level(dapm->codec, level); + ret = card->set_bias_level(card, dapm, level); + if (ret != 0) + goto out; + + if (dapm->codec) { + if (dapm->codec->driver->set_bias_level) + ret = dapm->codec->driver->set_bias_level(dapm->codec, + level); else dapm->bias_level = level; } - if (ret == 0) { - if (card && card->set_bias_level_post) - ret = card->set_bias_level_post(card, level); - } + if (ret != 0) + goto out; + if (card && card->set_bias_level_post) + ret = card->set_bias_level_post(card, dapm, level); +out: trace_snd_soc_bias_level_done(card, level); return ret; @@ -194,7 +226,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = snd_soc_read(w->codec, reg); + val = soc_widget_read(w, reg); val = (val >> shift) & mask; if ((invert && !val) || (!invert && val)) @@ -209,8 +241,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val, item, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; - val = snd_soc_read(w->codec, e->reg); + ; + val = soc_widget_read(w, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; @@ -240,7 +272,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, w->kcontrol_news[i].private_value; int val, item; - val = snd_soc_read(w->codec, e->reg); + val = soc_widget_read(w, e->reg); val = (val >> e->shift_l) & e->mask; for (item = 0; item < e->max; item++) { if (val == e->values[item]) @@ -606,6 +638,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + if (path->weak) + continue; + if (path->walked) continue; @@ -656,6 +691,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + if (path->weak) + continue; + if (path->walked) continue; @@ -681,7 +719,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - snd_soc_update_bits(w->codec, -(w->reg + 1), + soc_widget_update_bits(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -737,6 +775,9 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + if (path->weak) + continue; + if (path->connected && !path->connected(path->source, path->sink)) continue; @@ -885,11 +926,17 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, } if (reg >= 0) { + /* Any widget will do, they should all be updating the + * same register. + */ + w = list_first_entry(pending, struct snd_soc_dapm_widget, + power_list); + pop_dbg(dapm->dev, card->pop_time, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - snd_soc_update_bits(dapm->codec, reg, mask, value); + soc_widget_update_bits(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1041,16 +1088,17 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) struct snd_soc_dapm_context *d = data; int ret; - if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) { + /* If we're off and we're not supposed to be go into STANDBY */ + if (d->bias_level == SND_SOC_BIAS_OFF && + d->target_bias_level != SND_SOC_BIAS_OFF) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, "Failed to turn on bias: %d\n", ret); } - /* If we're changing to all on or all off then prepare */ - if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) || - (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) { + /* Prepare for a STADDBY->ON or ON->STANDBY transition */ + if (d->bias_level != d->target_bias_level) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, @@ -1067,7 +1115,9 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) int ret; /* If we just powered the last thing off drop to standby bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) { + if (d->bias_level == SND_SOC_BIAS_PREPARE && + (d->target_bias_level == SND_SOC_BIAS_STANDBY || + d->target_bias_level == SND_SOC_BIAS_OFF)) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, "Failed to apply standby bias: %d\n", @@ -1075,14 +1125,16 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } /* If we're in standby and can support bias off then do that */ - if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) { + if (d->bias_level == SND_SOC_BIAS_STANDBY && + d->target_bias_level == SND_SOC_BIAS_OFF) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); if (ret != 0) dev_err(d->dev, "Failed to turn off bias: %d\n", ret); } /* If we just powered up then move to active bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) { + if (d->bias_level == SND_SOC_BIAS_PREPARE && + d->target_bias_level == SND_SOC_BIAS_ON) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON); if (ret != 0) dev_err(d->dev, "Failed to apply active bias: %d\n", @@ -1107,13 +1159,19 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(up_list); LIST_HEAD(down_list); LIST_HEAD(async_domain); + enum snd_soc_bias_level bias; int power; trace_snd_soc_dapm_start(card); - list_for_each_entry(d, &card->dapm_list, list) - if (d->n_widgets || d->codec == NULL) - d->dev_power = 0; + list_for_each_entry(d, &card->dapm_list, list) { + if (d->n_widgets || d->codec == NULL) { + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; + } + } /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. @@ -1135,8 +1193,27 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) power = w->power_check(w); else power = 1; - if (power) - w->dapm->dev_power = 1; + + if (power) { + d = w->dapm; + + /* Supplies and micbiases only bring + * the context up to STANDBY as unless + * something else is active and + * passing audio they generally don't + * require full power. + */ + switch (w->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_micbias: + if (d->target_bias_level < SND_SOC_BIAS_STANDBY) + d->target_bias_level = SND_SOC_BIAS_STANDBY; + break; + default: + d->target_bias_level = SND_SOC_BIAS_ON; + break; + } + } if (w->power == power) continue; @@ -1160,24 +1237,19 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) switch (event) { case SND_SOC_DAPM_STREAM_START: case SND_SOC_DAPM_STREAM_RESUME: - dapm->dev_power = 1; + dapm->target_bias_level = SND_SOC_BIAS_ON; break; case SND_SOC_DAPM_STREAM_STOP: - dapm->dev_power = !!dapm->codec->active; + if (dapm->codec->active) + dapm->target_bias_level = SND_SOC_BIAS_ON; + else + dapm->target_bias_level = SND_SOC_BIAS_STANDBY; break; case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->dev_power = 0; + dapm->target_bias_level = SND_SOC_BIAS_STANDBY; break; case SND_SOC_DAPM_STREAM_NOP: - switch (dapm->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - dapm->dev_power = 0; - break; - default: - dapm->dev_power = 1; - break; - } + dapm->target_bias_level = dapm->bias_level; break; default: break; @@ -1185,12 +1257,12 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } /* Force all contexts in the card to the same bias state */ - power = 0; + bias = SND_SOC_BIAS_OFF; list_for_each_entry(d, &card->dapm_list, list) - if (d->dev_power) - power = 1; + if (d->target_bias_level > bias) + bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - d->dev_power = power; + d->target_bias_level = bias; /* Run all the bias changes in parallel */ @@ -1794,6 +1866,84 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); +static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route) +{ + struct snd_soc_dapm_widget *source = dapm_find_widget(dapm, + route->source, + true); + struct snd_soc_dapm_widget *sink = dapm_find_widget(dapm, + route->sink, + true); + struct snd_soc_dapm_path *path; + int count = 0; + + if (!source) { + dev_err(dapm->dev, "Unable to find source %s for weak route\n", + route->source); + return -ENODEV; + } + + if (!sink) { + dev_err(dapm->dev, "Unable to find sink %s for weak route\n", + route->sink); + return -ENODEV; + } + + if (route->control || route->connected) + dev_warn(dapm->dev, "Ignoring control for weak route %s->%s\n", + route->source, route->sink); + + list_for_each_entry(path, &source->sinks, list_source) { + if (path->sink == sink) { + path->weak = 1; + count++; + } + } + + if (count == 0) + dev_err(dapm->dev, "No path found for weak route %s->%s\n", + route->source, route->sink); + if (count > 1) + dev_warn(dapm->dev, "%d paths found for weak route %s->%s\n", + count, route->source, route->sink); + + return 0; +} + +/** + * snd_soc_dapm_weak_routes - Mark routes between DAPM widgets as weak + * @dapm: DAPM context + * @route: audio routes + * @num: number of routes + * + * Mark existing routes matching those specified in the passed array + * as being weak, meaning that they are ignored for the purpose of + * power decisions. The main intended use case is for sidetone paths + * which couple audio between other independent paths if they are both + * active in order to make the combination work better at the user + * level but which aren't intended to be "used". + * + * Note that CODEC drivers should not use this as sidetone type paths + * can frequently also be used as bypass paths. + */ +int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num) +{ + int i, err; + int ret = 0; + + for (i = 0; i < num; i++) { + err = snd_soc_dapm_weak_route(dapm, route); + if (err) + ret = err; + route++; + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); + /** * snd_soc_dapm_new_widgets - add new dapm widgets * @dapm: DAPM context @@ -1865,7 +2015,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) /* Read the initial power state from the device */ if (w->reg >= 0) { - val = snd_soc_read(w->codec, w->reg); + val = soc_widget_read(w, w->reg); val &= 1 << w->shift; if (w->invert) val = !val; @@ -2353,6 +2503,7 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, dapm->n_widgets++; w->dapm = dapm; w->codec = dapm->codec; + w->platform = dapm->platform; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c new file mode 100644 index 00000000000..cca490c8058 --- /dev/null +++ b/sound/soc/soc-io.c @@ -0,0 +1,396 @@ +/* + * soc-io.c -- ASoC register I/O helpers + * + * Copyright 2009-2011 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie@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. + */ + +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +#include <trace/events/asoc.h> + +#ifdef CONFIG_SPI_MASTER +static int do_spi_write(void *control, const char *data, int len) +{ + struct spi_device *spi = control; + int ret; + + ret = spi_write(spi, data, len); + if (ret < 0) + return ret; + + return len; +} +#endif + +static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value, const void *data, int len) +{ + int ret; + + if (!snd_soc_codec_volatile_register(codec, reg) && + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } + + if (codec->cache_only) { + codec->cache_sync = 1; + return 0; + } + + ret = codec->hw_write(codec->control_data, data, len); + if (ret == len) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) +{ + int ret; + unsigned int val; + + if (reg >= codec->driver->reg_cache_size || + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { + if (codec->cache_only) + return -1; + + BUG_ON(!codec->hw_read); + return codec->hw_read(codec, reg); + } + + ret = snd_soc_cache_read(codec, reg, &val); + if (ret < 0) + return -1; + return val; +} + +static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 data; + + data = cpu_to_be16((reg << 12) | (value & 0xffffff)); + + return do_hw_write(codec, reg, value, &data, 2); +} + +static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 data; + + data = cpu_to_be16((reg << 9) | (value & 0x1ff)); + + return do_hw_write(codec, reg, value, &data, 2); +} + +static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + reg &= 0xff; + data[0] = reg; + data[1] = value & 0xff; + + return do_hw_write(codec, reg, value, data, 2); +} + +static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + u16 val = cpu_to_be16(value); + + data[0] = reg; + memcpy(&data[1], &val, sizeof(val)); + + return do_hw_write(codec, reg, value, data, 3); +} + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int do_i2c_read(struct snd_soc_codec *codec, + void *reg, int reglen, + void *data, int datalen) +{ + struct i2c_msg xfer[2]; + int ret; + struct i2c_client *client = codec->control_data; + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = reglen; + xfer[0].buf = reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = datalen; + xfer[1].buf = data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret == 2) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u8 reg = r; + u8 data; + int ret; + + ret = do_i2c_read(codec, ®, 1, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_8_8_read_i2c NULL +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u8 reg = r; + u16 data; + int ret; + + ret = do_i2c_read(codec, ®, 1, &data, 2); + if (ret < 0) + return 0; + return (data >> 8) | ((data & 0xff) << 8); +} +#else +#define snd_soc_8_16_read_i2c NULL +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u16 reg = r; + u8 data; + int ret; + + ret = do_i2c_read(codec, ®, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_i2c NULL +#endif + +static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[3]; + u16 rval = cpu_to_be16(reg); + + memcpy(data, &rval, sizeof(rval)); + data[2] = value; + + return do_hw_write(codec, reg, value, data, 3); +} + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u16 reg = cpu_to_be16(r); + u16 data; + int ret; + + ret = do_i2c_read(codec, ®, 2, &data, 2); + if (ret < 0) + return 0; + return be16_to_cpu(data); +} +#else +#define snd_soc_16_16_read_i2c NULL +#endif + +static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 data[2]; + + data[0] = cpu_to_be16(reg); + data[1] = cpu_to_be16(value); + + return do_hw_write(codec, reg, value, data, sizeof(data)); +} + +/* Primitive bulk write support for soc-cache. The data pointed to by + * `data' needs to already be in the form the hardware expects + * including any leading register specific data. Any data written + * through this function will not go through the cache as it only + * handles writing to volatile or out of bounds registers. + */ +static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, + const void *data, size_t len) +{ + int ret; + + /* To ensure that we don't get out of sync with the cache, check + * whether the base register is volatile or if we've directly asked + * to bypass the cache. Out of bounds registers are considered + * volatile. + */ + if (!codec->cache_bypass + && !snd_soc_codec_volatile_register(codec, reg) + && reg < codec->driver->reg_cache_size) + return -EINVAL; + + switch (codec->control_type) { +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) + case SND_SOC_I2C: + ret = i2c_master_send(to_i2c_client(codec->dev), data, len); + break; +#endif +#if defined(CONFIG_SPI_MASTER) + case SND_SOC_SPI: + ret = spi_write(to_spi_device(codec->dev), data, len); + break; +#endif + default: + BUG(); + } + + if (ret == len) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static struct { + int addr_bits; + int data_bits; + int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); +} io_types[] = { + { + .addr_bits = 4, .data_bits = 12, + .write = snd_soc_4_12_write, + }, + { + .addr_bits = 7, .data_bits = 9, + .write = snd_soc_7_9_write, + }, + { + .addr_bits = 8, .data_bits = 8, + .write = snd_soc_8_8_write, + .i2c_read = snd_soc_8_8_read_i2c, + }, + { + .addr_bits = 8, .data_bits = 16, + .write = snd_soc_8_16_write, + .i2c_read = snd_soc_8_16_read_i2c, + }, + { + .addr_bits = 16, .data_bits = 8, + .write = snd_soc_16_8_write, + .i2c_read = snd_soc_16_8_read_i2c, + }, + { + .addr_bits = 16, .data_bits = 16, + .write = snd_soc_16_16_write, + .i2c_read = snd_soc_16_16_read_i2c, + }, +}; + +/** + * snd_soc_codec_set_cache_io: Set up standard I/O functions. + * + * @codec: CODEC to configure. + * @addr_bits: Number of bits of register address data. + * @data_bits: Number of bits of data per register. + * @control: Control bus used. + * + * Register formats are frequently shared between many I2C and SPI + * devices. In order to promote code reuse the ASoC core provides + * some standard implementations of CODEC read and write operations + * which can be set up using this function. + * + * The caller is responsible for allocating and initialising the + * actual cache. + * + * Note that at present this code cannot be used by CODECs with + * volatile registers. + */ +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(io_types); i++) + if (io_types[i].addr_bits == addr_bits && + io_types[i].data_bits == data_bits) + break; + if (i == ARRAY_SIZE(io_types)) { + printk(KERN_ERR + "No I/O functions for %d bit address %d bit data\n", + addr_bits, data_bits); + return -EINVAL; + } + + codec->write = io_types[i].write; + codec->read = hw_read; + codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; + + switch (control) { + case SND_SOC_I2C: +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) + codec->hw_write = (hw_write_t)i2c_master_send; +#endif + if (io_types[i].i2c_read) + codec->hw_read = io_types[i].i2c_read; + + codec->control_data = container_of(codec->dev, + struct i2c_client, + dev); + break; + + case SND_SOC_SPI: +#ifdef CONFIG_SPI_MASTER + codec->hw_write = do_spi_write; +#endif + + codec->control_data = container_of(codec->dev, + struct spi_device, + dev); + break; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); + diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c new file mode 100644 index 00000000000..b5759397afa --- /dev/null +++ b/sound/soc/soc-pcm.c @@ -0,0 +1,639 @@ +/* + * soc-pcm.c -- ALSA SoC PCM + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Authors: Liam Girdwood <lrg@ti.com> + * Mark Brown <broonie@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. + * + */ + +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +static DEFINE_MUTEX(pcm_mutex); + +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + /* This can happen if multiple streams are starting simultaneously - + * the second can need to get its constraints before the first has + * picked a rate. Complain and allow the application to carry on. + */ + if (!rtd->rate) { + dev_warn(&rtd->dev, + "Not enforcing symmetric_rates due to race\n"); + return 0; + } + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + + return 0; +} + +/* + * Called by ALSA when a PCM substream is opened, the runtime->hw record is + * then initialized and any private data can be allocated. This also calls + * startup for the cpu DAI, platform, machine and codec DAI. + */ +static int soc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + /* startup the audio subsystem */ + if (cpu_dai->driver->ops->startup) { + ret = cpu_dai->driver->ops->startup(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open interface %s\n", + cpu_dai->name); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->open) { + ret = platform->driver->ops->open(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + goto platform_err; + } + } + + if (codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't open codec %s\n", + codec_dai->name); + goto codec_dai_err; + } + } + + if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + ret = rtd->dai_link->ops->startup(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + goto machine_err; + } + } + + /* Check that the codec and cpu DAIs are compatible */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw.rate_min = + max(codec_dai_drv->playback.rate_min, + cpu_dai_drv->playback.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->playback.rate_max, + cpu_dai_drv->playback.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->playback.channels_min, + cpu_dai_drv->playback.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->playback.channels_max, + cpu_dai_drv->playback.channels_max); + runtime->hw.formats = + codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; + runtime->hw.rates = + codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; + if (codec_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->playback.rates; + if (cpu_dai_drv->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->playback.rates; + } else { + runtime->hw.rate_min = + max(codec_dai_drv->capture.rate_min, + cpu_dai_drv->capture.rate_min); + runtime->hw.rate_max = + min(codec_dai_drv->capture.rate_max, + cpu_dai_drv->capture.rate_max); + runtime->hw.channels_min = + max(codec_dai_drv->capture.channels_min, + cpu_dai_drv->capture.channels_min); + runtime->hw.channels_max = + min(codec_dai_drv->capture.channels_max, + cpu_dai_drv->capture.channels_max); + runtime->hw.formats = + codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; + runtime->hw.rates = + codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; + if (codec_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai_drv->capture.rates; + if (cpu_dai_drv->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai_drv->capture.rates; + } + + ret = -EINVAL; + snd_pcm_limit_hw_rates(runtime); + if (!runtime->hw.rates) { + printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.formats) { + printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { + printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", + codec_dai->name, cpu_dai->name); + goto config_err; + } + + /* Symmetry only applies if we've already got an active stream. */ + if (cpu_dai->active || codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream); + if (ret != 0) + goto config_err; + } + + pr_debug("asoc: %s <-> %s info:\n", + codec_dai->name, cpu_dai->name); + pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); + pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, + runtime->hw.channels_max); + pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, + runtime->hw.rate_max); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + cpu_dai->active++; + codec_dai->active++; + rtd->codec->active++; + mutex_unlock(&rtd->pcm_mutex); + return 0; + +config_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + +machine_err: + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + +codec_dai_err: + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + +platform_err: + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); +out: + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + +/* + * Power down the audio subsystem pmdown_time msecs after close is called. + * This is to ensure there are no pops or clicks in between any music tracks + * due to DAPM power cycling. + */ +static void close_delayed_work(struct work_struct *work) +{ + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (codec_dai->pop_wait == 1) { + codec_dai->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&rtd->pcm_mutex); +} + +/* + * Called by ALSA when a PCM substream is closed. Private data can be + * freed here. The cpu DAI, codec DAI, machine and platform are also + * shutdown. + */ +static int soc_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + cpu_dai->active--; + codec_dai->active--; + codec->active--; + + /* Muting the DAC suppresses artifacts caused during digital + * shutdown, for example from stopping clocks. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_digital_mute(codec_dai, 1); + + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); + + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); + + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); + + if (platform->driver->ops && platform->driver->ops->close) + platform->driver->ops->close(substream); + cpu_dai->runtime = NULL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* start delayed pop wq here for playback streams */ + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } else { + /* capture streams can be powered down now */ + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_STOP); + } + + mutex_unlock(&rtd->pcm_mutex); + return 0; +} + +/* + * Called by ALSA when the PCM substream is prepared, can set format, sample + * rate, etc. This function is non atomic and can be called multiple times, + * it can refer to the runtime info. + */ +static int soc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + ret = rtd->dai_link->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: machine prepare error\n"); + goto out; + } + } + + if (platform->driver->ops && platform->driver->ops->prepare) { + ret = platform->driver->ops->prepare(substream); + if (ret < 0) { + printk(KERN_ERR "asoc: platform prepare error\n"); + goto out; + } + } + + if (codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: codec DAI prepare error\n"); + goto out; + } + } + + if (cpu_dai->driver->ops->prepare) { + ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + goto out; + } + } + + /* cancel any delayed stream shutdown that is pending */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&rtd->delayed_work); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, + SND_SOC_DAPM_STREAM_START); + + snd_soc_dai_digital_mute(codec_dai, 0); + +out: + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + +/* + * Called by ALSA when the hardware params are set by application. This + * function can also be called multiple times and can allocate buffers + * (using snd_pcm_lib_* ). It's non-atomic. + */ +static int soc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + ret = rtd->dai_link->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: machine hw_params failed\n"); + goto out; + } + } + + if (codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: can't set codec %s hw params\n", + codec_dai->name); + goto codec_err; + } + } + + if (cpu_dai->driver->ops->hw_params) { + ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); + if (ret < 0) { + printk(KERN_ERR "asoc: interface %s hw params failed\n", + cpu_dai->name); + goto interface_err; + } + } + + if (platform->driver->ops && platform->driver->ops->hw_params) { + ret = platform->driver->ops->hw_params(substream, params); + if (ret < 0) { + printk(KERN_ERR "asoc: platform %s hw params failed\n", + platform->name); + goto platform_err; + } + } + + rtd->rate = params_rate(params); + +out: + mutex_unlock(&rtd->pcm_mutex); + return ret; + +platform_err: + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + +interface_err: + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + +codec_err: + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + +/* + * Frees resources allocated by hw_params, can be called multiple times + */ +static int soc_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + /* apply codec digital mute */ + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); + + /* free any machine hw params */ + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); + + /* free any DMA resources */ + if (platform->driver->ops && platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); + + /* now free hw params for the DAIs */ + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); + + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); + + mutex_unlock(&rtd->pcm_mutex); + return 0; +} + +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->ops && platform->driver->ops->trigger) { + ret = platform->driver->ops->trigger(substream, cmd); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->trigger) { + ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} + +/* + * soc level wrapper for pointer callback + * If cpu_dai, codec_dai, platform driver has the delay callback, than + * the runtime->delay will be updated accordingly. + */ +static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t offset = 0; + snd_pcm_sframes_t delay = 0; + + if (platform->driver->ops && platform->driver->ops->pointer) + offset = platform->driver->ops->pointer(substream); + + if (cpu_dai->driver->ops->delay) + delay += cpu_dai->driver->ops->delay(substream, cpu_dai); + + if (codec_dai->driver->ops->delay) + delay += codec_dai->driver->ops->delay(substream, codec_dai); + + if (platform->driver->delay) + delay += platform->driver->delay(substream, codec_dai); + + runtime->delay = delay; + + return offset; +} + +/* ASoC PCM operations */ +static struct snd_pcm_ops soc_pcm_ops = { + .open = soc_pcm_open, + .close = soc_pcm_close, + .hw_params = soc_pcm_hw_params, + .hw_free = soc_pcm_hw_free, + .prepare = soc_pcm_prepare, + .trigger = soc_pcm_trigger, + .pointer = soc_pcm_pointer, +}; + +/* create a new pcm */ +int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm *pcm; + char new_name[64]; + int ret = 0, playback = 0, capture = 0; + + /* check client and interface hw capabilities */ + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + + if (codec_dai->driver->playback.channels_min) + playback = 1; + if (codec_dai->driver->capture.channels_min) + capture = 1; + + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + ret = snd_pcm_new(rtd->card->snd_card, new_name, + num, playback, capture, &pcm); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + return ret; + } + + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + + rtd->pcm = pcm; + pcm->private_data = rtd; + if (platform->driver->ops) { + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; + } + + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd); + if (ret < 0) { + pr_err("asoc: platform pcm constructor failed\n"); + return ret; + } + } + + pcm->private_free = platform->driver->pcm_free; + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + cpu_dai->name); + return ret; +} diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 035d39a4beb..c6af1fd707f 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -12,6 +12,15 @@ config SND_SOC_TEGRA_I2S Tegra I2S interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA_SPDIF + tristate + depends on SND_SOC_TEGRA + default m + help + Say Y or M if you want to add support for the SPDIF interface. + You will also need to select the individual machine drivers to support + below. + config MACH_HAS_SND_SOC_TEGRA_WM8903 bool help diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index fa6574d92a3..4d943b3fe15 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -2,12 +2,14 @@ snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-i2s-objs := tegra_i2s.o +snd-soc-tegra-spdif-objs := tegra_spdif.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 95f03c10b4f..f36b9969cfe 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -354,7 +354,6 @@ struct snd_soc_dai_driver tegra_i2s_dai[] = { static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) { struct tegra_i2s * i2s; - char clk_name[12]; /* tegra-i2s.0 */ struct resource *mem, *memregion, *dmareq; int ret; @@ -389,8 +388,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, i2s); - snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id); - i2s->clk_i2s = clk_get_sys(clk_name, NULL); + i2s->clk_i2s = clk_get(&pdev->dev, NULL); if (IS_ERR(i2s->clk_i2s)) { dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); ret = PTR_ERR(i2s->clk_i2s); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 3c271f95358..ff86e5e3db6 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -322,9 +322,11 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) static u64 tegra_dma_mask = DMA_BIT_MASK(32); -static int tegra_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c new file mode 100644 index 00000000000..abe606b0a29 --- /dev/null +++ b/sound/soc/tegra/tegra_spdif.c @@ -0,0 +1,371 @@ +/* + * tegra_spdif.c - Tegra SPDIF driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/debugfs.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/seq_file.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <mach/iomap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra_spdif.h" + +#define DRV_NAME "tegra-spdif" + +static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, + u32 val) +{ + __raw_writel(val, spdif->regs + reg); +} + +static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) +{ + return __raw_readl(spdif->regs + reg); +} + +#ifdef CONFIG_DEBUG_FS +static int tegra_spdif_show(struct seq_file *s, void *unused) +{ +#define REG(r) { r, #r } + static const struct { + int offset; + const char *name; + } regs[] = { + REG(TEGRA_SPDIF_CTRL), + REG(TEGRA_SPDIF_STATUS), + REG(TEGRA_SPDIF_STROBE_CTRL), + REG(TEGRA_SPDIF_DATA_FIFO_CSR), + REG(TEGRA_SPDIF_CH_STA_RX_A), + REG(TEGRA_SPDIF_CH_STA_RX_B), + REG(TEGRA_SPDIF_CH_STA_RX_C), + REG(TEGRA_SPDIF_CH_STA_RX_D), + REG(TEGRA_SPDIF_CH_STA_RX_E), + REG(TEGRA_SPDIF_CH_STA_RX_F), + REG(TEGRA_SPDIF_CH_STA_TX_A), + REG(TEGRA_SPDIF_CH_STA_TX_B), + REG(TEGRA_SPDIF_CH_STA_TX_C), + REG(TEGRA_SPDIF_CH_STA_TX_D), + REG(TEGRA_SPDIF_CH_STA_TX_E), + REG(TEGRA_SPDIF_CH_STA_TX_F), + }; +#undef REG + + struct tegra_spdif *spdif = s->private; + int i; + + for (i = 0; i < ARRAY_SIZE(regs); i++) { + u32 val = tegra_spdif_read(spdif, regs[i].offset); + seq_printf(s, "%s = %08x\n", regs[i].name, val); + } + + return 0; +} + +static int tegra_spdif_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_spdif_show, inode->i_private); +} + +static const struct file_operations tegra_spdif_debug_fops = { + .open = tegra_spdif_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ + spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, + snd_soc_debugfs_root, spdif, + &tegra_spdif_debug_fops); +} + +static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ + if (spdif->debug) + debugfs_remove(spdif->debug); +} +#else +static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ +} + +static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ +} +#endif + +static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + int ret, srate, spdifclock; + + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + switch (params_rate(params)) { + case 32000: + spdifclock = 4096000; + break; + case 44100: + spdifclock = 5644800; + break; + case 48000: + spdifclock = 6144000; + break; + case 88200: + spdifclock = 11289600; + break; + case 96000: + spdifclock = 12288000; + break; + case 176400: + spdifclock = 22579200; + break; + case 192000: + spdifclock = 24576000; + break; + default: + return -EINVAL; + } + + ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); + if (ret) { + dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); + return ret; + } + + return 0; +} + +static void tegra_spdif_start_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (!spdif->clk_refs) + clk_enable(spdif->clk_spdif_out); + spdif->clk_refs++; + tegra_spdif_start_playback(spdif); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + tegra_spdif_stop_playback(spdif); + spdif->clk_refs--; + if (!spdif->clk_refs) + clk_disable(spdif->clk_spdif_out); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_spdif_probe(struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = NULL; + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops tegra_spdif_dai_ops = { + .hw_params = tegra_spdif_hw_params, + .trigger = tegra_spdif_trigger, +}; + +struct snd_soc_dai_driver tegra_spdif_dai = { + .name = DRV_NAME, + .probe = tegra_spdif_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra_spdif_dai_ops, +}; + +static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) +{ + struct tegra_spdif *spdif; + struct resource *mem, *memregion, *dmareq; + int ret; + + spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); + if (!spdif) { + dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); + ret = -ENOMEM; + goto exit; + } + dev_set_drvdata(&pdev->dev, spdif); + + spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + if (IS_ERR(spdif->clk_spdif_out)) { + pr_err("Can't retrieve spdif clock\n"); + ret = PTR_ERR(spdif->clk_spdif_out); + goto err_free; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = request_mem_region(mem->start, resource_size(mem), + DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + spdif->regs = ioremap(mem->start, resource_size(mem)); + if (!spdif->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release; + } + + spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; + spdif->playback_dma_data.wrap = 4; + spdif->playback_dma_data.width = 32; + spdif->playback_dma_data.req_sel = dmareq->start; + + ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_unmap; + } + + tegra_spdif_debug_add(spdif); + + return 0; + +err_unmap: + iounmap(spdif->regs); +err_release: + release_mem_region(mem->start, resource_size(mem)); +err_clk_put: + clk_put(spdif->clk_spdif_out); +err_free: + kfree(spdif); +exit: + return ret; +} + +static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) +{ + struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); + struct resource *res; + + snd_soc_unregister_dai(&pdev->dev); + + tegra_spdif_debug_remove(spdif); + + iounmap(spdif->regs); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, resource_size(res)); + + clk_put(spdif->clk_spdif_out); + + kfree(spdif); + + return 0; +} + +static struct platform_driver tegra_spdif_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_spdif_platform_probe, + .remove = __devexit_p(tegra_spdif_platform_remove), +}; + +static int __init snd_tegra_spdif_init(void) +{ + return platform_driver_register(&tegra_spdif_driver); +} +module_init(snd_tegra_spdif_init); + +static void __exit snd_tegra_spdif_exit(void) +{ + platform_driver_unregister(&tegra_spdif_driver); +} +module_exit(snd_tegra_spdif_exit); + +MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); +MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h new file mode 100644 index 00000000000..2e03db43027 --- /dev/null +++ b/sound/soc/tegra/tegra_spdif.h @@ -0,0 +1,473 @@ +/* + * tegra_spdif.h - Definitions for Tegra SPDIF driver + * + * Author: Stephen Warren <swarren@nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * Based on code copyright/by: + * Copyright (c) 2008-2009, NVIDIA Corporation + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_SPDIF_H__ +#define __TEGRA_SPDIF_H__ + +#include "tegra_pcm.h" + +/* Offsets from TEGRA_SPDIF_BASE */ + +#define TEGRA_SPDIF_CTRL 0x0 +#define TEGRA_SPDIF_STATUS 0x4 +#define TEGRA_SPDIF_STROBE_CTRL 0x8 +#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C +#define TEGRA_SPDIF_DATA_OUT 0x40 +#define TEGRA_SPDIF_DATA_IN 0x80 +#define TEGRA_SPDIF_CH_STA_RX_A 0x100 +#define TEGRA_SPDIF_CH_STA_RX_B 0x104 +#define TEGRA_SPDIF_CH_STA_RX_C 0x108 +#define TEGRA_SPDIF_CH_STA_RX_D 0x10C +#define TEGRA_SPDIF_CH_STA_RX_E 0x110 +#define TEGRA_SPDIF_CH_STA_RX_F 0x114 +#define TEGRA_SPDIF_CH_STA_TX_A 0x140 +#define TEGRA_SPDIF_CH_STA_TX_B 0x144 +#define TEGRA_SPDIF_CH_STA_TX_C 0x148 +#define TEGRA_SPDIF_CH_STA_TX_D 0x14C +#define TEGRA_SPDIF_CH_STA_TX_E 0x150 +#define TEGRA_SPDIF_CH_STA_TX_F 0x154 +#define TEGRA_SPDIF_USR_STA_RX_A 0x180 +#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 + +/* Fields in TEGRA_SPDIF_CTRL */ + +/* Start capturing from 0=right, 1=left channel */ +#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) + +/* SPDIF receiver(RX) enable */ +#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) + +/* SPDIF Transmitter(TX) enable */ +#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) + +/* Transmit Channel status */ +#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) + +/* Transmit user Data */ +#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) + +/* Interrupt on transmit error */ +#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) + +/* Interrupt on receive error */ +#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) + +/* Interrupt on invalid preamble */ +#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) + +/* Interrupt on "B" preamble */ +#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) + +/* Interrupt when block of channel status received */ +#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) + +/* Interrupt when a valid information unit (IU) is received */ +#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) + +/* Interrupt when RX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) + +/* Interrupt when TX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) + +/* Interrupt when RX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) + +/* Interrupt when TX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) + +/* Loopback test mode enable */ +#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) + +/* + * Pack data mode: + * 0 = Single data (16 bit needs to be padded to match the + * interface data bit size). + * 1 = Packeted left/right channel data into a single word. + */ +#define TEGRA_SPDIF_CTRL_PACK (1 << 14) + +/* + * 00 = 16bit data + * 01 = 20bit data + * 10 = 24bit data + * 11 = raw data + */ +#define TEGRA_SPDIF_BIT_MODE_16BIT 0 +#define TEGRA_SPDIF_BIT_MODE_20BIT 1 +#define TEGRA_SPDIF_BIT_MODE_24BIT 2 +#define TEGRA_SPDIF_BIT_MODE_RAW 3 + +#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 +#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) + +/* Fields in TEGRA_SPDIF_STATUS */ + +/* + * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must + * write a 1 to the corresponding bit location to clear the status. + */ + +/* + * Receiver(RX) shifter is busy receiving data. + * This bit is asserted when the receiver first locked onto the + * preamble of the data stream after RX_EN is asserted. This bit is + * deasserted when either, + * (a) the end of a frame is reached after RX_EN is deeasserted, or + * (b) the SPDIF data stream becomes inactive. + */ +#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) + +/* + * Transmitter(TX) shifter is busy transmitting data. + * This bit is asserted when TX_EN is asserted. + * This bit is deasserted when the end of a frame is reached after + * TX_EN is deasserted. + */ +#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) + +/* + * TX is busy shifting out channel status. + * This bit is asserted when both TX_EN and TC_EN are asserted and + * data from CH_STA_TX_A register is loaded into the internal shifter. + * This bit is deasserted when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) CH_STA_TX_F register is loaded into the internal shifter. + */ +#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) + +/* + * TX User data FIFO busy. + * This bit is asserted when TX_EN and TXU_EN are asserted and + * there's data in the TX user FIFO. This bit is deassert when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) there's no data left in the TX user FIFO. + */ +#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) + +/* TX FIFO Underrun error status */ +#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) + +/* RX FIFO Overrun error status */ +#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) + +/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ +#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) + +/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ +#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) + +/* + * RX channel block data receive status: + * 0=entire block not recieved yet. + * 1=received entire block of channel status, + */ +#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) + +/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ +#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) + +/* + * RX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) + +/* + * TX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) + +/* + * RX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) + +/* + * TX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) + +/* Fields in TEGRA_SPDIF_STROBE_CTRL */ + +/* + * Indicates the approximate number of detected SPDIFIN clocks within a + * bi-phase period. + */ +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) + +/* Data strobe mode: 0=Auto-locked 1=Manual locked */ +#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) + +/* + * Manual data strobe time within the bi-phase clock period (in terms of + * the number of over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) + +/* + * Manual SPDIFIN bi-phase clock period (in terms of the number of + * over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) + +/* Fields in SPDIF_DATA_FIFO_CSR */ + +/* Clear Receiver User FIFO (RX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) + +/* Number of RX USR.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) + +/* Clear Transmitter User FIFO (TX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) + +/* Number of TX USR.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) + +/* Clear Receiver Data FIFO (RX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) + +/* Number of RX DATA.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) + +/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) + +/* Number of TX DATA.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_OUT */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + */ + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_IN */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + * + * Bits 31:24 are common to all modes except 16-bit packed + */ + +#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) +#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) +#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) +#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) + +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ + +/* + * The 6-word receive channel data page buffer holds a block (192 frames) of + * channel status information. The order of receive is from LSB to MSB + * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. + */ + +/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ + +/* + * The 6-word transmit channel data page buffer holds a block (192 frames) of + * channel status information. The order of transmission is from LSB to MSB + * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. + */ + +/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ + +/* + * This 4-word deep FIFO receives user FIFO field information. The order of + * receive is from LSB to MSB bit. + */ + +/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ + +/* + * This 4-word deep FIFO transmits user FIFO field information. The order of + * transmission is from LSB to MSB bit. + */ + +struct tegra_spdif { + struct clk *clk_spdif_out; + int clk_refs; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + void __iomem *regs; + struct dentry *debug; + u32 reg_ctrl; +}; + +#endif diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0d6738a8b29..a42e9ac30f2 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -267,7 +267,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) } machine->gpio_requested |= GPIO_HP_MUTE; - gpio_direction_output(pdata->gpio_hp_mute, 0); + gpio_direction_output(pdata->gpio_hp_mute, 1); } if (gpio_is_valid(pdata->gpio_int_mic_en)) { diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index f4aa4e03c88..34aa972669e 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -288,9 +288,10 @@ static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(dai->platform->dev); struct txx9aclc_soc_device *dev; struct resource *r; |