diff options
Diffstat (limited to 'sound/soc')
130 files changed, 7916 insertions, 1687 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 2ae9619..1d651b8 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -30,6 +30,9 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_TOPOLOGY + bool + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index e189903..669648b 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,9 @@ 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-compress.o soc-io.o soc-devres.o soc-ops.o + +ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o +endif ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..ba8def5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -290,7 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, int dir, dir_mask; int ret; - pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + pr_debug("atmel_ssc_startup: SSC_SR=0x%x\n", ssc_readl(ssc_p->ssc->regs, SR)); /* Enable PMC peripheral clock for this SSC */ diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index dd94fea..5741c0a 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -344,14 +344,8 @@ static int au1xpsc_pcm_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, dmadata); - return snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); -} - -static int au1xpsc_pcm_drvremove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &au1xpsc_soc_platform); } static struct platform_driver au1xpsc_pcm_driver = { @@ -359,7 +353,6 @@ static struct platform_driver au1xpsc_pcm_driver = { .name = "au1xpsc-pcm", }, .probe = au1xpsc_pcm_drvprobe, - .remove = au1xpsc_pcm_drvremove, }; module_platform_driver(au1xpsc_pcm_driver); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 24cc7f4..fcf5a9a 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -312,14 +312,8 @@ static int alchemy_pcm_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); - return snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); -} - -static int alchemy_pcm_drvremove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &alchemy_pcm_soc_platform); } static struct platform_driver alchemy_pcmdma_driver = { @@ -327,7 +321,6 @@ static struct platform_driver alchemy_pcmdma_driver = { .name = "alchemy-pcm-dma", }, .probe = alchemy_pcm_drvprobe, - .remove = alchemy_pcm_drvremove, }; module_platform_driver(alchemy_pcmdma_driver); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index e742ef6..38e853a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -305,19 +305,9 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) return -ENOMEM; iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) - return -ENODEV; - - ret = -EBUSY; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 03fa1cb..8c435be 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -862,6 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; +MODULE_DEVICE_TABLE(of, bcm2835_i2s_of_match); + static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, .driver = { diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 238913e..02ad260 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -450,13 +450,8 @@ static struct snd_soc_platform_driver bf5xx_ac97_soc_platform = { static int bf5xx_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &bf5xx_ac97_soc_platform); -} - -static int bf5xx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &bf5xx_ac97_soc_platform); } static struct platform_driver bf5xx_pcm_driver = { @@ -465,7 +460,6 @@ static struct platform_driver bf5xx_pcm_driver = { }, .probe = bf5xx_soc_platform_probe, - .remove = bf5xx_soc_platform_remove, }; module_platform_driver(bf5xx_pcm_driver); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index d95477a..6cba211d 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -342,13 +342,8 @@ static struct snd_soc_platform_driver bf5xx_i2s_soc_platform = { static int bfin_i2s_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &bf5xx_i2s_soc_platform); -} - -static int bfin_i2s_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &bf5xx_i2s_soc_platform); } static struct platform_driver bfin_i2s_pcm_driver = { @@ -357,7 +352,6 @@ static struct platform_driver bfin_i2s_pcm_driver = { }, .probe = bfin_i2s_soc_platform_probe, - .remove = bfin_i2s_soc_platform_remove, }; module_platform_driver(bfin_i2s_pcm_driver); diff --git a/sound/soc/blackfin/bfin-eval-adau1x61.c b/sound/soc/blackfin/bfin-eval-adau1x61.c index 4229f76..fddfe00c 100644 --- a/sound/soc/blackfin/bfin-eval-adau1x61.c +++ b/sound/soc/blackfin/bfin-eval-adau1x61.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link bfin_eval_adau1x61_dai = { static struct snd_soc_card bfin_eval_adau1x61 = { .name = "bfin-eval-adau1x61", + .owner = THIS_MODULE, .driver_name = "eval-adau1x61", .dai_link = &bfin_eval_adau1x61_dai, .num_links = 1, diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 38b3dad..4d91a6a 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1028,10 +1028,8 @@ static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (dir == PM860X_CLK_DIR_OUT) pm860x->dir = PM860X_CLK_DIR_OUT; - else { - pm860x->dir = PM860X_CLK_DIR_IN; + else /* Slave mode is not supported */ return -EINVAL; - } return 0; } diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 76125a2..6fd467c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_I2C if I2C select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C + select SND_SOC_CS4349 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -62,6 +63,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_GTM601 + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -404,6 +407,11 @@ config SND_SOC_CS42XX8_I2C select SND_SOC_CS42XX8 select REGMAP_I2C +# Cirrus Logic CS4349 HiFi DAC +config SND_SOC_CS4349 + tristate "Cirrus Logic CS4349 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY @@ -447,6 +455,12 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_GTM601 + tristate 'GTM601 UMTS modem audio codec' + +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3b58c45..f65bd7b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,6 +45,7 @@ snd-soc-cs4271-i2c-objs := cs4271-i2c.o snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o +snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -55,6 +56,8 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-gtm601-objs := gtm601.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -233,6 +236,7 @@ obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o +obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o @@ -243,6 +247,8 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3cc69a6..9ef20db 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -202,19 +202,21 @@ static struct snd_soc_dai_driver ad1980_dai = { .formats = SND_SOC_STD_AC97_FMTS, }, }; +#define AD1980_VENDOR_ID 0x41445300 +#define AD1980_VENDOR_MASK 0xffffff00 + static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); unsigned int retry_cnt = 0; + int ret; do { - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(ac97); - if (snd_soc_read(codec, AC97_RESET) == 0x0090) - return 1; - } + ret = snd_ac97_reset(ac97, true, AD1980_VENDOR_ID, + AD1980_VENDOR_MASK); + if (ret >= 0) + return 0; - soc_ac97_ops->reset(ac97); /* * Set bit 16slot in register 74h, then every slot will has only * 16 bits. This command is sent out in 20bit mode, in which @@ -223,8 +225,6 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) */ snd_soc_write(codec, AC97_AD_SERIAL_CFG, 0x9900); - if (snd_soc_read(codec, AC97_RESET) == 0x0090) - return 0; } while (retry_cnt++ < 10); dev_err(codec->dev, "Failed to reset: AC97 link error\n"); @@ -240,7 +240,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) u16 vendor_id2; u16 ext_status; - ac97 = snd_soc_new_ac97_codec(codec); + ac97 = snd_soc_new_ac97_codec(codec, 0, 0); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); @@ -260,22 +260,10 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) if (ret < 0) goto reset_err; - /* Read out vendor ID to make sure it is ad1980 */ - if (snd_soc_read(codec, AC97_VENDOR_ID1) != 0x4144) { - ret = -ENODEV; - goto reset_err; - } - vendor_id2 = snd_soc_read(codec, AC97_VENDOR_ID2); - - if (vendor_id2 != 0x5370) { - if (vendor_id2 != 0x5374) { - ret = -ENODEV; - goto reset_err; - } else { - dev_warn(codec->dev, - "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); - } + if (vendor_id2 == 0x5374) { + dev_warn(codec->dev, + "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); } /* unmute captures and playbacks volume */ diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 36d8425..69c63b9 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -865,7 +865,6 @@ const struct regmap_config adav80x_regmap_config = { .val_bits = 8, .pad_bits = 1, .reg_bits = 7, - .read_flag_mask = 0x01, .max_register = ADAV80X_PLL_OUTE, diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 66352f7..4a90143 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -64,12 +64,15 @@ #define FIL1_0 0x1c #define FIL1_1 0x1d #define FIL1_2 0x1e -#define FIL1_3 0x1f +#define FIL1_3 0x1f /* The maximum valid register for ak4642 */ #define PW_MGMT4 0x20 #define MD_CTL5 0x21 #define LO_MS 0x22 #define HP_MS 0x23 -#define SPK_MS 0x24 +#define SPK_MS 0x24 /* The maximum valid register for ak4643 */ +#define EQ_FBEQAB 0x25 +#define EQ_FBEQCD 0x26 +#define EQ_FBEQE 0x27 /* The maximum valid register for ak4648 */ /* PW_MGMT1*/ #define PMVCM (1 << 6) /* VCOM Power Management */ @@ -241,7 +244,7 @@ static const struct snd_soc_dapm_route ak4642_intercon[] = { /* * ak4642 register cache */ -static const struct reg_default ak4642_reg[] = { +static const struct reg_default ak4643_reg[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, { 8, 0xe1 }, { 9, 0xe1 }, { 10, 0x18 }, { 11, 0x00 }, @@ -254,6 +257,14 @@ static const struct reg_default ak4642_reg[] = { { 36, 0x00 }, }; +/* The default settings for 0x0 ~ 0x1f registers are the same for ak4642 + and ak4643. So we reuse the ak4643 reg_default for ak4642. + The valid registers for ak4642 are 0x0 ~ 0x1f which is a subset of ak4643, + so define NUM_AK4642_REG_DEFAULTS for ak4642. +*/ +#define ak4642_reg ak4643_reg +#define NUM_AK4642_REG_DEFAULTS (FIL1_3 + 1) + static const struct reg_default ak4648_reg[] = { { 0, 0x00 }, { 1, 0x00 }, { 2, 0x01 }, { 3, 0x00 }, { 4, 0x02 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, @@ -535,15 +546,23 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { static const struct regmap_config ak4642_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4642_reg) + 1, + .max_register = FIL1_3, .reg_defaults = ak4642_reg, - .num_reg_defaults = ARRAY_SIZE(ak4642_reg), + .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, +}; + +static const struct regmap_config ak4643_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = SPK_MS, + .reg_defaults = ak4643_reg, + .num_reg_defaults = ARRAY_SIZE(ak4643_reg), }; static const struct regmap_config ak4648_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(ak4648_reg) + 1, + .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), }; @@ -553,7 +572,7 @@ static const struct ak4642_drvdata ak4642_drvdata = { }; static const struct ak4642_drvdata ak4643_drvdata = { - .regmap_config = &ak4642_regmap, + .regmap_config = &ak4643_regmap, }; static const struct ak4642_drvdata ak4648_drvdata = { diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05e..2b55115 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1504,7 +1504,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; - wl = snd_pcm_format_width(params_format(params)); + wl = params_width(params); if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", @@ -1756,17 +1756,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_clock_ok(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "clock OK\n"); - - complete(&fll->ok); - - return IRQ_HANDLED; -} - static struct { unsigned int min; unsigned int max; @@ -2048,17 +2037,18 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll) static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - unsigned long time_left; bool use_sync = false; int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; + int i; + unsigned int val; if (already_enabled < 0) return already_enabled; if (already_enabled) { /* Facilitate smooth refclk across the transition */ - regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, @@ -2110,9 +2100,6 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (!already_enabled) pm_runtime_get(arizona->dev); - /* Clear any pending completions */ - try_wait_for_completion(&fll->ok); - regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) @@ -2124,10 +2111,24 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); - time_left = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(250)); - if (time_left == 0) + arizona_fll_dbg(fll, "Waiting for FLL lock...\n"); + val = 0; + for (i = 0; i < 15; i++) { + if (i < 5) + usleep_range(200, 400); + else + msleep(20); + + regmap_read(arizona->regmap, + ARIZONA_INTERRUPT_RAW_STATUS_5, + &val); + if (val & (ARIZONA_FLL1_CLOCK_OK_STS << (fll->id - 1))) + break; + } + if (i == 15) arizona_fll_warn(fll, "Timed out waiting for lock\n"); + else + arizona_fll_dbg(fll, "FLL locked (%d polls)\n", i); return 0; } @@ -2212,11 +2213,8 @@ EXPORT_SYMBOL_GPL(arizona_set_fll); int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll) { - int ret; unsigned int val; - init_completion(&fll->ok); - fll->id = id; fll->base = base; fll->arizona = arizona; @@ -2238,13 +2236,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, - arizona_fll_clock_ok, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", - id, ret); - } - regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); @@ -2313,6 +2304,82 @@ const struct snd_kcontrol_new arizona_adsp2_rate_controls[] = { }; EXPORT_SYMBOL_GPL(arizona_adsp2_rate_controls); +static bool arizona_eq_filter_unstable(bool mode, __be16 _a, __be16 _b) +{ + s16 a = be16_to_cpu(_a); + s16 b = be16_to_cpu(_b); + + if (!mode) { + return abs(a) >= 4096; + } else { + if (abs(b) >= 4096) + return true; + + return (abs((a << 16) / (4096 - b)) >= 4096 << 4); + } +} + +int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct soc_bytes *params = (void *)kcontrol->private_value; + unsigned int val; + __be16 *data; + int len; + int ret; + + len = params->num_regs * regmap_get_val_bytes(arizona->regmap); + + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + data[0] &= cpu_to_be16(ARIZONA_EQ1_B1_MODE); + + if (arizona_eq_filter_unstable(!!data[0], data[1], data[2]) || + arizona_eq_filter_unstable(true, data[4], data[5]) || + arizona_eq_filter_unstable(true, data[8], data[9]) || + arizona_eq_filter_unstable(true, data[12], data[13]) || + arizona_eq_filter_unstable(false, data[16], data[17])) { + dev_err(arizona->dev, "Rejecting unstable EQ coefficients\n"); + ret = -EINVAL; + goto out; + } + + ret = regmap_read(arizona->regmap, params->base, &val); + if (ret != 0) + goto out; + + val &= ~ARIZONA_EQ1_B1_MODE; + data[0] |= cpu_to_be16(val); + + ret = regmap_raw_write(arizona->regmap, params->base, data, len); + +out: + kfree(data); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_eq_coeff_put); + +int arizona_lhpf_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + __be16 *data = (__be16 *)ucontrol->value.bytes.data; + s16 val = be16_to_cpu(*data); + + if (abs(val) >= 4096) { + dev_err(arizona->dev, "Rejecting unstable LHPF coefficients\n"); + return -EINVAL; + } + + return snd_soc_bytes_put(kcontrol, ucontrol); +} +EXPORT_SYMBOL_GPL(arizona_lhpf_coeff_put); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 43deb04..ada0a41 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -194,6 +194,20 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name " Preloader", name "L"), \ ARIZONA_MIXER_ROUTES(name " Preloader", name "R") +#define ARIZONA_EQ_CONTROL(xname, xbase) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ + .put = arizona_eq_coeff_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) { .base = xbase, \ + .num_regs = 20, .mask = ~ARIZONA_EQ1_B1_MODE }) } + +#define ARIZONA_LHPF_CONTROL(xname, xbase) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ + .put = arizona_lhpf_coeff_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) { .base = xbase, \ + .num_regs = 1 }) } + #define ARIZONA_RATE_ENUM_SIZE 4 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; @@ -229,6 +243,11 @@ extern int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +extern int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +extern int arizona_lhpf_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); @@ -242,7 +261,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion ok; unsigned int fout; int sync_src; diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 76564dc..094201d 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -74,33 +74,8 @@ static const struct reg_default cs35l32_reg_defaults[] = { static bool cs35l32_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_PWRCTL1: - case CS35L32_PWRCTL2: - case CS35L32_CLK_CTL: - case CS35L32_BATT_THRESHOLD: - case CS35L32_VMON: - case CS35L32_BST_CPCP_CTL: - case CS35L32_IMON_SCALING: - case CS35L32_AUDIO_LED_MNGR: - case CS35L32_ADSP_CTL: - case CS35L32_CLASSD_CTL: - case CS35L32_PROTECT_CTL: - case CS35L32_INT_MASK_1: - case CS35L32_INT_MASK_2: - case CS35L32_INT_MASK_3: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: - case CS35L32_FLASH_MODE: - case CS35L32_MOVIE_MODE: - case CS35L32_FLASH_TIMER: - case CS35L32_FLASH_INHIBIT: + case CS35L32_DEVID_AB ... CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL ... CS35L32_FLASH_INHIBIT: return true; default: return false; @@ -110,15 +85,8 @@ static bool cs35l32_readable_register(struct device *dev, unsigned int reg) static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_DEVID_AB ... CS35L32_REV_ID: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; @@ -128,10 +96,7 @@ static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) static bool cs35l32_precious_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; @@ -441,8 +406,7 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, if (IS_ERR(cs35l32->reset_gpio)) return PTR_ERR(cs35l32->reset_gpio); - if (cs35l32->reset_gpio) - gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); @@ -536,8 +500,7 @@ static int cs35l32_i2c_remove(struct i2c_client *i2c_client) snd_soc_unregister_codec(&i2c_client->dev); /* Hold down reset */ - if (cs35l32->reset_gpio) - gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); return 0; } @@ -551,8 +514,7 @@ static int cs35l32_runtime_suspend(struct device *dev) regcache_mark_dirty(cs35l32->regmap); /* Hold down reset */ - if (cs35l32->reset_gpio) - gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); /* remove power */ regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), @@ -575,8 +537,7 @@ static int cs35l32_runtime_resume(struct device *dev) return ret; } - if (cs35l32->reset_gpio) - gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); + gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); regcache_cache_only(cs35l32->regmap, false); regcache_sync(cs35l32->regmap); diff --git a/sound/soc/codecs/cs35l32.h b/sound/soc/codecs/cs35l32.h index 31ab804..1d6c250 100644 --- a/sound/soc/codecs/cs35l32.h +++ b/sound/soc/codecs/cs35l32.h @@ -80,7 +80,7 @@ struct cs35l32_platform_data { #define CS35L32_GAIN_MGR_MASK 0x08 #define CS35L32_ADSP_SHARE_MASK 0x08 #define CS35L32_ADSP_DATACFG_MASK 0x30 -#define CS35L32_SDOUT_3ST 0x80 +#define CS35L32_SDOUT_3ST 0x08 #define CS35L32_BATT_REC_MASK 0x0E #define CS35L32_BATT_THRESH_MASK 0x30 diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d1a77c7..55db19d 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,23 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_PWRCTL: - case CS4265_DAC_CTL: - case CS4265_ADC_CTL: - case CS4265_MCLK_FREQ: - case CS4265_SIG_SEL: - case CS4265_CHB_PGA_CTL: - case CS4265_CHA_PGA_CTL: - case CS4265_ADC_CTL2: - case CS4265_DAC_CHA_VOL: - case CS4265_DAC_CHB_VOL: - case CS4265_DAC_CTL2: - case CS4265_SPDIF_CTL1: - case CS4265_SPDIF_CTL2: - case CS4265_INT_MASK: - case CS4265_STATUS_MODE_MSB: - case CS4265_STATUS_MODE_LSB: - case CS4265_CHIP_ID: + case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: return true; default: return false; @@ -457,14 +441,14 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_RIGHT_J: if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (1 << 5)); + CS4265_DAC_CTL_DIF, (2 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (2 << 6)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (3 << 5)); + CS4265_DAC_CTL_DIF, (3 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (3 << 6)); } break; case SND_SOC_DAIFMT_LEFT_J: @@ -473,7 +457,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 6)); + CS4265_SPDIF_CTL2_DIF, 0); break; default: diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index b82d8e5..f4f41b2 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -110,58 +110,7 @@ static const struct reg_default cs42l52_reg_defaults[] = { static bool cs42l52_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS42L52_CHIP: - case CS42L52_PWRCTL1: - case CS42L52_PWRCTL2: - case CS42L52_PWRCTL3: - case CS42L52_CLK_CTL: - case CS42L52_IFACE_CTL1: - case CS42L52_IFACE_CTL2: - case CS42L52_ADC_PGA_A: - case CS42L52_ADC_PGA_B: - case CS42L52_ANALOG_HPF_CTL: - case CS42L52_ADC_HPF_FREQ: - case CS42L52_ADC_MISC_CTL: - case CS42L52_PB_CTL1: - case CS42L52_MISC_CTL: - case CS42L52_PB_CTL2: - case CS42L52_MICA_CTL: - case CS42L52_MICB_CTL: - case CS42L52_PGAA_CTL: - case CS42L52_PGAB_CTL: - case CS42L52_PASSTHRUA_VOL: - case CS42L52_PASSTHRUB_VOL: - case CS42L52_ADCA_VOL: - case CS42L52_ADCB_VOL: - case CS42L52_ADCA_MIXER_VOL: - case CS42L52_ADCB_MIXER_VOL: - case CS42L52_PCMA_MIXER_VOL: - case CS42L52_PCMB_MIXER_VOL: - case CS42L52_BEEP_FREQ: - case CS42L52_BEEP_VOL: - case CS42L52_BEEP_TONE_CTL: - case CS42L52_TONE_CTL: - case CS42L52_MASTERA_VOL: - case CS42L52_MASTERB_VOL: - case CS42L52_HPA_VOL: - case CS42L52_HPB_VOL: - case CS42L52_SPKA_VOL: - case CS42L52_SPKB_VOL: - case CS42L52_ADC_PCM_MIXER: - case CS42L52_LIMITER_CTL1: - case CS42L52_LIMITER_CTL2: - case CS42L52_LIMITER_AT_RATE: - case CS42L52_ALC_CTL: - case CS42L52_ALC_RATE: - case CS42L52_ALC_THRESHOLD: - case CS42L52_NOISE_GATE_CTL: - case CS42L52_CLK_STATUS: - case CS42L52_BATT_COMPEN: - case CS42L52_BATT_LEVEL: - case CS42L52_SPK_STATUS: - case CS42L52_TEM_CTL: - case CS42L52_THE_FOLDBACK: - case CS42L52_CHARGE_PUMP: + case CS42L52_CHIP ... CS42L52_CHARGE_PUMP: return true; default: return false; @@ -919,7 +868,7 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) -static struct snd_soc_dai_ops cs42l52_ops = { +static const struct snd_soc_dai_ops cs42l52_ops = { .hw_params = cs42l52_pcm_hw_params, .digital_mute = cs42l52_digital_mute, .set_fmt = cs42l52_set_fmt, diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 4ae7933..52fe7a5 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -115,52 +115,7 @@ static const struct reg_default cs42l56_reg_defaults[] = { static bool cs42l56_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS42L56_CHIP_ID_1: - case CS42L56_CHIP_ID_2: - case CS42L56_PWRCTL_1: - case CS42L56_PWRCTL_2: - case CS42L56_CLKCTL_1: - case CS42L56_CLKCTL_2: - case CS42L56_SERIAL_FMT: - case CS42L56_CLASSH_CTL: - case CS42L56_MISC_CTL: - case CS42L56_INT_STATUS: - case CS42L56_PLAYBACK_CTL: - case CS42L56_DSP_MUTE_CTL: - case CS42L56_ADCA_MIX_VOLUME: - case CS42L56_ADCB_MIX_VOLUME: - case CS42L56_PCMA_MIX_VOLUME: - case CS42L56_PCMB_MIX_VOLUME: - case CS42L56_ANAINPUT_ADV_VOLUME: - case CS42L56_DIGINPUT_ADV_VOLUME: - case CS42L56_MASTER_A_VOLUME: - case CS42L56_MASTER_B_VOLUME: - case CS42L56_BEEP_FREQ_ONTIME: - case CS42L56_BEEP_FREQ_OFFTIME: - case CS42L56_BEEP_TONE_CFG: - case CS42L56_TONE_CTL: - case CS42L56_CHAN_MIX_SWAP: - case CS42L56_AIN_REFCFG_ADC_MUX: - case CS42L56_HPF_CTL: - case CS42L56_MISC_ADC_CTL: - case CS42L56_GAIN_BIAS_CTL: - case CS42L56_PGAA_MUX_VOLUME: - case CS42L56_PGAB_MUX_VOLUME: - case CS42L56_ADCA_ATTENUATOR: - case CS42L56_ADCB_ATTENUATOR: - case CS42L56_ALC_EN_ATTACK_RATE: - case CS42L56_ALC_RELEASE_RATE: - case CS42L56_ALC_THRESHOLD: - case CS42L56_NOISE_GATE_CTL: - case CS42L56_ALC_LIM_SFT_ZC: - case CS42L56_AMUTE_HPLO_MUX: - case CS42L56_HPA_VOLUME: - case CS42L56_HPB_VOLUME: - case CS42L56_LOA_VOLUME: - case CS42L56_LOB_VOLUME: - case CS42L56_LIM_THRESHOLD_CTL: - case CS42L56_LIM_CTL_RELEASE_RATE: - case CS42L56_LIM_ATTACK_RATE: + case CS42L56_CHIP_ID_1 ... CS42L56_LIM_ATTACK_RATE: return true; default: return false; @@ -989,7 +944,7 @@ static int cs42l56_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops cs42l56_ops = { +static const struct snd_soc_dai_ops cs42l56_ops = { .hw_params = cs42l56_pcm_hw_params, .digital_mute = cs42l56_digital_mute, .set_fmt = cs42l56_set_dai_fmt, diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 7cb1d70..a8f4686 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -153,100 +153,8 @@ static bool cs42l73_volatile_register(struct device *dev, unsigned int reg) static bool cs42l73_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS42L73_DEVID_AB: - case CS42L73_DEVID_CD: - case CS42L73_DEVID_E: - case CS42L73_REVID: - case CS42L73_PWRCTL1: - case CS42L73_PWRCTL2: - case CS42L73_PWRCTL3: - case CS42L73_CPFCHC: - case CS42L73_OLMBMSDC: - case CS42L73_DMMCC: - case CS42L73_XSPC: - case CS42L73_XSPMMCC: - case CS42L73_ASPC: - case CS42L73_ASPMMCC: - case CS42L73_VSPC: - case CS42L73_VSPMMCC: - case CS42L73_VXSPFS: - case CS42L73_MIOPC: - case CS42L73_ADCIPC: - case CS42L73_MICAPREPGAAVOL: - case CS42L73_MICBPREPGABVOL: - case CS42L73_IPADVOL: - case CS42L73_IPBDVOL: - case CS42L73_PBDC: - case CS42L73_HLADVOL: - case CS42L73_HLBDVOL: - case CS42L73_SPKDVOL: - case CS42L73_ESLDVOL: - case CS42L73_HPAAVOL: - case CS42L73_HPBAVOL: - case CS42L73_LOAAVOL: - case CS42L73_LOBAVOL: - case CS42L73_STRINV: - case CS42L73_XSPINV: - case CS42L73_ASPINV: - case CS42L73_VSPINV: - case CS42L73_LIMARATEHL: - case CS42L73_LIMRRATEHL: - case CS42L73_LMAXHL: - case CS42L73_LIMARATESPK: - case CS42L73_LIMRRATESPK: - case CS42L73_LMAXSPK: - case CS42L73_LIMARATEESL: - case CS42L73_LIMRRATEESL: - case CS42L73_LMAXESL: - case CS42L73_ALCARATE: - case CS42L73_ALCRRATE: - case CS42L73_ALCMINMAX: - case CS42L73_NGCAB: - case CS42L73_ALCNGMC: - case CS42L73_MIXERCTL: - case CS42L73_HLAIPAA: - case CS42L73_HLBIPBA: - case CS42L73_HLAXSPAA: - case CS42L73_HLBXSPBA: - case CS42L73_HLAASPAA: - case CS42L73_HLBASPBA: - case CS42L73_HLAVSPMA: - case CS42L73_HLBVSPMA: - case CS42L73_XSPAIPAA: - case CS42L73_XSPBIPBA: - case CS42L73_XSPAXSPAA: - case CS42L73_XSPBXSPBA: - case CS42L73_XSPAASPAA: - case CS42L73_XSPAASPBA: - case CS42L73_XSPAVSPMA: - case CS42L73_XSPBVSPMA: - case CS42L73_ASPAIPAA: - case CS42L73_ASPBIPBA: - case CS42L73_ASPAXSPAA: - case CS42L73_ASPBXSPBA: - case CS42L73_ASPAASPAA: - case CS42L73_ASPBASPBA: - case CS42L73_ASPAVSPMA: - case CS42L73_ASPBVSPMA: - case CS42L73_VSPAIPAA: - case CS42L73_VSPBIPBA: - case CS42L73_VSPAXSPAA: - case CS42L73_VSPBXSPBA: - case CS42L73_VSPAASPAA: - case CS42L73_VSPBASPBA: - case CS42L73_VSPAVSPMA: - case CS42L73_VSPBVSPMA: - case CS42L73_MMIXCTL: - case CS42L73_SPKMIPMA: - case CS42L73_SPKMXSPA: - case CS42L73_SPKMASPA: - case CS42L73_SPKMVSPMA: - case CS42L73_ESLMIPMA: - case CS42L73_ESLMXSPA: - case CS42L73_ESLMASPA: - case CS42L73_ESLMVSPMA: - case CS42L73_IM1: - case CS42L73_IM2: + case CS42L73_DEVID_AB ... CS42L73_DEVID_E: + case CS42L73_REVID ... CS42L73_IM2: return true; default: return false; @@ -1236,8 +1144,8 @@ static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) struct snd_soc_codec *codec = dai->codec; int id = dai->id; - return snd_soc_update_bits(codec, CS42L73_SPC(id), - 0x7F, tristate << 7); + return snd_soc_update_bits(codec, CS42L73_SPC(id), CS42L73_SP_3ST, + tristate << 7); } static const struct snd_pcm_hw_constraint_list constraints_12_24 = { diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c new file mode 100644 index 0000000..0ac8fc5 --- /dev/null +++ b/sound/soc/codecs/cs4349.c @@ -0,0 +1,392 @@ +/* + * cs4349.c -- CS4349 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Tim Howe <Tim.Howe@cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.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 "cs4349.h" + + +static const struct reg_default cs4349_reg_defaults[] = { + { 2, 0x00 }, /* r02 - Mode Control */ + { 3, 0x09 }, /* r03 - Volume, Mixing and Inversion Control */ + { 4, 0x81 }, /* r04 - Mute Control */ + { 5, 0x00 }, /* r05 - Channel A Volume Control */ + { 6, 0x00 }, /* r06 - Channel B Volume Control */ + { 7, 0xB1 }, /* r07 - Ramp and Filter Control */ + { 8, 0x1C }, /* r08 - Misc. Control */ +}; + +/* Private data for the CS4349 */ +struct cs4349_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + unsigned int mode; + int rate; +}; + +static bool cs4349_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_CHIPID ... CS4349_MISC: + return true; + default: + return false; + } +} + +static bool cs4349_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_MODE ... CS4349_MISC: + return true; + default: + return false; + } +} + +static int cs4349_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + unsigned int fmt; + + fmt = format & SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + cs4349->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs4349_pcm_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 cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); + int fmt, ret; + + cs4349->rate = params_rate(params); + + switch (cs4349->mode) { + case SND_SOC_DAIFMT_I2S: + fmt = DIF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + fmt = DIF_LEFT_JST; + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + fmt = DIF_RGHT_JST16; + break; + case 24: + fmt = DIF_RGHT_JST24; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, CS4349_MODE, DIF_MASK, + MODE_FORMAT(fmt)); + if (ret < 0) + return ret; + + return 0; +} + +static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = 0; + if (mute) + reg = MUTE_AB_MASK; + + return snd_soc_update_bits(codec, CS4349_MUTE, MUTE_AB_MASK, reg); +} + +static DECLARE_TLV_DB_SCALE(dig_tlv, -12750, 50, 0); + +static const char * const chan_mix_texts[] = { + "Mute", "MuteA", "MuteA SwapB", "MuteA MonoB", "SwapA MuteB", + "BothR", "Swap", "SwapA MonoB", "MuteB", "Normal", "BothL", + "MonoB", "MonoA MuteB", "MonoA", "MonoA SwapB", "Mono", + /*Normal == Channel A = Left, Channel B = Right*/ +}; + +static const char * const fm_texts[] = { + "Auto", "Single", "Double", "Quad", +}; + +static const char * const deemph_texts[] = { + "None", "44.1k", "48k", "32k", +}; + +static const char * const softr_zeroc_texts[] = { + "Immediate", "Zero Cross", "Soft Ramp", "SR on ZC", +}; + +static int deemph_values[] = { + 0, 4, 8, 12, +}; + +static int softr_zeroc_values[] = { + 0, 64, 128, 192, +}; + +static const struct soc_enum chan_mix_enum = + SOC_ENUM_SINGLE(CS4349_VMI, 0, + ARRAY_SIZE(chan_mix_texts), + chan_mix_texts); + +static const struct soc_enum fm_mode_enum = + SOC_ENUM_SINGLE(CS4349_MODE, 0, + ARRAY_SIZE(fm_texts), + fm_texts); + +static SOC_VALUE_ENUM_SINGLE_DECL(deemph_enum, CS4349_MODE, 0, DEM_MASK, + deemph_texts, deemph_values); + +static SOC_VALUE_ENUM_SINGLE_DECL(softr_zeroc_enum, CS4349_RMPFLT, 0, + SR_ZC_MASK, softr_zeroc_texts, + softr_zeroc_values); + +static const struct snd_kcontrol_new cs4349_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + CS4349_VOLA, CS4349_VOLB, 0, 0xFF, 1, dig_tlv), + SOC_ENUM("Functional Mode", fm_mode_enum), + SOC_ENUM("De-Emphasis Control", deemph_enum), + SOC_ENUM("Soft Ramp Zero Cross Control", softr_zeroc_enum), + SOC_ENUM("Channel Mixer", chan_mix_enum), + SOC_SINGLE("VolA = VolB Switch", CS4349_VMI, 7, 1, 0), + SOC_SINGLE("InvertA Switch", CS4349_VMI, 6, 1, 0), + SOC_SINGLE("InvertB Switch", CS4349_VMI, 5, 1, 0), + SOC_SINGLE("Auto-Mute Switch", CS4349_MUTE, 7, 1, 0), + SOC_SINGLE("MUTEC A = B Switch", CS4349_MUTE, 5, 1, 0), + SOC_SINGLE("Soft Ramp Up Switch", CS4349_RMPFLT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Down Switch", CS4349_RMPFLT, 4, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4349_RMPFLT, 2, 1, 0), + SOC_SINGLE("Freeze Switch", CS4349_MISC, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4349_MISC, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget cs4349_dapm_widgets[] = { + SND_SOC_DAPM_DAC("HiFi DAC", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("OutputA"), + SND_SOC_DAPM_OUTPUT("OutputB"), +}; + +static const struct snd_soc_dapm_route cs4349_routes[] = { + {"DAC Playback", NULL, "OutputA"}, + {"DAC Playback", NULL, "OutputB"}, + + {"OutputA", NULL, "HiFi DAC"}, + {"OutputB", NULL, "HiFi DAC"}, +}; + +#define CS4349_PCM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + 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) + +#define CS4349_PCM_RATES SNDRV_PCM_RATE_8000_192000 + +static const struct snd_soc_dai_ops cs4349_dai_ops = { + .hw_params = cs4349_pcm_hw_params, + .set_fmt = cs4349_set_dai_fmt, + .digital_mute = cs4349_digital_mute, +}; + +static struct snd_soc_dai_driver cs4349_dai = { + .name = "cs4349_hifi", + .playback = { + .stream_name = "DAC Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS4349_PCM_RATES, + .formats = CS4349_PCM_FORMATS, + }, + .ops = &cs4349_dai_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs4349 = { + .controls = cs4349_snd_controls, + .num_controls = ARRAY_SIZE(cs4349_snd_controls), + + .dapm_widgets = cs4349_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4349_dapm_widgets), + .dapm_routes = cs4349_routes, + .num_dapm_routes = ARRAY_SIZE(cs4349_routes), +}; + +static const struct regmap_config cs4349_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS4349_MISC, + .reg_defaults = cs4349_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), + .readable_reg = cs4349_readable_register, + .writeable_reg = cs4349_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs4349_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4349_private *cs4349; + int ret; + + cs4349 = devm_kzalloc(&client->dev, sizeof(*cs4349), GFP_KERNEL); + if (!cs4349) + return -ENOMEM; + + cs4349->regmap = devm_regmap_init_i2c(client, &cs4349_regmap); + if (IS_ERR(cs4349->regmap)) { + ret = PTR_ERR(cs4349->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + /* Reset the Device */ + cs4349->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4349->reset_gpio)) + return PTR_ERR(cs4349->reset_gpio); + + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + i2c_set_clientdata(client, cs4349); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4349, + &cs4349_dai, 1); +} + +static int cs4349_i2c_remove(struct i2c_client *client) +{ + struct cs4349_private *cs4349 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs4349_runtime_suspend(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, PWR_DWN); + if (ret < 0) + return ret; + + regcache_cache_only(cs4349->regmap, true); + + /* Hold down reset */ + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + + return 0; +} + +static int cs4349_runtime_resume(struct device *dev) +{ + struct cs4349_private *cs4349 = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, 0); + if (ret < 0) + return ret; + + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + + regcache_cache_only(cs4349->regmap, false); + regcache_sync(cs4349->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs4349_runtime_pm = { + SET_RUNTIME_PM_OPS(cs4349_runtime_suspend, cs4349_runtime_resume, + NULL) +}; + +static const struct of_device_id cs4349_of_match[] = { + { .compatible = "cirrus,cs4349", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs4349_of_match); + +static const struct i2c_device_id cs4349_i2c_id[] = { + {"cs4349", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs4349_i2c_id); + +static struct i2c_driver cs4349_i2c_driver = { + .driver = { + .name = "cs4349", + .of_match_table = cs4349_of_match, + }, + .id_table = cs4349_i2c_id, + .probe = cs4349_i2c_probe, + .remove = cs4349_i2c_remove, +}; + +module_i2c_driver(cs4349_i2c_driver); + +MODULE_AUTHOR("Tim Howe <tim.howe@cirrus.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4349 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4349.h b/sound/soc/codecs/cs4349.h new file mode 100644 index 0000000..d58c06a --- /dev/null +++ b/sound/soc/codecs/cs4349.h @@ -0,0 +1,136 @@ +/* + * ALSA SoC CS4349 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Tim Howe <Tim.Howe@cirrus.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. + * + */ + +#ifndef __CS4349_H__ +#define __CS4349_H__ + +/* CS4349 registers addresses */ +#define CS4349_CHIPID 0x01 /* Device and Rev ID, Read Only */ +#define CS4349_MODE 0x02 /* Mode Control */ +#define CS4349_VMI 0x03 /* Volume, Mixing, Inversion Control */ +#define CS4349_MUTE 0x04 /* Mute Control */ +#define CS4349_VOLA 0x05 /* DAC Channel A Volume Control */ +#define CS4349_VOLB 0x06 /* DAC Channel B Volume Control */ +#define CS4349_RMPFLT 0x07 /* Ramp and Filter Control */ +#define CS4349_MISC 0x08 /* Power Down,Freeze Control,Pop Stop*/ + +#define CS4349_I2C_INCR 0x80 + + +/* Device and Revision ID */ +#define CS4349_REVA 0xF0 /* Rev A */ +#define CS4349_REVB 0xF1 /* Rev B */ +#define CS4349_REVC2 0xFF /* Rev C2 */ + + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define PDN_POLL_MAX 900 + + +/* Bitfield Definitions */ + +/* CS4349_MODE */ +/* (Digital Interface Format, De-Emphasis Control, Functional Mode */ +#define DIF2 (1 << 6) +#define DIF1 (1 << 5) +#define DIF0 (1 << 4) +#define DEM1 (1 << 3) +#define DEM0 (1 << 2) +#define FM1 (1 << 1) +#define DIF_LEFT_JST 0x00 +#define DIF_I2S 0x01 +#define DIF_RGHT_JST16 0x02 +#define DIF_RGHT_JST24 0x03 +#define DIF_TDM0 0x04 +#define DIF_TDM1 0x05 +#define DIF_TDM2 0x06 +#define DIF_TDM3 0x07 +#define DIF_MASK 0x70 +#define MODE_FORMAT(x) (((x)&7)<<4) +#define DEM_MASK 0x0C +#define NO_DEM 0x00 +#define DEM_441 0x04 +#define DEM_48K 0x08 +#define DEM_32K 0x0C +#define FM_AUTO 0x00 +#define FM_SNGL 0x01 +#define FM_DBL 0x02 +#define FM_QUAD 0x03 +#define FM_SNGL_MIN 30000 +#define FM_SNGL_MAX 54000 +#define FM_DBL_MAX 108000 +#define FM_QUAD_MAX 216000 +#define FM_MASK 0x03 + +/* CS4349_VMI (VMI = Volume, Mixing and Inversion Controls) */ +#define VOLBISA (1 << 7) +#define VOLAISB (1 << 7) +/* INVERT_A only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_A (1 << 6) +/* INVERT_B only available for Left Jstfd, Right Jstfd16 and Right Jstfd24 */ +#define INVERT_B (1 << 5) +#define ATAPI3 (1 << 3) +#define ATAPI2 (1 << 2) +#define ATAPI1 (1 << 1) +#define ATAPI0 (1 << 0) +#define MUTEAB 0x00 +#define MUTEA_RIGHTB 0x01 +#define MUTEA_LEFTB 0x02 +#define MUTEA_SUMLRDIV2B 0x03 +#define RIGHTA_MUTEB 0x04 +#define RIGHTA_RIGHTB 0x05 +#define RIGHTA_LEFTB 0x06 +#define RIGHTA_SUMLRDIV2B 0x07 +#define LEFTA_MUTEB 0x08 +#define LEFTA_RIGHTB 0x09 /* Default */ +#define LEFTA_LEFTB 0x0A +#define LEFTA_SUMLRDIV2B 0x0B +#define SUMLRDIV2A_MUTEB 0x0C +#define SUMLRDIV2A_RIGHTB 0x0D +#define SUMLRDIV2A_LEFTB 0x0E +#define SUMLRDIV2_AB 0x0F +#define CHMIX_MASK 0x0F + +/* CS4349_MUTE */ +#define AUTOMUTE (1 << 7) +#define MUTEC_AB (1 << 5) +#define MUTE_A (1 << 4) +#define MUTE_B (1 << 3) +#define MUTE_AB_MASK 0x18 + +/* CS4349_RMPFLT (Ramp and Filter Control) */ +#define SCZ1 (1 << 7) +#define SCZ0 (1 << 6) +#define RMP_UP (1 << 5) +#define RMP_DN (1 << 4) +#define FILT_SEL (1 << 2) +#define IMMDT_CHNG 0x31 +#define ZEROCRSS 0x71 +#define SOFT_RMP 0xB1 +#define SFTRMP_ZEROCRSS 0xF1 +#define SR_ZC_MASK 0xC0 + +/* CS4349_MISC */ +#define PWR_DWN (1 << 7) +#define FREEZE (1 << 5) +#define POPG_EN (1 << 4) + +#endif /* __CS4349_H__ */ diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 5446d04..295f0c7 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1196,13 +1196,7 @@ static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, #define DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops da732x_dai1_ops = { - .hw_params = da732x_hw_params, - .set_fmt = da732x_set_dai_fmt, - .set_sysclk = da732x_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops da732x_dai2_ops = { +static const struct snd_soc_dai_ops da732x_dai_ops = { .hw_params = da732x_hw_params, .set_fmt = da732x_set_dai_fmt, .set_sysclk = da732x_set_dai_sysclk, @@ -1227,7 +1221,7 @@ static struct snd_soc_dai_driver da732x_dai[] = { .rates = DA732X_RATES, .formats = DA732X_FORMATS, }, - .ops = &da732x_dai1_ops, + .ops = &da732x_dai_ops, }, { .name = "DA732X_AIFB", @@ -1247,7 +1241,7 @@ static struct snd_soc_dai_driver da732x_dai[] = { .rates = DA732X_RATES, .formats = DA732X_FORMATS, }, - .ops = &da732x_dai2_ops, + .ops = &da732x_dai_ops, }, }; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 7d5baa6..ede9bc4 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1533,6 +1533,7 @@ static const struct of_device_id da9055_of_match[] = { { .compatible = "dlg,da9055-codec", }, { } }; +MODULE_DEVICE_TABLE(of, da9055_of_match); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { diff --git a/sound/soc/codecs/gtm601.c b/sound/soc/codecs/gtm601.c new file mode 100644 index 0000000..0b80052 --- /dev/null +++ b/sound/soc/codecs/gtm601.c @@ -0,0 +1,95 @@ +/* + * This is a simple driver for the GTM601 Voice PCM interface + * + * Copyright (C) 2015 Goldelico GmbH + * + * Author: Marek Belisko <marek@goldelico.com> + * + * Based on wm8727.c driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/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 const struct snd_soc_dapm_widget gtm601_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("AOUT"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const struct snd_soc_dapm_route gtm601_dapm_routes[] = { + { "AOUT", NULL, "Playback" }, + { "Capture", NULL, "AIN" }, +}; + +static struct snd_soc_dai_driver gtm601_dai = { + .name = "gtm601", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static const struct snd_soc_codec_driver soc_codec_dev_gtm601 = { + .dapm_widgets = gtm601_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(gtm601_dapm_widgets), + .dapm_routes = gtm601_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(gtm601_dapm_routes), +}; + +static int gtm601_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_gtm601, >m601_dai, 1); +} + +static int gtm601_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#if defined(CONFIG_OF) +static const struct of_device_id gtm601_codec_of_match[] = { + { .compatible = "option,gtm601", }, + {}, +}; +MODULE_DEVICE_TABLE(of, gtm601_codec_of_match); +#endif + +static struct platform_driver gtm601_codec_driver = { + .driver = { + .name = "gtm601", + .of_match_table = of_match_ptr(gtm601_codec_of_match), + }, + .probe = gtm601_platform_probe, + .remove = gtm601_platform_remove, +}; + +module_platform_driver(gtm601_codec_driver); + +MODULE_DESCRIPTION("ASoC gtm601 driver"); +MODULE_AUTHOR("Marek Belisko <marek@goldelico.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:gtm601"); diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..dd850b9 --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,76 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL v2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw@axis.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 58a43b1..6bb8e6b 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1016,25 +1016,25 @@ static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops isabelle_hs_dai_ops = { +static const struct snd_soc_dai_ops isabelle_hs_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_hs_mute, }; -static struct snd_soc_dai_ops isabelle_hf_dai_ops = { +static const struct snd_soc_dai_ops isabelle_hf_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_hf_mute, }; -static struct snd_soc_dai_ops isabelle_line_dai_ops = { +static const struct snd_soc_dai_ops isabelle_line_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, .digital_mute = isabelle_line_mute, }; -static struct snd_soc_dai_ops isabelle_ul_dai_ops = { +static const struct snd_soc_dai_ops isabelle_ul_dai_ops = { .hw_params = isabelle_hw_params, .set_fmt = isabelle_set_dai_fmt, }; diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 9b2e383..af4e35e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -188,7 +188,6 @@ static struct reg_default lm49453_reg_defs[] = { /* codec private data */ struct lm49453_priv { struct regmap *regmap; - int fs_rate; }; /* capture path controls */ @@ -1112,13 +1111,10 @@ static int lm49453_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); u16 clk_div = 0; - lm49453->fs_rate = params_rate(params); - /* Setting DAC clock dividers based on substream sample rate. */ - switch (lm49453->fs_rate) { + switch (params_rate(params)) { case 8000: case 16000: case 32000: @@ -1291,35 +1287,35 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec, #define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops lm49453_headset_dai_ops = { +static const struct snd_soc_dai_ops lm49453_headset_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_hp_mute, }; -static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { +static const struct snd_soc_dai_ops lm49453_speaker_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ls_mute, }; -static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { +static const struct snd_soc_dai_ops lm49453_haptic_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ha_mute, }; -static struct snd_soc_dai_ops lm49453_ep_dai_ops = { +static const struct snd_soc_dai_ops lm49453_ep_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, .digital_mute = lm49453_ep_mute, }; -static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { +static const struct snd_soc_dai_ops lm49453_lineout_dai_ops = { .hw_params = lm49453_hw_params, .set_sysclk = lm49453_set_dai_sysclk, .set_fmt = lm49453_set_dai_fmt, diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 99c2daa..2c2df17 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -258,292 +258,36 @@ static const struct reg_default max98088_reg[] = { { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; -static struct { - int readable; - int writable; - int vol; -} max98088_access[M98088_REG_CNT] = { - { 0xFF, 0xFF, 1 }, /* 00 IRQ status */ - { 0xFF, 0x00, 1 }, /* 01 MIC status */ - { 0xFF, 0x00, 1 }, /* 02 jack status */ - { 0x1F, 0x1F, 1 }, /* 03 battery voltage */ - { 0xFF, 0xFF, 0 }, /* 04 */ - { 0xFF, 0xFF, 0 }, /* 05 */ - { 0xFF, 0xFF, 0 }, /* 06 */ - { 0xFF, 0xFF, 0 }, /* 07 */ - { 0xFF, 0xFF, 0 }, /* 08 */ - { 0xFF, 0xFF, 0 }, /* 09 */ - { 0xFF, 0xFF, 0 }, /* 0A */ - { 0xFF, 0xFF, 0 }, /* 0B */ - { 0xFF, 0xFF, 0 }, /* 0C */ - { 0xFF, 0xFF, 0 }, /* 0D */ - { 0xFF, 0xFF, 0 }, /* 0E */ - { 0xFF, 0xFF, 0 }, /* 0F interrupt enable */ - - { 0xFF, 0xFF, 0 }, /* 10 master clock */ - { 0xFF, 0xFF, 0 }, /* 11 DAI1 clock mode */ - { 0xFF, 0xFF, 0 }, /* 12 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 13 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 14 DAI1 format */ - { 0xFF, 0xFF, 0 }, /* 15 DAI1 clock */ - { 0xFF, 0xFF, 0 }, /* 16 DAI1 config */ - { 0xFF, 0xFF, 0 }, /* 17 DAI1 TDM */ - { 0xFF, 0xFF, 0 }, /* 18 DAI1 filters */ - { 0xFF, 0xFF, 0 }, /* 19 DAI2 clock mode */ - { 0xFF, 0xFF, 0 }, /* 1A DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1B DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1C DAI2 format */ - { 0xFF, 0xFF, 0 }, /* 1D DAI2 clock */ - { 0xFF, 0xFF, 0 }, /* 1E DAI2 config */ - { 0xFF, 0xFF, 0 }, /* 1F DAI2 TDM */ - - { 0xFF, 0xFF, 0 }, /* 20 DAI2 filters */ - { 0xFF, 0xFF, 0 }, /* 21 data config */ - { 0xFF, 0xFF, 0 }, /* 22 DAC mixer */ - { 0xFF, 0xFF, 0 }, /* 23 left ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 24 right ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 25 left HP mixer */ - { 0xFF, 0xFF, 0 }, /* 26 right HP mixer */ - { 0xFF, 0xFF, 0 }, /* 27 HP control */ - { 0xFF, 0xFF, 0 }, /* 28 left REC mixer */ - { 0xFF, 0xFF, 0 }, /* 29 right REC mixer */ - { 0xFF, 0xFF, 0 }, /* 2A REC control */ - { 0xFF, 0xFF, 0 }, /* 2B left SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2C right SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2D SPK control */ - { 0xFF, 0xFF, 0 }, /* 2E sidetone */ - { 0xFF, 0xFF, 0 }, /* 2F DAI1 playback level */ - - { 0xFF, 0xFF, 0 }, /* 30 DAI1 playback level */ - { 0xFF, 0xFF, 0 }, /* 31 DAI2 playback level */ - { 0xFF, 0xFF, 0 }, /* 32 DAI2 playbakc level */ - { 0xFF, 0xFF, 0 }, /* 33 left ADC level */ - { 0xFF, 0xFF, 0 }, /* 34 right ADC level */ - { 0xFF, 0xFF, 0 }, /* 35 MIC1 level */ - { 0xFF, 0xFF, 0 }, /* 36 MIC2 level */ - { 0xFF, 0xFF, 0 }, /* 37 INA level */ - { 0xFF, 0xFF, 0 }, /* 38 INB level */ - { 0xFF, 0xFF, 0 }, /* 39 left HP volume */ - { 0xFF, 0xFF, 0 }, /* 3A right HP volume */ - { 0xFF, 0xFF, 0 }, /* 3B left REC volume */ - { 0xFF, 0xFF, 0 }, /* 3C right REC volume */ - { 0xFF, 0xFF, 0 }, /* 3D left SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3E right SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3F MIC config */ - - { 0xFF, 0xFF, 0 }, /* 40 MIC threshold */ - { 0xFF, 0xFF, 0 }, /* 41 excursion limiter filter */ - { 0xFF, 0xFF, 0 }, /* 42 excursion limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 43 ALC */ - { 0xFF, 0xFF, 0 }, /* 44 power limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 45 power limiter config */ - { 0xFF, 0xFF, 0 }, /* 46 distortion limiter config */ - { 0xFF, 0xFF, 0 }, /* 47 audio input */ - { 0xFF, 0xFF, 0 }, /* 48 microphone */ - { 0xFF, 0xFF, 0 }, /* 49 level control */ - { 0xFF, 0xFF, 0 }, /* 4A bypass switches */ - { 0xFF, 0xFF, 0 }, /* 4B jack detect */ - { 0xFF, 0xFF, 0 }, /* 4C input enable */ - { 0xFF, 0xFF, 0 }, /* 4D output enable */ - { 0xFF, 0xFF, 0 }, /* 4E bias control */ - { 0xFF, 0xFF, 0 }, /* 4F DAC power */ - - { 0xFF, 0xFF, 0 }, /* 50 DAC power */ - { 0xFF, 0xFF, 0 }, /* 51 system */ - { 0xFF, 0xFF, 0 }, /* 52 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 53 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 54 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 55 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 56 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 57 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 58 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 59 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5A DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5B DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5C DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5D DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5E DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5F DAI1 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 60 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 61 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 62 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 63 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 64 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 65 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 66 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 67 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 68 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 69 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6A DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6B DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6C DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6D DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6E DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6F DAI1 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* 70 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 71 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 72 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 73 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 74 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 75 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 76 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 77 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 78 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 79 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 7A DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7B DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7C DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7D DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7E DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7F DAI1 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* 80 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 81 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 82 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 83 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 84 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 85 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 86 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 87 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 88 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 89 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8A DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8B DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8C DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8D DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8E DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 8F DAI2 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 90 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 91 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 92 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 93 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 94 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 95 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 96 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 97 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 98 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 99 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9A DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9B DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9C DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9D DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9E DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9F DAI2 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* A0 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A1 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A2 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A3 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A4 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A5 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A6 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A7 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A8 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A9 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AA DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AB DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AC DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AD DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AE DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AF DAI2 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* B0 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B1 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B2 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B3 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B4 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B5 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B6 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B7 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B8 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B9 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BA DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BB DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BC DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BD DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BE DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BF DAI1 biquad */ - - { 0xFF, 0xFF, 0 }, /* C0 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C1 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C2 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C3 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C4 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C5 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C6 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C7 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C8 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C9 DAI2 biquad */ - { 0x00, 0x00, 0 }, /* CA */ - { 0x00, 0x00, 0 }, /* CB */ - { 0x00, 0x00, 0 }, /* CC */ - { 0x00, 0x00, 0 }, /* CD */ - { 0x00, 0x00, 0 }, /* CE */ - { 0x00, 0x00, 0 }, /* CF */ - - { 0x00, 0x00, 0 }, /* D0 */ - { 0x00, 0x00, 0 }, /* D1 */ - { 0x00, 0x00, 0 }, /* D2 */ - { 0x00, 0x00, 0 }, /* D3 */ - { 0x00, 0x00, 0 }, /* D4 */ - { 0x00, 0x00, 0 }, /* D5 */ - { 0x00, 0x00, 0 }, /* D6 */ - { 0x00, 0x00, 0 }, /* D7 */ - { 0x00, 0x00, 0 }, /* D8 */ - { 0x00, 0x00, 0 }, /* D9 */ - { 0x00, 0x00, 0 }, /* DA */ - { 0x00, 0x00, 0 }, /* DB */ - { 0x00, 0x00, 0 }, /* DC */ - { 0x00, 0x00, 0 }, /* DD */ - { 0x00, 0x00, 0 }, /* DE */ - { 0x00, 0x00, 0 }, /* DF */ - - { 0x00, 0x00, 0 }, /* E0 */ - { 0x00, 0x00, 0 }, /* E1 */ - { 0x00, 0x00, 0 }, /* E2 */ - { 0x00, 0x00, 0 }, /* E3 */ - { 0x00, 0x00, 0 }, /* E4 */ - { 0x00, 0x00, 0 }, /* E5 */ - { 0x00, 0x00, 0 }, /* E6 */ - { 0x00, 0x00, 0 }, /* E7 */ - { 0x00, 0x00, 0 }, /* E8 */ - { 0x00, 0x00, 0 }, /* E9 */ - { 0x00, 0x00, 0 }, /* EA */ - { 0x00, 0x00, 0 }, /* EB */ - { 0x00, 0x00, 0 }, /* EC */ - { 0x00, 0x00, 0 }, /* ED */ - { 0x00, 0x00, 0 }, /* EE */ - { 0x00, 0x00, 0 }, /* EF */ - - { 0x00, 0x00, 0 }, /* F0 */ - { 0x00, 0x00, 0 }, /* F1 */ - { 0x00, 0x00, 0 }, /* F2 */ - { 0x00, 0x00, 0 }, /* F3 */ - { 0x00, 0x00, 0 }, /* F4 */ - { 0x00, 0x00, 0 }, /* F5 */ - { 0x00, 0x00, 0 }, /* F6 */ - { 0x00, 0x00, 0 }, /* F7 */ - { 0x00, 0x00, 0 }, /* F8 */ - { 0x00, 0x00, 0 }, /* F9 */ - { 0x00, 0x00, 0 }, /* FA */ - { 0x00, 0x00, 0 }, /* FB */ - { 0x00, 0x00, 0 }, /* FC */ - { 0x00, 0x00, 0 }, /* FD */ - { 0x00, 0x00, 0 }, /* FE */ - { 0xFF, 0x00, 1 }, /* FF */ -}; - static bool max98088_readable_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].readable; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... 0xC9: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } +} + +static bool max98088_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98088_REG_03_BATTERY_VOLTAGE ... 0xC9: + return true; + default: + return false; + } } static bool max98088_volatile_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].vol; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... M98088_REG_03_BATTERY_VOLTAGE: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } } static const struct regmap_config max98088_regmap = { @@ -551,6 +295,7 @@ static const struct regmap_config max98088_regmap = { .val_bits = 8, .readable_reg = max98088_readable_register, + .writeable_reg = max98088_writeable_register, .volatile_reg = max98088_volatile_register, .max_register = 0xff, diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index be89a4f..efa39bf 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -16,7 +16,7 @@ */ #define M98088_REG_00_IRQ_STATUS 0x00 #define M98088_REG_01_MIC_STATUS 0x01 -#define M98088_REG_02_JACK_STAUS 0x02 +#define M98088_REG_02_JACK_STATUS 0x02 #define M98088_REG_03_BATTERY_VOLTAGE 0x03 #define M98088_REG_0F_IRQ_ENABLE 0x0F #define M98088_REG_10_SYS_CLK 0x10 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c9db085..e09c130 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -850,6 +850,19 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, return 0; } +static int max98090_shdn_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (event & SND_SOC_DAPM_POST_PMU) + max98090->shdn_pending = true; + + return 0; + +} + static const char *mic1_mux_text[] = { "IN12", "IN56" }; static SOC_ENUM_SINGLE_DECL(mic1_mux_enum, @@ -1158,9 +1171,11 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, M98090_SDOEN_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICL_SHIFT, 0, NULL, 0), + M98090_DIGMICL_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICR_SHIFT, 0, NULL, 0), + M98090_DIGMICR_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, M98090_AHPF_SHIFT, 0, NULL, 0), @@ -1205,10 +1220,12 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_adc_mixer_controls[0], ARRAY_SIZE(max98090_right_adc_mixer_controls)), - SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADLEN_SHIFT, 0), - SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADREN_SHIFT, 0), + SND_SOC_DAPM_ADC_E("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), @@ -2383,7 +2400,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops max98090_dai_ops = { +static const struct snd_soc_dai_ops max98090_dai_ops = { .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, @@ -2536,9 +2553,26 @@ static int max98090_remove(struct snd_soc_codec *codec) return 0; } +static void max98090_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (max98090->shdn_pending) { + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, 0); + msleep(40); + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, M98090_SHDNN_MASK); + max98090->shdn_pending = false; + } +} + static struct snd_soc_codec_driver soc_codec_dev_max98090 = { .probe = max98090_probe, .remove = max98090_remove, + .seq_notifier = max98090_seq_notifier, .set_bias_level = max98090_set_bias_level, }; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 21ff743..bc610d9 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1543,6 +1543,7 @@ struct max98090_priv { unsigned int pa2en; unsigned int sidetone; bool master; + bool shdn_pending; }; int max98090_mic_detect(struct snd_soc_codec *codec, diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 3a2fda0..c4a211d 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -79,7 +79,7 @@ static struct snd_soc_codec_driver max98357a_codec_driver = { .num_dapm_routes = ARRAY_SIZE(max98357a_dapm_routes), }; -static struct snd_soc_dai_ops max98357a_dai_ops = { +static const struct snd_soc_dai_ops max98357a_dai_ops = { .trigger = max98357a_daiops_trigger, }; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 3d44fc5..3e770cb 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -650,14 +650,14 @@ static int mc13783_remove(struct snd_soc_codec *codec) #define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops mc13783_ops_dac = { +static const struct snd_soc_dai_ops mc13783_ops_dac = { .hw_params = mc13783_pcm_hw_params_dac, .set_fmt = mc13783_set_fmt_async, .set_sysclk = mc13783_set_sysclk_dac, .set_tdm_slot = mc13783_set_tdm_slot_dac, }; -static struct snd_soc_dai_ops mc13783_ops_codec = { +static const struct snd_soc_dai_ops mc13783_ops_codec = { .hw_params = mc13783_pcm_hw_params_codec, .set_fmt = mc13783_set_fmt_async, .set_sysclk = mc13783_set_sysclk_codec, @@ -698,7 +698,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { }, }; -static struct snd_soc_dai_ops mc13783_ops_sync = { +static const struct snd_soc_dai_ops mc13783_ops_sync = { .hw_params = mc13783_pcm_hw_params_sync, .set_fmt = mc13783_set_fmt_sync, .set_sysclk = mc13783_set_sysclk_sync, diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index b2c990f..5832523 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -95,17 +95,22 @@ static int pcm1681_set_deemph(struct snd_soc_codec *codec) struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); int i = 0, val = -1, enable = 0; - if (priv->deemph) - for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) - if (pcm1681_deemph[i] == priv->rate) + if (priv->deemph) { + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) { + if (pcm1681_deemph[i] == priv->rate) { val = i; + break; + } + } + } if (val != -1) { regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, - PCM1681_DEEMPH_RATE_MASK, val); + PCM1681_DEEMPH_RATE_MASK, val << 3); enable = 1; - } else + } else { enable = 0; + } /* enable/disable deemphasis functionality */ return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4a780ef..e6691a1 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -984,6 +984,35 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, RT5640_PWR_LM); + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, 0); + break; + + default: + return 0; + } + + return 0; +} + static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1179,13 +1208,16 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, @@ -1500,8 +1532,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HP R Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HP L Playback"}, {"HPOR", NULL, "HP R Playback"}, - {"LOUTL", NULL, "LOUT MIX"}, - {"LOUTR", NULL, "LOUT MIX"}, + + {"LOUT amp", NULL, "LOUT MIX"}, + {"LOUTL", NULL, "LOUT amp"}, + {"LOUTR", NULL, "LOUT amp"}, }; static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 23a7e8d..3614340 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -21,6 +21,7 @@ #include <linux/gpio/consumer.h> #include <linux/acpi.h> #include <linux/dmi.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -223,6 +224,39 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +static const char *const rt5645_supply_names[] = { + "avdd", + "cpvdd", +}; + +struct rt5645_priv { + struct snd_soc_codec *codec; + struct rt5645_platform_data pdata; + struct regmap *regmap; + struct i2c_client *i2c; + struct gpio_desc *gpiod_hp_det; + struct snd_soc_jack *hp_jack; + struct snd_soc_jack *mic_jack; + struct snd_soc_jack *btn_jack; + struct delayed_work jack_detect_work; + struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + + int codec_type; + int sysclk; + int sysclk_src; + int lrck[RT5645_AIFS]; + int bclk[RT5645_AIFS]; + int master[RT5645_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; + bool en_button_func; + bool hp_on; +}; + static int rt5645_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, RT5645_RESET, 0); @@ -360,6 +394,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DEPOP_M1: case RT5645_DEPOP_M2: case RT5645_DEPOP_M3: + case RT5645_CHARGE_PUMP: case RT5645_MICBIAS: case RT5645_A_JD_CTRL1: case RT5645_VAD_CTRL4: @@ -1331,15 +1366,23 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on) if (on) { if (hp_amp_power_count <= 0) { if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M2, 0x3100); snd_soc_write(codec, RT5645_CHARGE_PUMP, 0x0e06); - snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d); + snd_soc_write(codec, RT5645_DEPOP_M1, 0x000d); + regmap_write(rt5645->regmap, RT5645_PR_BASE + + RT5645_HP_DCC_INT1, 0x9f01); + msleep(20); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_CO_MASK, RT5645_HP_CO_EN); regmap_write(rt5645->regmap, RT5645_PR_BASE + 0x3e, 0x7400); snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140); + mdelay(5); + rt5645->hp_on = true; } else { /* depop parameters */ snd_soc_update_bits(codec, RT5645_DEPOP_M2, @@ -1553,6 +1596,27 @@ static int rt5645_bst2_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5650_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt5645->hp_on) { + msleep(100); + rt5645->hp_on = false; + } + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5645_PWR_MIXER, RT5645_PWR_LDO2_BIT, 0, NULL, 0), @@ -1697,15 +1761,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), /* IF1 2 Mux */ - SND_SOC_DAPM_MUX("RT5645 IF1 ADC1 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc1_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC2 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc2_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC3 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc3_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc_in_mux), - SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5645_if2_adc_in_mux), @@ -1716,14 +1771,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac0_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac1_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac2_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac3_tdm_sel_mux), SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1854,6 +1901,26 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("PDM1R"), SND_SOC_DAPM_OUTPUT("SPOL"), SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_POST("DAPM_POST", rt5650_hp_event), +}; + +static const struct snd_soc_dapm_widget rt5645_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC1 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc1_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC2 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC3 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc3_in_mux), }; static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { @@ -2642,7 +2709,7 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_PREPARE: - if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) { snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2, @@ -2686,94 +2753,15 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5650_calibration(struct rt5645_priv *rt5645) -{ - int val, i; - int ret = -1; - - regcache_cache_bypass(rt5645->regmap, true); - regmap_write(rt5645->regmap, RT5645_RESET, 0); - regmap_write(rt5645->regmap, RT5645_GEN_CTRL3, 0x0800); - regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_CHOP_DAC_ADC, - 0x3600); - regmap_write(rt5645->regmap, RT5645_PR_BASE + 0x25, 0x7000); - regmap_write(rt5645->regmap, RT5645_I2S1_SDP, 0x8008); - /* headset type */ - regmap_write(rt5645->regmap, RT5645_GEN_CTRL1, 0x2061); - regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0006); - regmap_write(rt5645->regmap, RT5645_PWR_ANLG1, 0x2012); - regmap_write(rt5645->regmap, RT5645_PWR_MIXER, 0x0002); - regmap_write(rt5645->regmap, RT5645_PWR_VOL, 0x0020); - regmap_write(rt5645->regmap, RT5645_JD_CTRL3, 0x00f0); - regmap_write(rt5645->regmap, RT5645_IN1_CTRL1, 0x0006); - regmap_write(rt5645->regmap, RT5645_IN1_CTRL2, 0x1827); - regmap_write(rt5645->regmap, RT5645_IN1_CTRL2, 0x0827); - msleep(400); - /* Inline command */ - regmap_write(rt5645->regmap, RT5645_DEPOP_M1, 0x0001); - regmap_write(rt5645->regmap, RT5650_4BTN_IL_CMD2, 0xc000); - regmap_write(rt5645->regmap, RT5650_4BTN_IL_CMD1, 0x0008); - /* Calbration */ - regmap_write(rt5645->regmap, RT5645_GLB_CLK, 0x8000); - regmap_write(rt5645->regmap, RT5645_DEPOP_M1, 0x0000); - regmap_write(rt5645->regmap, RT5650_4BTN_IL_CMD2, 0xc000); - regmap_write(rt5645->regmap, RT5650_4BTN_IL_CMD1, 0x0008); - regmap_write(rt5645->regmap, RT5645_PWR_DIG2, 0x8800); - regmap_write(rt5645->regmap, RT5645_PWR_ANLG1, 0xe8fa); - regmap_write(rt5645->regmap, RT5645_PWR_ANLG2, 0x8c04); - regmap_write(rt5645->regmap, RT5645_DEPOP_M2, 0x3100); - regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0e06); - regmap_write(rt5645->regmap, RT5645_BASS_BACK, 0x8a13); - regmap_write(rt5645->regmap, RT5645_GEN_CTRL3, 0x0820); - regmap_write(rt5645->regmap, RT5645_DEPOP_M1, 0x000d); - /* Power on and Calbration */ - regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_HP_DCC_INT1, - 0x9f01); - msleep(200); - for (i = 0; i < 5; i++) { - regmap_read(rt5645->regmap, RT5645_PR_BASE + 0x7a, &val); - if (val != 0 && val != 0x3f3f) { - ret = 0; - break; - } - msleep(50); - } - pr_debug("%s: PR-7A = 0x%x\n", __func__, val); - - /* mute */ - regmap_write(rt5645->regmap, RT5645_PR_BASE + 0x3e, 0x7400); - regmap_write(rt5645->regmap, RT5645_DEPOP_M3, 0x0737); - regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, - 0xfc00); - regmap_write(rt5645->regmap, RT5645_DEPOP_M2, 0x1140); - regmap_write(rt5645->regmap, RT5645_DEPOP_M1, 0x0000); - regmap_write(rt5645->regmap, RT5645_GEN_CTRL2, 0x4020); - regmap_write(rt5645->regmap, RT5645_PWR_ANLG2, 0x0006); - regmap_write(rt5645->regmap, RT5645_PWR_DIG2, 0x0000); - msleep(350); - - regcache_cache_bypass(rt5645->regmap, false); - - return ret; -} - static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, bool enable) { - struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); if (enable) { - snd_soc_dapm_mutex_lock(&codec->dapm); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "ADC L power"); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "ADC R power"); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "LDO2"); - snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync_unlocked(&codec->dapm); - snd_soc_dapm_mutex_unlock(&codec->dapm); + snd_soc_dapm_force_enable_pin(dapm, "ADC L power"); + snd_soc_dapm_force_enable_pin(dapm, "ADC R power"); + snd_soc_dapm_sync(dapm); snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x8); @@ -2786,36 +2774,26 @@ static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); - snd_soc_dapm_mutex_lock(&codec->dapm); - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "ADC L power"); - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "ADC R power"); - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "LDO2"); - snd_soc_dapm_disable_pin_unlocked(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync_unlocked(&codec->dapm); - snd_soc_dapm_mutex_unlock(&codec->dapm); + snd_soc_dapm_disable_pin(dapm, "ADC L power"); + snd_soc_dapm_disable_pin(dapm, "ADC R power"); + snd_soc_dapm_sync(dapm); } } static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); unsigned int val; if (jack_insert) { regmap_write(rt5645->regmap, RT5645_CHARGE_PUMP, 0x0006); - if (codec->component.card->instantiated) { - /* for jack type detect */ - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); - } else { + /* for jack type detect */ + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); + if (!dapm->card->instantiated) { /* Power up necessary bits for JD if dapm is not ready yet */ regmap_update_bits(rt5645->regmap, RT5645_PWR_ANLG1, @@ -2828,14 +2806,15 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) } regmap_write(rt5645->regmap, RT5645_JD_CTRL3, 0x00f0); - regmap_write(rt5645->regmap, RT5645_IN1_CTRL1, 0x0006); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x1000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); msleep(100); - regmap_update_bits(rt5645->regmap, - RT5645_IN1_CTRL2, 0x1000, 0x0000); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, 0); - msleep(450); + msleep(600); regmap_read(rt5645->regmap, RT5645_IN1_CTRL3, &val); val &= 0x7; dev_dbg(codec->dev, "val = %d\n", val); @@ -2846,43 +2825,46 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) rt5645_enable_push_button_irq(codec, true); } } else { - if (codec->component.card->instantiated) { - snd_soc_dapm_disable_pin(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); - } else - regmap_update_bits(rt5645->regmap, - RT5645_PWR_VOL, RT5645_PWR_MIC_DET, 0); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } + snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); + snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d); + snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001); } else { /* jack out */ rt5645->jack_type = 0; + + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, + RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, 0); + if (rt5645->en_button_func) rt5645_enable_push_button_irq(codec, false); - else { - if (codec->component.card->instantiated) { - if (rt5645->pdata.jd_mode == 0) - snd_soc_dapm_disable_pin(&codec->dapm, - "LDO2"); - snd_soc_dapm_disable_pin(&codec->dapm, - "Mic Det Power"); - snd_soc_dapm_sync(&codec->dapm); - } else { - if (rt5645->pdata.jd_mode == 0) - regmap_update_bits(rt5645->regmap, - RT5645_PWR_MIXER, - RT5645_PWR_LDO2, 0); - regmap_update_bits(rt5645->regmap, - RT5645_PWR_VOL, RT5645_PWR_MIC_DET, 0); - } - } + + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(dapm, "LDO2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_sync(dapm); } return rt5645->jack_type; } -static int rt5645_irq_detection(struct rt5645_priv *rt5645); +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + static irqreturn_t rt5645_irq(int irq, void *data); int rt5645_set_jack_detect(struct snd_soc_codec *codec, @@ -2913,36 +2895,11 @@ static void rt5645_jack_detect_work(struct work_struct *work) { struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - - rt5645_irq_detection(rt5645); -} - -static irqreturn_t rt5645_irq(int irq, void *data) -{ - struct rt5645_priv *rt5645 = data; - - queue_delayed_work(system_power_efficient_wq, - &rt5645->jack_detect_work, msecs_to_jiffies(250)); - - return IRQ_HANDLED; -} - -static int rt5645_button_detect(struct snd_soc_codec *codec) -{ - int btn_type, val; - - val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); - pr_debug("val=0x%x\n", val); - btn_type = val & 0xfff0; - snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); - - return btn_type; -} - -static int rt5645_irq_detection(struct rt5645_priv *rt5645) -{ int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { @@ -2955,7 +2912,7 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) report, SND_JACK_HEADPHONE); snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); - return report; + return; case 1: /* 2 port */ val = snd_soc_read(rt5645->codec, RT5645_A_JD_CTRL1) & 0x0070; break; @@ -3037,27 +2994,39 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) snd_soc_jack_report(rt5645->btn_jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); +} + +static irqreturn_t rt5645_irq(int irq, void *data) +{ + struct rt5645_priv *rt5645 = data; + + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(250)); - return report; + return IRQ_HANDLED; } static int rt5645_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); rt5645->codec = codec; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - snd_soc_dapm_add_routes(&codec->dapm, + snd_soc_dapm_new_controls(dapm, + rt5645_specific_dapm_widgets, + ARRAY_SIZE(rt5645_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); break; case CODEC_TYPE_RT5650: - snd_soc_dapm_new_controls(&codec->dapm, + snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - snd_soc_dapm_add_routes(&codec->dapm, + snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); break; @@ -3067,9 +3036,9 @@ static int rt5645_probe(struct snd_soc_codec *codec) /* for JD function */ if (rt5645->pdata.jd_mode) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + snd_soc_dapm_sync(dapm); } return 0; @@ -3110,7 +3079,7 @@ static int rt5645_resume(struct snd_soc_codec *codec) #define RT5645_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5645_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5645_aif_dai_ops = { .hw_params = rt5645_hw_params, .set_fmt = rt5645_set_dai_fmt, .set_sysclk = rt5645_set_dai_sysclk, @@ -3221,7 +3190,7 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } -static struct dmi_system_id dmi_platform_intel_braswell[] = { +static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", .callback = strago_quirk_cb, @@ -3229,6 +3198,13 @@ static struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Strago"), }, }, + { + .ident = "Google Celes", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), + }, + }, { } }; @@ -3251,7 +3227,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, { struct rt5645_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt5645_priv *rt5645; - int ret; + int ret, i; unsigned int val; rt5645 = devm_kzalloc(&i2c->dev, sizeof(struct rt5645_priv), @@ -3285,6 +3261,24 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, return ret; } + for (i = 0; i < ARRAY_SIZE(rt5645->supplies); i++) + rt5645->supplies[i].supply = rt5645_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(rt5645->supplies), + rt5645->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5645->supplies), + rt5645->supplies); + if (ret) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); switch (val) { @@ -3296,16 +3290,10 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, break; default: dev_err(&i2c->dev, - "Device with ID register %x is not rt5645 or rt5650\n", + "Device with ID register %#x is not rt5645 or rt5650\n", val); - return -ENODEV; - } - - if (rt5645->codec_type == CODEC_TYPE_RT5650) { - ret = rt5650_calibration(rt5645); - - if (ret < 0) - pr_err("calibration failed!\n"); + ret = -ENODEV; + goto err_enable; } regmap_write(rt5645->regmap, RT5645_RESET, 0); @@ -3338,6 +3326,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, break; case RT5645_DMIC_DATA_GPIO5: + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_I2S2_DAC_PIN_MASK, RT5645_I2S2_DAC_PIN_GPIO); regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, RT5645_DMIC_1_DP_MASK, RT5645_DMIC_1_DP_GPIO5); regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, @@ -3393,8 +3383,6 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, RT5645_IRQ_CLK_GATE_CTRL, RT5645_IRQ_CLK_GATE_CTRL); - regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, - RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT); regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, @@ -3434,12 +3422,25 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5645", rt5645); - if (ret) + if (ret) { dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + goto err_enable; + } } - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, - rt5645_dai, ARRAY_SIZE(rt5645_dai)); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, + rt5645_dai, ARRAY_SIZE(rt5645_dai)); + if (ret) + goto err_irq; + + return 0; + +err_irq: + if (rt5645->i2c->irq) + free_irq(rt5645->i2c->irq, rt5645); +err_enable: + regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); + return ret; } static int rt5645_i2c_remove(struct i2c_client *i2c) @@ -3452,17 +3453,31 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); snd_soc_unregister_codec(&i2c->dev); + regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); return 0; } +static void rt5645_i2c_shutdown(struct i2c_client *i2c) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(i2c); + + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_RING2_SLEEVE_GND, RT5645_RING2_SLEEVE_GND); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2, RT5645_CBJ_MN_JD, + RT5645_CBJ_MN_JD); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, RT5645_CBJ_BST1_EN, + 0); +} + static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, - .remove = rt5645_i2c_remove, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, .id_table = rt5645_i2c_id, }; module_i2c_driver(rt5645_i2c_driver); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 0353a6a..0e4cfc6 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1693,6 +1693,10 @@ #define RT5645_GP6_PIN_SFT 6 #define RT5645_GP6_PIN_GPIO6 (0x0 << 6) #define RT5645_GP6_PIN_DMIC2_SDA (0x1 << 6) +#define RT5645_I2S2_DAC_PIN_MASK (0x1 << 4) +#define RT5645_I2S2_DAC_PIN_SFT 4 +#define RT5645_I2S2_DAC_PIN_I2S (0x0 << 4) +#define RT5645_I2S2_DAC_PIN_GPIO (0x1 << 4) #define RT5645_GP8_PIN_MASK (0x1 << 3) #define RT5645_GP8_PIN_SFT 3 #define RT5645_GP8_PIN_GPIO8 (0x0 << 3) @@ -2111,6 +2115,7 @@ enum { #define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) +#define RT5645_RING2_SLEEVE_GND (0x1 << 5) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 @@ -2177,32 +2182,6 @@ enum { int rt5645_sel_asrc_clk_src(struct snd_soc_codec *codec, unsigned int filter_mask, unsigned int clk_src); -struct rt5645_priv { - struct snd_soc_codec *codec; - struct rt5645_platform_data pdata; - struct regmap *regmap; - struct i2c_client *i2c; - struct gpio_desc *gpiod_hp_det; - struct snd_soc_jack *hp_jack; - struct snd_soc_jack *mic_jack; - struct snd_soc_jack *btn_jack; - struct delayed_work jack_detect_work; - - int codec_type; - int sysclk; - int sysclk_src; - int lrck[RT5645_AIFS]; - int bclk[RT5645_AIFS]; - int master[RT5645_AIFS]; - - int pll_src; - int pll_in; - int pll_out; - - int jack_type; - bool en_button_func; -}; - int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, struct snd_soc_jack *btn_jack); diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8f9ab2b..d5bf49d 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2720,7 +2720,7 @@ static int rt5670_resume(struct snd_soc_codec *codec) #define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5670_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .hw_params = rt5670_hw_params, .set_fmt = rt5670_set_dai_fmt, .set_sysclk = rt5670_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 03afec7..2313fbf 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4863,7 +4863,7 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) #define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5677_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5677_aif_dai_ops = { .hw_params = rt5677_hw_params, .set_fmt = rt5677_set_dai_fmt, .set_sysclk = rt5677_set_dai_sysclk, diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index bd7a344..1c317de 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -275,7 +275,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 0 +#define SGTL5000_SMALL_POP 1 /* * SGTL5000_CHIP_MIC_CTRL diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 3e72964..a8402d0 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -208,7 +208,7 @@ out: return err; } -static struct snd_soc_dai_ops si476x_dai_ops = { +static const struct snd_soc_dai_ops si476x_dai_ops = { .hw_params = si476x_codec_hw_params, .set_fmt = si476x_codec_set_dai_fmt, }; diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index f3f1f68..e619d56 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -10,6 +10,7 @@ * Licensed under the GPL-2. */ +#include <linux/acpi.h> #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> @@ -173,6 +174,12 @@ static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1, &ssm4567_amplifier_boost_control), + SND_SOC_DAPM_SIGGEN("Sense"), + + SND_SOC_DAPM_PGA("Current Sense", SSM4567_REG_POWER_CTRL, 4, 1, NULL, 0), + SND_SOC_DAPM_PGA("Voltage Sense", SSM4567_REG_POWER_CTRL, 5, 1, NULL, 0), + SND_SOC_DAPM_PGA("VBAT Sense", SSM4567_REG_POWER_CTRL, 6, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("OUT"), }; @@ -180,6 +187,13 @@ static const struct snd_soc_dapm_route ssm4567_routes[] = { { "OUT", NULL, "Amplifier Boost" }, { "Amplifier Boost", "Switch", "DAC" }, { "OUT", NULL, "DAC" }, + + { "Current Sense", NULL, "Sense" }, + { "Voltage Sense", NULL, "Sense" }, + { "VBAT Sense", NULL, "Sense" }, + { "Capture Sense", NULL, "Current Sense" }, + { "Capture Sense", NULL, "Voltage Sense" }, + { "Capture Sense", NULL, "VBAT Sense" }, }; static int ssm4567_hw_params(struct snd_pcm_substream *substream, @@ -315,7 +329,13 @@ static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (invert_fclk) ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; - return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1); + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, + SSM4567_SAI_CTRL_1_BCLK | + SSM4567_SAI_CTRL_1_FSYNC | + SSM4567_SAI_CTRL_1_LJ | + SSM4567_SAI_CTRL_1_TDM | + SSM4567_SAI_CTRL_1_PDM, + ctrl1); } static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) @@ -381,6 +401,14 @@ static struct snd_soc_dai_driver ssm4567_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32, }, + .capture = { + .stream_name = "Capture Sense", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32, + }, .ops = &ssm4567_dai_ops, }; @@ -450,9 +478,20 @@ static const struct i2c_device_id ssm4567_i2c_ids[] = { }; MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); +#ifdef CONFIG_ACPI + +static const struct acpi_device_id ssm4567_acpi_match[] = { + { "INT343B", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, ssm4567_acpi_match); + +#endif + static struct i2c_driver ssm4567_driver = { .driver = { .name = "ssm4567", + .acpi_match_table = ACPI_PTR(ssm4567_acpi_match), }, .probe = ssm4567_i2c_probe, .remove = ssm4567_i2c_remove, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index ed4cca7..0945c51 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -28,6 +28,9 @@ #include "stac9766.h" +#define STAC9766_VENDOR_ID 0x83847666 +#define STAC9766_VENDOR_ID_MASK 0xffffffff + /* * STAC9766 register cache */ @@ -239,45 +242,12 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) -{ - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(ac97); - if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) - return 1; - } - - soc_ac97_ops->reset(ac97); - if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(ac97); - if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) - return -EIO; - return 0; -} - static int stac9766_codec_resume(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 id, reset; - reset = 0; - /* give the codec an AC97 warm reset to start the link */ -reset: - if (reset > 5) { - dev_err(codec->dev, "Failed to resume\n"); - return -EIO; - } - ac97->bus->ops->warm_reset(ac97); - id = soc_ac97_ops->read(ac97, AC97_VENDOR_ID2); - if (id != 0x4c13) { - stac9766_reset(codec, 0); - reset++; - goto reset; - } - - return 0; + return snd_ac97_reset(ac97, true, STAC9766_VENDOR_ID, + STAC9766_VENDOR_ID_MASK); } static const struct snd_soc_dai_ops stac9766_dai_ops_analog = { @@ -330,28 +300,15 @@ static struct snd_soc_dai_driver stac9766_dai[] = { static int stac9766_codec_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; - int ret = 0; - ac97 = snd_soc_new_ac97_codec(codec); + ac97 = snd_soc_new_ac97_codec(codec, STAC9766_VENDOR_ID, + STAC9766_VENDOR_ID_MASK); if (IS_ERR(ac97)) return PTR_ERR(ac97); snd_soc_codec_set_drvdata(codec, ac97); - /* do a cold reset for the controller and then try - * a warm reset followed by an optional cold reset for codec */ - stac9766_reset(codec, 0); - ret = stac9766_reset(codec, 1); - if (ret < 0) { - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - goto codec_err; - } - return 0; - -codec_err: - snd_soc_free_ac97_codec(ac97); - return ret; } static int stac9766_codec_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 083b6b3..5e0a8a5 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -520,7 +520,7 @@ static const struct dev_pm_ops tas2552_pm = { NULL) }; -static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { +static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .hw_params = tas2552_hw_params, .prepare = tas2552_prepare, .set_sysclk = tas2552_set_dai_sysclk, diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 85bcc37..39307ad 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -179,7 +179,7 @@ static int tas571x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { if (!IS_ERR(priv->mclk)) { ret = clk_prepare_enable(priv->mclk); if (ret) { diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 48dd9b2..ee4def4 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1121,7 +1121,7 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), }; -static struct snd_soc_dai_ops aic31xx_dai_ops = { +static const struct snd_soc_dai_ops aic31xx_dai_ops = { .hw_params = aic31xx_hw_params, .set_sysclk = aic31xx_set_dai_sysclk, .set_fmt = aic31xx_set_dai_fmt, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index d097f09e5..64637d1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -788,8 +788,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -801,8 +800,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -814,8 +812,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -827,8 +824,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, @@ -851,10 +847,10 @@ ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), -SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), -SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), +ARIZONA_LHPF_CONTROL("LHPF1 Coefficients", ARIZONA_HPLPF1_2), +ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), +ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), +ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), ARIZONA_MIXER_CONTROLS("DSP1L", ARIZONA_DSP1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP1R", ARIZONA_DSP1RMIX_INPUT_1_SOURCE), @@ -1883,7 +1879,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) ret = snd_soc_add_codec_controls(codec, arizona_adsp2_rate_controls, 1); if (ret) - return ret; + goto err_adsp2_codec_probe; arizona_init_spk(codec); arizona_init_gpio(codec); @@ -1893,6 +1889,11 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) priv->core.arizona->dapm = dapm; return 0; + +err_adsp2_codec_probe: + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + + return ret; } static int wm5102_codec_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 709fcc6..2d1168c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,8 +247,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -260,8 +259,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -273,8 +271,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -286,8 +283,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, @@ -314,10 +310,10 @@ ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), -SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), -SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), +ARIZONA_LHPF_CONTROL("LHPF1 Coefficients", ARIZONA_HPLPF1_2), +ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), +ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), +ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), @@ -1611,18 +1607,24 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) for (i = 0; i < WM5110_NUM_ADSP; ++i) { ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); if (ret) - return ret; + goto err_adsp2_codec_probe; } ret = snd_soc_add_codec_controls(codec, arizona_adsp2_rate_controls, WM5110_NUM_ADSP); if (ret) - return ret; + goto err_adsp2_codec_probe; snd_soc_dapm_disable_pin(dapm, "HAPTICS"); return 0; + +err_adsp2_codec_probe: + for (--i; i >= 0; --i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); + + return ret; } static int wm5110_codec_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 3cff5a6..b098a83 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -598,6 +598,7 @@ static const struct of_device_id wm8510_of_match[] = { { .compatible = "wlf,wm8510" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8510_of_match); static const struct regmap_config wm8510_regmap = { .reg_bits = 7, diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5f8fde5..aa287a3 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -430,6 +430,7 @@ static const struct of_device_id wm8523_of_match[] = { { .compatible = "wlf,wm8523" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8523_of_match); static const struct regmap_config wm8523_regmap = { .reg_bits = 8, diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index abf6035..66602bf 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -916,6 +916,7 @@ static const struct of_device_id wm8580_of_match[] = { { .compatible = "wlf,wm8580" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8580_of_match); static const struct regmap_config wm8580_regmap = { .reg_bits = 7, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 962e1d3..2ccbb32 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1942,14 +1942,16 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1R" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2R" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCL" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCR" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1R" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2R" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACL" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACR" }, + + { "AIF3ADCDAT", NULL, "AIF3ADC Mux" }, /* Loopback */ { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4134dc7..b4dba3a 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -174,8 +174,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -187,8 +186,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -200,8 +198,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -213,8 +210,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, @@ -242,10 +238,10 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), -SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), -SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), -SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), -SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), +ARIZONA_LHPF_CONTROL("LHPF1 Coefficients", ARIZONA_HPLPF1_2), +ARIZONA_LHPF_CONTROL("LHPF2 Coefficients", ARIZONA_HPLPF2_2), +ARIZONA_LHPF_CONTROL("LHPF3 Coefficients", ARIZONA_HPLPF3_2), +ARIZONA_LHPF_CONTROL("LHPF4 Coefficients", ARIZONA_HPLPF4_2), SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 5cc457e..744842c 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -22,6 +22,9 @@ #include "wm9705.h" +#define WM9705_VENDOR_ID 0x574d4c05 +#define WM9705_VENDOR_ID_MASK 0xffffffff + /* * WM9705 register cache */ @@ -293,21 +296,6 @@ static struct snd_soc_dai_driver wm9705_dai[] = { } }; -static int wm9705_reset(struct snd_soc_codec *codec) -{ - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - if (soc_ac97_ops->reset) { - soc_ac97_ops->reset(ac97); - if (ac97_read(codec, 0) == wm9705_reg[0]) - return 0; /* Success */ - } - - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - - return -EIO; -} - #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { @@ -324,7 +312,8 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) int i, ret; u16 *cache = codec->reg_cache; - ret = wm9705_reset(codec); + ret = snd_ac97_reset(ac97, true, WM9705_VENDOR_ID, + WM9705_VENDOR_ID_MASK); if (ret < 0) return ret; @@ -342,30 +331,17 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) static int wm9705_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; - int ret = 0; - ac97 = snd_soc_alloc_ac97_codec(codec); + ac97 = snd_soc_new_ac97_codec(codec, WM9705_VENDOR_ID, + WM9705_VENDOR_ID_MASK); if (IS_ERR(ac97)) { - ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec\n"); - return ret; + return PTR_ERR(ac97); } - ret = wm9705_reset(codec); - if (ret) - goto err_put_device; - - ret = device_add(&ac97->dev); - if (ret) - goto err_put_device; - snd_soc_codec_set_drvdata(codec, ac97); return 0; - -err_put_device: - put_device(&ac97->dev); - return ret; } static int wm9705_soc_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fda104..488a922 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -23,6 +23,9 @@ #include <sound/tlv.h> #include "wm9712.h" +#define WM9712_VENDOR_ID 0x574d4c12 +#define WM9712_VENDOR_ID_MASK 0xffffffff + struct wm9712_priv { struct snd_ac97 *ac97; unsigned int hp_mixer[2]; @@ -613,35 +616,14 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) -{ - struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(wm9712->ac97); - if (ac97_read(codec, 0) == wm9712_reg[0]) - return 1; - } - - soc_ac97_ops->reset(wm9712->ac97); - if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(wm9712->ac97); - if (ac97_read(codec, 0) != wm9712_reg[0]) - goto err; - return 0; - -err: - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - return -EIO; -} - static int wm9712_soc_resume(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int i, ret; u16 *cache = codec->reg_cache; - ret = wm9712_reset(codec, 1); + ret = snd_ac97_reset(wm9712->ac97, true, WM9712_VENDOR_ID, + WM9712_VENDOR_ID_MASK); if (ret < 0) return ret; @@ -663,31 +645,20 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) static int wm9712_soc_probe(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); - int ret = 0; + int ret; - wm9712->ac97 = snd_soc_alloc_ac97_codec(codec); + wm9712->ac97 = snd_soc_new_ac97_codec(codec, WM9712_VENDOR_ID, + WM9712_VENDOR_ID_MASK); if (IS_ERR(wm9712->ac97)) { ret = PTR_ERR(wm9712->ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); return ret; } - ret = wm9712_reset(codec, 0); - if (ret < 0) - goto err_put_device; - - ret = device_add(&wm9712->ac97->dev); - if (ret) - goto err_put_device; - /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); return 0; - -err_put_device: - put_device(&wm9712->ac97->dev); - return ret; } static int wm9712_soc_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 89cd2d6..955e651 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -29,6 +29,9 @@ #include "wm9713.h" +#define WM9713_VENDOR_ID 0x574d4c13 +#define WM9713_VENDOR_ID_MASK 0xffffffff + struct wm9713_priv { struct snd_ac97 *ac97; u32 pll_in; /* PLL input frequency */ @@ -1123,28 +1126,6 @@ static struct snd_soc_dai_driver wm9713_dai[] = { }, }; -int wm9713_reset(struct snd_soc_codec *codec, int try_warm) -{ - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(wm9713->ac97); - if (ac97_read(codec, 0) == wm9713_reg[0]) - return 1; - } - - soc_ac97_ops->reset(wm9713->ac97); - if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(wm9713->ac97); - if (ac97_read(codec, 0) != wm9713_reg[0]) { - dev_err(codec->dev, "Failed to reset: AC97 link error\n"); - return -EIO; - } - - return 0; -} -EXPORT_SYMBOL_GPL(wm9713_reset); - static int wm9713_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1196,7 +1177,8 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) int i, ret; u16 *cache = codec->reg_cache; - ret = wm9713_reset(codec, 1); + ret = snd_ac97_reset(wm9713->ac97, true, WM9713_VENDOR_ID, + WM9713_VENDOR_ID_MASK); if (ret < 0) return ret; @@ -1222,32 +1204,18 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int ret = 0, reg; + int reg; - wm9713->ac97 = snd_soc_alloc_ac97_codec(codec); + wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, + WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); - /* do a cold reset for the controller and then try - * a warm reset followed by an optional cold reset for codec */ - wm9713_reset(codec, 0); - ret = wm9713_reset(codec, 1); - if (ret < 0) - goto err_put_device; - - ret = device_add(&wm9713->ac97->dev); - if (ret) - goto err_put_device; - /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); return 0; - -err_put_device: - put_device(&wm9713->ac97->dev); - return ret; } static int wm9713_soc_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h index 793da86..53df11b 100644 --- a/sound/soc/codecs/wm9713.h +++ b/sound/soc/codecs/wm9713.h @@ -45,6 +45,4 @@ #define WM9713_DAI_AC97_AUX 1 #define WM9713_DAI_PCM_VOICE 2 -int wm9713_reset(struct snd_soc_codec *codec, int try_warm); - #endif diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 56cb4d9..ec98548 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -651,23 +651,15 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea, *res; + struct resource *mem, *res; + void __iomem *io_base; int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "no mem resource?\n"); - return -ENODEV; - } - - ioarea = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), - pdev->name); - if (!ioarea) { - dev_err(&pdev->dev, "McBSP region already claimed\n"); - return -EBUSY; - } + io_base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(io_base)) + return PTR_ERR(io_base); dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); @@ -679,12 +671,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; clk_enable(dev->clk); - dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!dev->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_release_clk; - } + dev->base = io_base; dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b960e62..add6bb9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1613,7 +1613,7 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; - struct resource *mem, *ioarea, *res, *dat; + struct resource *mem, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; @@ -1648,22 +1648,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } } - ioarea = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), pdev->name); - if (!ioarea) { - dev_err(&pdev->dev, "Audio region already claimed\n"); - return -EBUSY; - } + mcasp->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(mcasp->base)) + return PTR_ERR(mcasp->base); pm_runtime_enable(&pdev->dev); - mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!mcasp->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err; - } - mcasp->op_mode = pdata->op_mode; /* sanity check for tdm slots parameter */ if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) { diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index fabd05f..c77d921 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -231,8 +231,9 @@ static int davinci_vcif_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, davinci_vcif_dev); - ret = snd_soc_register_component(&pdev->dev, &davinci_vcif_component, - &davinci_vcif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &davinci_vcif_component, + &davinci_vcif_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "could not register dai\n"); return ret; @@ -241,23 +242,14 @@ static int davinci_vcif_probe(struct platform_device *pdev) ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - snd_soc_unregister_component(&pdev->dev); return ret; } return 0; } -static int davinci_vcif_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; -} - static struct platform_driver davinci_vcif_driver = { .probe = davinci_vcif_probe, - .remove = davinci_vcif_remove, .driver = { .name = "davinci-vcif", }, diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e1aa3834..883087f 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -182,7 +182,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else { if (np) { - /* The eukrea,asoc-tlv320 driver was explicitely + /* The eukrea,asoc-tlv320 driver was explicitly * requested (through the device tree). */ dev_err(&pdev->dev, diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..5aeb6ed 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" +#include "../codecs/wm8960.h" #define RX 0 #define TX 1 @@ -407,6 +408,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; struct clk *codec_clk; + const char *codec_dai_name; u32 width; int ret; @@ -459,6 +461,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + codec_dai_name = "cs42888"; priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; @@ -467,14 +470,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + codec_dai_name = "wm8962"; priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { + codec_dai_name = "wm8960-hifi"; + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; @@ -521,7 +532,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].codec_dai_name = codec_dai_name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +541,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; @@ -578,6 +589,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, + { .compatible = "fsl,imx-audio-wm8960", }, {} }; diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index c068494..9f087d4 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -931,14 +931,29 @@ static int fsl_asrc_probe(struct platform_device *pdev) static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); - int i; + int i, ret; - clk_prepare_enable(asrc_priv->mem_clk); - clk_prepare_enable(asrc_priv->ipg_clk); - for (i = 0; i < ASRC_CLK_MAX_NUM; i++) - clk_prepare_enable(asrc_priv->asrck_clk[i]); + ret = clk_prepare_enable(asrc_priv->mem_clk); + if (ret) + return ret; + ret = clk_prepare_enable(asrc_priv->ipg_clk); + if (ret) + goto disable_mem_clk; + for (i = 0; i < ASRC_CLK_MAX_NUM; i++) { + ret = clk_prepare_enable(asrc_priv->asrck_clk[i]); + if (ret) + goto disable_asrck_clk; + } return 0; + +disable_asrck_clk: + for (i--; i >= 0; i--) + clk_disable_unprepare(asrc_priv->asrck_clk[i]); + clk_disable_unprepare(asrc_priv->ipg_clk); +disable_mem_clk: + clk_disable_unprepare(asrc_priv->mem_clk); + return ret; } static int fsl_asrc_runtime_suspend(struct device *dev) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 5c75971..8c2ddc1 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -839,7 +839,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); + ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); if (ret) dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..a18fd92 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -791,7 +791,7 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; if (sai->sai_on_imx) - return imx_pcm_dma_init(pdev); + return imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE); else return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..b95fbc3 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ @@ -45,7 +46,7 @@ #define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) -/* SAI Transmit/Recieve Control Register */ +/* SAI Transmit/Receive Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) #define FSL_SAI_CSR_SR BIT(24) @@ -67,10 +68,10 @@ #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit and Recieve Configuration 1 Register */ +/* SAI Transmit and Receive Configuration 1 Register */ #define FSL_SAI_CR1_RFW_MASK 0x1f -/* SAI Transmit and Recieve Configuration 2 Register */ +/* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -82,12 +83,12 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) #define FSL_SAI_CR2_DIV_MASK 0xff -/* SAI Transmit and Recieve Configuration 3 Register */ +/* SAI Transmit and Receive Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f -/* SAI Transmit and Recieve Configuration 4 Register */ +/* SAI Transmit and Receive Configuration 4 Register */ #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -97,7 +98,7 @@ #define FSL_SAI_CR4_FSP BIT(1) #define FSL_SAI_CR4_FSD_MSTR BIT(0) -/* SAI Transmit and Recieve Configuration 5 Register */ +/* SAI Transmit and Receive Configuration 5 Register */ #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8e93221..ab729f2 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -454,7 +454,8 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct platform_device *pdev = spdif_priv->pdev; struct regmap *regmap = spdif_priv->regmap; - u32 scr, mask, i; + u32 scr, mask; + int i; int ret; /* Reset module and interrupts only for first initialization */ @@ -482,13 +483,18 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | SCR_TXFIFO_FSEL_MASK; - for (i = 0; i < SPDIF_TXRATE_MAX; i++) - clk_prepare_enable(spdif_priv->txclk[i]); + for (i = 0; i < SPDIF_TXRATE_MAX; i++) { + ret = clk_prepare_enable(spdif_priv->txclk[i]); + if (ret) + goto disable_txclk; + } } else { scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC; mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; - clk_prepare_enable(spdif_priv->rxclk); + ret = clk_prepare_enable(spdif_priv->rxclk); + if (ret) + goto err; } regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); @@ -497,6 +503,9 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, return 0; +disable_txclk: + for (i--; i >= 0; i--) + clk_disable_unprepare(spdif_priv->txclk[i]); err: clk_disable_unprepare(spdif_priv->coreclk); @@ -707,7 +716,7 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, return ret; } -/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */ +/* Q-subcode information. The byte size is SPDIF_UBITS_SIZE/8 */ static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -739,7 +748,7 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, return ret; } -/* Valid bit infomation */ +/* Valid bit information */ static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -767,7 +776,7 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, return 0; } -/* DPLL lock infomation */ +/* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1255,7 +1264,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); + ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); if (ret) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..8ec6fb2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -156,7 +156,7 @@ struct fsl_ssi_soc_data { * * @dbg_stats: Debugging statistics * - * @soc: SoC specifc data + * @soc: SoC specific data */ struct fsl_ssi_private { struct regmap *regs; @@ -633,7 +633,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, sub *= 100000; do_div(sub, freq); - if (sub < savesub) { + if (sub < savesub && !(i == 0 && psr == 0 && div2 == 0)) { baudrate = tmprate; savesub = sub; pm = i; @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; } @@ -1101,6 +1103,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -1127,10 +1130,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1151,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1163,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1182,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, ®_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; } @@ -1210,7 +1232,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, } } - /* For those SLAVE implementations, we ingore non-baudclk cases + /* For those SLAVE implementations, we ignore non-baudclk cases * and, instead, abandon MASTER mode that needs baud clock. */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); @@ -1257,7 +1279,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, if (ret) goto error_pcm; } else { - ret = imx_pcm_dma_init(pdev); + ret = imx_pcm_dma_init(pdev, IMX_SSI_DMABUF_SIZE); if (ret) goto error_pcm; } @@ -1320,7 +1342,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1357,7 +1383,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } @@ -1434,6 +1462,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card: @@ -1458,6 +1507,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; } diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 0db94f49..1fc01ed 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -40,7 +40,7 @@ static const struct snd_pcm_hardware imx_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 65535, /* Limited by SDMA engine */ .periods_min = 2, @@ -52,13 +52,30 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { .pcm_hardware = &imx_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, .compat_filter_fn = filter, - .prealloc_buffer_size = IMX_SSI_DMABUF_SIZE, + .prealloc_buffer_size = IMX_DEFAULT_DMABUF_SIZE, }; -int imx_pcm_dma_init(struct platform_device *pdev) +int imx_pcm_dma_init(struct platform_device *pdev, size_t size) { + struct snd_dmaengine_pcm_config *config; + struct snd_pcm_hardware *pcm_hardware; + + config = devm_kzalloc(&pdev->dev, + sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL); + *config = imx_dmaengine_pcm_config; + if (size) + config->prealloc_buffer_size = size; + + pcm_hardware = devm_kzalloc(&pdev->dev, + sizeof(struct snd_pcm_hardware), GFP_KERNEL); + *pcm_hardware = imx_pcm_hardware; + if (size) + pcm_hardware->buffer_bytes_max = size; + + config->pcm_hardware = pcm_hardware; + return devm_snd_dmaengine_pcm_register(&pdev->dev, - &imx_dmaengine_pcm_config, + config, SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index c79cb27..133c4470a 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -20,6 +20,11 @@ */ #define IMX_SSI_DMABUF_SIZE (64 * 1024) +#define IMX_DEFAULT_DMABUF_SIZE (64 * 1024) +#define IMX_SAI_DMABUF_SIZE (64 * 1024) +#define IMX_SPDIF_DMABUF_SIZE (64 * 1024) +#define IMX_ESAI_DMABUF_SIZE (256 * 1024) + static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, int dma, enum sdma_peripheral_type peripheral_type) @@ -39,9 +44,9 @@ struct imx_pcm_fiq_params { }; #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) -int imx_pcm_dma_init(struct platform_device *pdev); +int imx_pcm_dma_init(struct platform_device *pdev, size_t size); #else -static inline int imx_pcm_dma_init(struct platform_device *pdev) +static inline int imx_pcm_dma_init(struct platform_device *pdev, size_t size) { return -ENODEV; } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 461ce27..48b2d24 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -603,7 +603,7 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx; ssi->fiq_init = imx_pcm_fiq_init(pdev, &ssi->fiq_params); - ssi->dma_init = imx_pcm_dma_init(pdev); + ssi->dma_init = imx_pcm_dma_init(pdev, IMX_SSI_DMABUF_SIZE); if (ssi->fiq_init && ssi->dma_init) { ret = ssi->fiq_init; diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f3060a4..05fde5e6e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -26,14 +26,9 @@ config SND_SST_IPC_ACPI depends on ACPI config SND_SOC_INTEL_SST - tristate "ASoC support for Intel(R) Smart Sound Technology" + tristate select SND_SOC_INTEL_SST_ACPI if ACPI depends on (X86 || COMPILE_TEST) - depends on DW_DMAC_CORE - help - This adds support for Intel(R) Smart Sound Technology (SST). - Say Y if you have such a device - If unsure select "N". config SND_SOC_INTEL_SST_ACPI tristate @@ -46,8 +41,9 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \ - I2C_DESIGNWARE_PLATFORM + depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM + depends on DW_DMAC_CORE + select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -58,7 +54,9 @@ config SND_SOC_INTEL_HASWELL_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C + depends on DW_DMAC_CORE + select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help @@ -67,7 +65,9 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C + depends on DW_DMAC_CORE + select SND_SOC_INTEL_SST select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help @@ -76,8 +76,10 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \ + depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM + depends on DW_DMAC_CORE + select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help @@ -132,3 +134,8 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with MAX98090 audio codec it also can support TI jack chip as aux device. If unsure select "N". + +config SND_SOC_INTEL_SKYLAKE + tristate + select SND_HDA_EXT_CORE + select SND_SOC_INTEL_SST diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 3853ec2..2b45435 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -5,6 +5,7 @@ obj-$(CONFIG_SND_SOC_INTEL_SST) += common/ obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/ +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += skylake/ # Machine support -obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ +obj-$(CONFIG_SND_SOC) += boards/ diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 31e9b9e..d55388e 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -132,7 +132,7 @@ static int sst_send_slot_map(struct sst_data *drv) sizeof(cmd.header) + cmd.header.length); } -int sst_slot_enum_info(struct snd_kcontrol *kcontrol, +static int sst_slot_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct sst_enum *e = (struct sst_enum *)kcontrol->private_value; @@ -1298,7 +1298,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dev_dbg(dai->dev, "Stream name=%s\n", dai->playback_widget->name); w = dai->playback_widget; - list_for_each_entry(p, &w->sinks, list_source) { + snd_soc_dapm_widget_for_each_sink_path(w, p) { if (p->connected && !p->connected(w, p->sink)) continue; @@ -1317,7 +1317,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dev_dbg(dai->dev, "Stream name=%s\n", dai->capture_widget->name); w = dai->capture_widget; - list_for_each_entry(p, &w->sources, list_sink) { + snd_soc_dapm_widget_for_each_source_path(w, p) { if (p->connected && !p->connected(w, p->sink)) continue; diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 641ebe6..683e501 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -33,7 +33,6 @@ struct sst_device *sst; static DEFINE_MUTEX(sst_lock); -extern struct snd_compr_ops sst_platform_compr_ops; int sst_register_dsp(struct sst_device *dev) { diff --git a/sound/soc/intel/atom/sst-mfld-platform.h b/sound/soc/intel/atom/sst-mfld-platform.h index 2409b23..cb32cc7 100644 --- a/sound/soc/intel/atom/sst-mfld-platform.h +++ b/sound/soc/intel/atom/sst-mfld-platform.h @@ -25,6 +25,7 @@ #include "sst-atom-controls.h" extern struct sst_device *sst; +extern struct snd_compr_ops sst_platform_compr_ops; #define SST_MONO 1 #define SST_STEREO 2 diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d..ce689c5 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -42,6 +42,11 @@ #define MIN_FRAGMENT_SIZE (50 * 1024) #define MAX_FRAGMENT_SIZE (1024 * 1024) #define SST_GET_BYTES_PER_SAMPLE(pcm_wd_sz) (((pcm_wd_sz + 15) >> 4) << 1) +#ifdef CONFIG_PM +#define GET_USAGE_COUNT(dev) (atomic_read(&dev->power.usage_count)) +#else +#define GET_USAGE_COUNT(dev) 1 +#endif int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id) { @@ -141,17 +146,12 @@ static int sst_power_control(struct device *dev, bool state) int ret = 0; int usage_count = 0; -#ifdef CONFIG_PM - usage_count = atomic_read(&dev->power.usage_count); -#else - usage_count = 1; -#endif - if (state == true) { ret = pm_runtime_get_sync(dev); - + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { + pm_runtime_put_sync(dev); dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } @@ -164,6 +164,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } @@ -204,8 +205,10 @@ static int sst_cdev_open(struct device *dev, struct intel_sst_drv *ctx = dev_get_drvdata(dev); retval = pm_runtime_get_sync(ctx->dev); - if (retval < 0) + if (retval < 0) { + pm_runtime_put_sync(ctx->dev); return retval; + } str_id = sst_get_stream(ctx, str_params); if (str_id > 0) { @@ -672,8 +675,10 @@ static int sst_send_byte_stream(struct device *dev, if (NULL == bytes) return -EINVAL; ret_val = pm_runtime_get_sync(ctx->dev); - if (ret_val < 0) + if (ret_val < 0) { + pm_runtime_put_sync(ctx->dev); return ret_val; + } ret_val = sst_send_byte_stream_mrfld(ctx, bytes); sst_pm_runtime_put(ctx); diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 5a27861..3dc7358 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -352,10 +352,9 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, * copy from mailbox **/ if (msg_high.part.large) { - data = kzalloc(msg_low, GFP_KERNEL); + data = kmemdup((void *)msg->mailbox_data, msg_low, GFP_KERNEL); if (!data) return; - memcpy(data, (void *) msg->mailbox_data, msg_low); /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; cmd_id = dsp_hdr->cmd_id; diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 4c01bb4..5bbaa66 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -701,6 +701,8 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) if (byt == NULL) return -ENOMEM; + byt->dev = dev; + ipc = &byt->ipc; ipc->dev = dev; ipc->ops.tx_msg = byt_tx_msg; diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 7ab8cc9..d9f81b8 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -126,6 +126,7 @@ static struct snd_soc_dai_link byt_max98090_dais[] = { static struct snd_soc_card byt_max98090_card = { .name = "byt-max98090", + .owner = THIS_MODULE, .dai_link = byt_max98090_dais, .num_links = ARRAY_SIZE(byt_max98090_dais), .dapm_widgets = byt_max98090_widgets, diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index ae89b9b9..de9788a 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -197,6 +197,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { static struct snd_soc_card byt_rt5640_card = { .name = "byt-rt5640", + .owner = THIS_MODULE, .dai_link = byt_rt5640_dais, .num_links = ARRAY_SIZE(byt_rt5640_dais), .dapm_widgets = byt_rt5640_widgets, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7f55d59..c445312 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -185,6 +185,7 @@ static struct snd_soc_dai_link byt_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_byt = { .name = "baytrailcraudio", + .owner = THIS_MODULE, .dai_link = byt_dailink, .num_links = ARRAY_SIZE(byt_dailink), .dapm_widgets = byt_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index d604ee8..49f4869 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -69,12 +69,12 @@ static const struct snd_soc_dapm_route cht_audio_map[] = { {"Headphone", NULL, "HPR"}, {"Ext Spk", NULL, "SPKL"}, {"Ext Spk", NULL, "SPKR"}, - {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"HiFi Playback", NULL, "ssp2 Tx"}, {"ssp2 Tx", NULL, "codec_out0"}, {"ssp2 Tx", NULL, "codec_out1"}, {"codec_in0", NULL, "ssp2 Rx" }, {"codec_in1", NULL, "ssp2 Rx" }, - {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"ssp2 Rx", NULL, "HiFi Capture"}, }; static const struct snd_kcontrol_new cht_mc_controls[] = { @@ -104,21 +104,17 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, static int cht_ti_jack_event(struct notifier_block *nb, unsigned long event, void *data) { - struct snd_soc_jack *jack = (struct snd_soc_jack *)data; - struct snd_soc_dai *codec_dai = jack->card->rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &jack->card->dapm; if (event & SND_JACK_MICROPHONE) { - - snd_soc_dapm_force_enable_pin(&codec->dapm, "SHDN"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_force_enable_pin(dapm, "SHDN"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_sync(dapm); } else { - - snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS"); - snd_soc_dapm_disable_pin(&codec->dapm, "SHDN"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SHDN"); + snd_soc_dapm_sync(dapm); } return 0; @@ -279,6 +275,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "chtmax98090", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .aux_dev = &cht_max98090_headset_dev, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..7be8461 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -305,6 +305,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_chtrt5645 = { .name = "chtrt5645", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, @@ -317,6 +318,7 @@ static struct snd_soc_card snd_soc_card_chtrt5645 = { static struct snd_soc_card snd_soc_card_chtrt5650 = { .name = "chtrt5650", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5b..23fe040 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -323,6 +323,7 @@ static int cht_resume_post(struct snd_soc_card *card) /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "cherrytrailcraudio", + .owner = THIS_MODULE, .dai_link = cht_dailink, .num_links = ARRAY_SIZE(cht_dailink), .dapm_widgets = cht_dapm_widgets, diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h index 396d545..cbd568e 100644 --- a/sound/soc/intel/common/sst-dsp-priv.h +++ b/sound/soc/intel/common/sst-dsp-priv.h @@ -22,6 +22,8 @@ #include <linux/interrupt.h> #include <linux/firmware.h> +#include "../skylake/skl-sst-dsp.h" + struct sst_mem_block; struct sst_module; struct sst_fw; @@ -258,6 +260,8 @@ struct sst_mem_block { */ struct sst_dsp { + /* Shared for all platforms */ + /* runtime */ struct sst_dsp_device *sst_dev; spinlock_t spinlock; /* IPC locking */ @@ -268,10 +272,6 @@ struct sst_dsp { int irq; u32 id; - /* list of free and used ADSP memory blocks */ - struct list_head used_block_list; - struct list_head free_block_list; - /* operations */ struct sst_ops *ops; @@ -284,6 +284,12 @@ struct sst_dsp { /* mailbox */ struct sst_mailbox mailbox; + /* HSW/Byt data */ + + /* list of free and used ADSP memory blocks */ + struct list_head used_block_list; + struct list_head free_block_list; + /* SST FW files loaded and their modules */ struct list_head module_list; struct list_head fw_list; @@ -299,6 +305,15 @@ struct sst_dsp { /* DMA FW loading */ struct sst_dma *dma; bool fw_use_dma; + + /* SKL data */ + + /* To allocate CL dma buffers */ + struct skl_dsp_loader_ops dsp_ops; + struct skl_dsp_fw_ops fw_ops; + int sst_state; + struct skl_cl_dev cl_dev; + u32 intr_status; }; /* Size optimised DRAM/IRAM memcpy */ diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index 64e9421..a627236 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -20,6 +20,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/io.h> +#include <linux/delay.h> #include "sst-dsp.h" #include "sst-dsp-priv.h" @@ -196,6 +197,22 @@ int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, } EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64_unlocked); +/* This is for registers bits with attribute RWC */ +void sst_dsp_shim_update_bits_forced_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + unsigned int old, new; + u32 ret; + + ret = sst_dsp_shim_read_unlocked(sst, offset); + + old = ret; + new = (old & (~mask)) | (value & mask); + + sst_dsp_shim_write_unlocked(sst, offset, new); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits_forced_unlocked); + int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, u32 mask, u32 value) { @@ -222,6 +239,60 @@ int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, } EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); +/* This is for registers bits with attribute RWC */ +void sst_dsp_shim_update_bits_forced(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst_dsp_shim_update_bits_forced_unlocked(sst, offset, mask, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits_forced); + +int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask, + u32 target, u32 timeout, char *operation) +{ + int time, ret; + u32 reg; + bool done = false; + + /* + * we will poll for couple of ms using mdelay, if not successful + * then go to longer sleep using usleep_range + */ + + /* check if set state successful */ + for (time = 0; time < 5; time++) { + if ((sst_dsp_shim_read_unlocked(ctx, offset) & mask) == target) { + done = true; + break; + } + mdelay(1); + } + + if (done == false) { + /* sleeping in 10ms steps so adjust timeout value */ + timeout /= 10; + + for (time = 0; time < timeout; time++) { + if ((sst_dsp_shim_read_unlocked(ctx, offset) & mask) == target) + break; + + usleep_range(5000, 10000); + } + } + + reg = sst_dsp_shim_read_unlocked(ctx, offset); + dev_info(ctx->dev, "FW Poll Status: reg=%#x %s %s\n", reg, operation, + (time < timeout) ? "successful" : "timedout"); + ret = time < timeout ? 0 : -ETIME; + + return ret; +} +EXPORT_SYMBOL_GPL(sst_dsp_register_poll); + void sst_dsp_dump(struct sst_dsp *sst) { if (sst->ops->dump) diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index 96aeb25..1f45f18 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -230,6 +230,8 @@ void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value); u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset); int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, u64 mask, u64 value); +void sst_dsp_shim_update_bits_forced(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); /* SHIM Read / Write Unlocked for callers already holding sst lock */ void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value); @@ -240,6 +242,8 @@ void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value); u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, u64 mask, u64 value); +void sst_dsp_shim_update_bits_forced_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); /* Internal generic low-level SST IO functions - can be overidden */ void sst_shim32_write(void __iomem *addr, u32 offset, u32 value); @@ -278,6 +282,8 @@ void sst_dsp_inbox_read(struct sst_dsp *dsp, void *message, size_t bytes); void sst_dsp_outbox_write(struct sst_dsp *dsp, void *message, size_t bytes); void sst_dsp_outbox_read(struct sst_dsp *dsp, void *message, size_t bytes); void sst_dsp_mailbox_dump(struct sst_dsp *dsp, size_t bytes); +int sst_dsp_register_poll(struct sst_dsp *dsp, u32 offset, u32 mask, + u32 expected_value, u32 timeout, char *operation); /* Debug */ void sst_dsp_dump(struct sst_dsp *sst); diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f95f271..f6efa9d 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2119,6 +2119,8 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (hsw == NULL) return -ENOMEM; + hsw->dev = dev; + ipc = &hsw->ipc; ipc->dev = dev; ipc->ops.tx_msg = hsw_tx_msg; diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile new file mode 100644 index 0000000..27db221 --- /dev/null +++ b/sound/soc/intel/skylake/Makefile @@ -0,0 +1,9 @@ +snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o + +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o + +# Skylake IPC Support +snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ + skl-sst.o + +obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c new file mode 100644 index 0000000..826d4fd --- /dev/null +++ b/sound/soc/intel/skylake/skl-messages.c @@ -0,0 +1,884 @@ +/* + * skl-message.c - HDA DSP interface for FW registration, Pipe and Module + * configurations + * + * Copyright (C) 2015 Intel Corp + * Author:Rafal Redzimski <rafal.f.redzimski@intel.com> + * Jeeja KP <jeeja.kp@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ + +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include "skl-sst-dsp.h" +#include "skl-sst-ipc.h" +#include "skl.h" +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-topology.h" +#include "skl-tplg-interface.h" + +static int skl_alloc_dma_buf(struct device *dev, + struct snd_dma_buffer *dmab, size_t size) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + return bus->io_ops->dma_alloc_pages(bus, SNDRV_DMA_TYPE_DEV, size, dmab); +} + +static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + bus->io_ops->dma_free_pages(bus, dmab); + + return 0; +} + +int skl_init_dsp(struct skl *skl) +{ + void __iomem *mmio_base; + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + int irq = bus->irq; + struct skl_dsp_loader_ops loader_ops; + int ret; + + loader_ops.alloc_dma_buf = skl_alloc_dma_buf; + loader_ops.free_dma_buf = skl_free_dma_buf; + + /* enable ppcap interrupt */ + snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + + /* read the BAR of the ADSP MMIO */ + mmio_base = pci_ioremap_bar(skl->pci, 4); + if (mmio_base == NULL) { + dev_err(bus->dev, "ioremap error\n"); + return -ENXIO; + } + + ret = skl_sst_dsp_init(bus->dev, mmio_base, irq, + loader_ops, &skl->skl_sst); + + dev_dbg(bus->dev, "dsp registration status=%d\n", ret); + + return ret; +} + +void skl_free_dsp(struct skl *skl) +{ + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl_sst *ctx = skl->skl_sst; + + /* disable ppcap interrupt */ + snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); + + skl_sst_dsp_cleanup(bus->dev, ctx); + if (ctx->dsp->addr.lpe) + iounmap(ctx->dsp->addr.lpe); +} + +int skl_suspend_dsp(struct skl *skl) +{ + struct skl_sst *ctx = skl->skl_sst; + int ret; + + /* if ppcap is not supported return 0 */ + if (!skl->ebus.ppcap) + return 0; + + ret = skl_dsp_sleep(ctx->dsp); + if (ret < 0) + return ret; + + /* disable ppcap interrupt */ + snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, false); + snd_hdac_ext_bus_ppcap_enable(&skl->ebus, false); + + return 0; +} + +int skl_resume_dsp(struct skl *skl) +{ + struct skl_sst *ctx = skl->skl_sst; + + /* if ppcap is not supported return 0 */ + if (!skl->ebus.ppcap) + return 0; + + /* enable ppcap interrupt */ + snd_hdac_ext_bus_ppcap_enable(&skl->ebus, true); + snd_hdac_ext_bus_ppcap_int_enable(&skl->ebus, true); + + return skl_dsp_wake(ctx->dsp); +} + +enum skl_bitdepth skl_get_bit_depth(int params) +{ + switch (params) { + case 8: + return SKL_DEPTH_8BIT; + + case 16: + return SKL_DEPTH_16BIT; + + case 24: + return SKL_DEPTH_24BIT; + + case 32: + return SKL_DEPTH_32BIT; + + default: + return SKL_DEPTH_INVALID; + + } +} + +static u32 skl_create_channel_map(enum skl_ch_cfg ch_cfg) +{ + u32 config; + + switch (ch_cfg) { + case SKL_CH_CFG_MONO: + config = (0xFFFFFFF0 | SKL_CHANNEL_LEFT); + break; + + case SKL_CH_CFG_STEREO: + config = (0xFFFFFF00 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_RIGHT << 4)); + break; + + case SKL_CH_CFG_2_1: + config = (0xFFFFF000 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_RIGHT << 4) + | (SKL_CHANNEL_LFE << 8)); + break; + + case SKL_CH_CFG_3_0: + config = (0xFFFFF000 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_CENTER << 4) + | (SKL_CHANNEL_RIGHT << 8)); + break; + + case SKL_CH_CFG_3_1: + config = (0xFFFF0000 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_CENTER << 4) + | (SKL_CHANNEL_RIGHT << 8) + | (SKL_CHANNEL_LFE << 12)); + break; + + case SKL_CH_CFG_QUATRO: + config = (0xFFFF0000 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_RIGHT << 4) + | (SKL_CHANNEL_LEFT_SURROUND << 8) + | (SKL_CHANNEL_RIGHT_SURROUND << 12)); + break; + + case SKL_CH_CFG_4_0: + config = (0xFFFF0000 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_CENTER << 4) + | (SKL_CHANNEL_RIGHT << 8) + | (SKL_CHANNEL_CENTER_SURROUND << 12)); + break; + + case SKL_CH_CFG_5_0: + config = (0xFFF00000 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_CENTER << 4) + | (SKL_CHANNEL_RIGHT << 8) + | (SKL_CHANNEL_LEFT_SURROUND << 12) + | (SKL_CHANNEL_RIGHT_SURROUND << 16)); + break; + + case SKL_CH_CFG_5_1: + config = (0xFF000000 | SKL_CHANNEL_CENTER + | (SKL_CHANNEL_LEFT << 4) + | (SKL_CHANNEL_RIGHT << 8) + | (SKL_CHANNEL_LEFT_SURROUND << 12) + | (SKL_CHANNEL_RIGHT_SURROUND << 16) + | (SKL_CHANNEL_LFE << 20)); + break; + + case SKL_CH_CFG_DUAL_MONO: + config = (0xFFFFFF00 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_LEFT << 4)); + break; + + case SKL_CH_CFG_I2S_DUAL_STEREO_0: + config = (0xFFFFFF00 | SKL_CHANNEL_LEFT + | (SKL_CHANNEL_RIGHT << 4)); + break; + + case SKL_CH_CFG_I2S_DUAL_STEREO_1: + config = (0xFFFF00FF | (SKL_CHANNEL_LEFT << 8) + | (SKL_CHANNEL_RIGHT << 12)); + break; + + default: + config = 0xFFFFFFFF; + break; + + } + + return config; +} + +/* + * Each module in DSP expects a base module configuration, which consists of + * PCM format information, which we calculate in driver and resource values + * which are read from widget information passed through topology binary + * This is send when we create a module with INIT_INSTANCE IPC msg + */ +static void skl_set_base_module_format(struct skl_sst *ctx, + struct skl_module_cfg *mconfig, + struct skl_base_cfg *base_cfg) +{ + struct skl_module_fmt *format = &mconfig->in_fmt; + + base_cfg->audio_fmt.number_of_channels = (u8)format->channels; + + base_cfg->audio_fmt.s_freq = format->s_freq; + base_cfg->audio_fmt.bit_depth = format->bit_depth; + base_cfg->audio_fmt.valid_bit_depth = format->valid_bit_depth; + base_cfg->audio_fmt.ch_cfg = format->ch_cfg; + + dev_dbg(ctx->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n", + format->bit_depth, format->valid_bit_depth, + format->ch_cfg); + + base_cfg->audio_fmt.channel_map = skl_create_channel_map( + base_cfg->audio_fmt.ch_cfg); + + base_cfg->audio_fmt.interleaving = SKL_INTERLEAVING_PER_CHANNEL; + + base_cfg->cps = mconfig->mcps; + base_cfg->ibs = mconfig->ibs; + base_cfg->obs = mconfig->obs; +} + +/* + * Copies copier capabilities into copier module and updates copier module + * config size. + */ +static void skl_copy_copier_caps(struct skl_module_cfg *mconfig, + struct skl_cpr_cfg *cpr_mconfig) +{ + if (mconfig->formats_config.caps_size == 0) + return; + + memcpy(cpr_mconfig->gtw_cfg.config_data, + mconfig->formats_config.caps, + mconfig->formats_config.caps_size); + + cpr_mconfig->gtw_cfg.config_length = + (mconfig->formats_config.caps_size) / 4; +} + +/* + * Calculate the gatewat settings required for copier module, type of + * gateway and index of gateway to use + */ +static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx, + struct skl_module_cfg *mconfig, + struct skl_cpr_cfg *cpr_mconfig) +{ + union skl_connector_node_id node_id = {0}; + struct skl_pipe_params *params = mconfig->pipe->p_params; + + switch (mconfig->dev_type) { + case SKL_DEVICE_BT: + node_id.node.dma_type = + (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? + SKL_DMA_I2S_LINK_OUTPUT_CLASS : + SKL_DMA_I2S_LINK_INPUT_CLASS; + node_id.node.vindex = params->host_dma_id + + (mconfig->vbus_id << 3); + break; + + case SKL_DEVICE_I2S: + node_id.node.dma_type = + (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? + SKL_DMA_I2S_LINK_OUTPUT_CLASS : + SKL_DMA_I2S_LINK_INPUT_CLASS; + node_id.node.vindex = params->host_dma_id + + (mconfig->time_slot << 1) + + (mconfig->vbus_id << 3); + break; + + case SKL_DEVICE_DMIC: + node_id.node.dma_type = SKL_DMA_DMIC_LINK_INPUT_CLASS; + node_id.node.vindex = mconfig->vbus_id + + (mconfig->time_slot); + break; + + case SKL_DEVICE_HDALINK: + node_id.node.dma_type = + (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? + SKL_DMA_HDA_LINK_OUTPUT_CLASS : + SKL_DMA_HDA_LINK_INPUT_CLASS; + node_id.node.vindex = params->link_dma_id; + break; + + default: + node_id.node.dma_type = + (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? + SKL_DMA_HDA_HOST_OUTPUT_CLASS : + SKL_DMA_HDA_HOST_INPUT_CLASS; + node_id.node.vindex = params->host_dma_id; + break; + } + + cpr_mconfig->gtw_cfg.node_id = node_id.val; + + if (SKL_CONN_SOURCE == mconfig->hw_conn_type) + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs; + else + cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs; + + cpr_mconfig->cpr_feature_mask = 0; + cpr_mconfig->gtw_cfg.config_length = 0; + + skl_copy_copier_caps(mconfig, cpr_mconfig); +} + +static void skl_setup_out_format(struct skl_sst *ctx, + struct skl_module_cfg *mconfig, + struct skl_audio_data_format *out_fmt) +{ + struct skl_module_fmt *format = &mconfig->out_fmt; + + out_fmt->number_of_channels = (u8)format->channels; + out_fmt->s_freq = format->s_freq; + out_fmt->bit_depth = format->bit_depth; + out_fmt->valid_bit_depth = format->valid_bit_depth; + out_fmt->ch_cfg = format->ch_cfg; + + out_fmt->channel_map = skl_create_channel_map(out_fmt->ch_cfg); + out_fmt->interleaving = SKL_INTERLEAVING_PER_CHANNEL; + + dev_dbg(ctx->dev, "copier out format chan=%d fre=%d bitdepth=%d\n", + out_fmt->number_of_channels, format->s_freq, format->bit_depth); +} + +/* + * DSP needs SRC module for frequency conversion, SRC takes base module + * configuration and the target frequency as extra parameter passed as src + * config + */ +static void skl_set_src_format(struct skl_sst *ctx, + struct skl_module_cfg *mconfig, + struct skl_src_module_cfg *src_mconfig) +{ + struct skl_module_fmt *fmt = &mconfig->out_fmt; + + skl_set_base_module_format(ctx, mconfig, + (struct skl_base_cfg *)src_mconfig); + + src_mconfig->src_cfg = fmt->s_freq; +} + +/* + * DSP needs updown module to do channel conversion. updown module take base + * module configuration and channel configuration + * It also take coefficients and now we have defaults applied here + */ +static void skl_set_updown_mixer_format(struct skl_sst *ctx, + struct skl_module_cfg *mconfig, + struct skl_up_down_mixer_cfg *mixer_mconfig) +{ + struct skl_module_fmt *fmt = &mconfig->out_fmt; + int i = 0; + + skl_set_base_module_format(ctx, mconfig, + (struct skl_base_cfg *)mixer_mconfig); + mixer_mconfig->out_ch_cfg = fmt->ch_cfg; + + /* Select F/W default coefficient */ + mixer_mconfig->coeff_sel = 0x0; + + /* User coeff, don't care since we are selecting F/W defaults */ + for (i = 0; i < UP_DOWN_MIXER_MAX_COEFF; i++) + mixer_mconfig->coeff[i] = 0xDEADBEEF; +} + +/* + * 'copier' is DSP internal module which copies data from Host DMA (HDA host + * dma) or link (hda link, SSP, PDM) + * Here we calculate the copier module parameters, like PCM format, output + * format, gateway settings + * copier_module_config is sent as input buffer with INIT_INSTANCE IPC msg + */ +static void skl_set_copier_format(struct skl_sst *ctx, + struct skl_module_cfg *mconfig, + struct skl_cpr_cfg *cpr_mconfig) +{ + struct skl_audio_data_format *out_fmt = &cpr_mconfig->out_fmt; + struct skl_base_cfg *base_cfg = (struct skl_base_cfg *)cpr_mconfig; + + skl_set_base_module_format(ctx, mconfig, base_cfg); + + skl_setup_out_format(ctx, mconfig, out_fmt); + skl_setup_cpr_gateway_cfg(ctx, mconfig, cpr_mconfig); +} + +static u16 skl_get_module_param_size(struct skl_sst *ctx, + struct skl_module_cfg *mconfig) +{ + u16 param_size; + + switch (mconfig->m_type) { + case SKL_MODULE_TYPE_COPIER: + param_size = sizeof(struct skl_cpr_cfg); + param_size += mconfig->formats_config.caps_size; + return param_size; + + case SKL_MODULE_TYPE_SRCINT: + return sizeof(struct skl_src_module_cfg); + + case SKL_MODULE_TYPE_UPDWMIX: + return sizeof(struct skl_up_down_mixer_cfg); + + default: + /* + * return only base cfg when no specific module type is + * specified + */ + return sizeof(struct skl_base_cfg); + } + + return 0; +} + +/* + * DSP firmware supports various modules like copier, SRC, updown etc. + * These modules required various parameters to be calculated and sent for + * the module initialization to DSP. By default a generic module needs only + * base module format configuration + */ + +static int skl_set_module_format(struct skl_sst *ctx, + struct skl_module_cfg *module_config, + u16 *module_config_size, + void **param_data) +{ + u16 param_size; + + param_size = skl_get_module_param_size(ctx, module_config); + + *param_data = kzalloc(param_size, GFP_KERNEL); + if (NULL == *param_data) + return -ENOMEM; + + *module_config_size = param_size; + + switch (module_config->m_type) { + case SKL_MODULE_TYPE_COPIER: + skl_set_copier_format(ctx, module_config, *param_data); + break; + + case SKL_MODULE_TYPE_SRCINT: + skl_set_src_format(ctx, module_config, *param_data); + break; + + case SKL_MODULE_TYPE_UPDWMIX: + skl_set_updown_mixer_format(ctx, module_config, *param_data); + break; + + default: + skl_set_base_module_format(ctx, module_config, *param_data); + break; + + } + + dev_dbg(ctx->dev, "Module type=%d config size: %d bytes\n", + module_config->id.module_id, param_size); + print_hex_dump(KERN_DEBUG, "Module params:", DUMP_PREFIX_OFFSET, 8, 4, + *param_data, param_size, false); + return 0; +} + +static int skl_get_queue_index(struct skl_module_pin *mpin, + struct skl_module_inst_id id, int max) +{ + int i; + + for (i = 0; i < max; i++) { + if (mpin[i].id.module_id == id.module_id && + mpin[i].id.instance_id == id.instance_id) + return i; + } + + return -EINVAL; +} + +/* + * Allocates queue for each module. + * if dynamic, the pin_index is allocated 0 to max_pin. + * In static, the pin_index is fixed based on module_id and instance id + */ +static int skl_alloc_queue(struct skl_module_pin *mpin, + struct skl_module_inst_id id, int max) +{ + int i; + + /* + * if pin in dynamic, find first free pin + * otherwise find match module and instance id pin as topology will + * ensure a unique pin is assigned to this so no need to + * allocate/free + */ + for (i = 0; i < max; i++) { + if (mpin[i].is_dynamic) { + if (!mpin[i].in_use) { + mpin[i].in_use = true; + mpin[i].id.module_id = id.module_id; + mpin[i].id.instance_id = id.instance_id; + return i; + } + } else { + if (mpin[i].id.module_id == id.module_id && + mpin[i].id.instance_id == id.instance_id) + return i; + } + } + + return -EINVAL; +} + +static void skl_free_queue(struct skl_module_pin *mpin, int q_index) +{ + if (mpin[q_index].is_dynamic) { + mpin[q_index].in_use = false; + mpin[q_index].id.module_id = 0; + mpin[q_index].id.instance_id = 0; + } +} + +/* + * A module needs to be instanataited in DSP. A mdoule is present in a + * collection of module referred as a PIPE. + * We first calculate the module format, based on module type and then + * invoke the DSP by sending IPC INIT_INSTANCE using ipc helper + */ +int skl_init_module(struct skl_sst *ctx, + struct skl_module_cfg *mconfig, char *param) +{ + u16 module_config_size = 0; + void *param_data = NULL; + int ret; + struct skl_ipc_init_instance_msg msg; + + dev_dbg(ctx->dev, "%s: module_id = %d instance=%d\n", __func__, + mconfig->id.module_id, mconfig->id.instance_id); + + if (mconfig->pipe->state != SKL_PIPE_CREATED) { + dev_err(ctx->dev, "Pipe not created state= %d pipe_id= %d\n", + mconfig->pipe->state, mconfig->pipe->ppl_id); + return -EIO; + } + + ret = skl_set_module_format(ctx, mconfig, + &module_config_size, ¶m_data); + if (ret < 0) { + dev_err(ctx->dev, "Failed to set module format ret=%d\n", ret); + return ret; + } + + msg.module_id = mconfig->id.module_id; + msg.instance_id = mconfig->id.instance_id; + msg.ppl_instance_id = mconfig->pipe->ppl_id; + msg.param_data_size = module_config_size; + msg.core_id = mconfig->core_id; + + ret = skl_ipc_init_instance(&ctx->ipc, &msg, param_data); + if (ret < 0) { + dev_err(ctx->dev, "Failed to init instance ret=%d\n", ret); + kfree(param_data); + return ret; + } + mconfig->m_state = SKL_MODULE_INIT_DONE; + + return ret; +} + +static void skl_dump_bind_info(struct skl_sst *ctx, struct skl_module_cfg + *src_module, struct skl_module_cfg *dst_module) +{ + dev_dbg(ctx->dev, "%s: src module_id = %d src_instance=%d\n", + __func__, src_module->id.module_id, src_module->id.instance_id); + dev_dbg(ctx->dev, "%s: dst_module=%d dst_instacne=%d\n", __func__, + dst_module->id.module_id, dst_module->id.instance_id); + + dev_dbg(ctx->dev, "src_module state = %d dst module state = %d\n", + src_module->m_state, dst_module->m_state); +} + +/* + * On module freeup, we need to unbind the module with modules + * it is already bind. + * Find the pin allocated and unbind then using bind_unbind IPC + */ +int skl_unbind_modules(struct skl_sst *ctx, + struct skl_module_cfg *src_mcfg, + struct skl_module_cfg *dst_mcfg) +{ + int ret; + struct skl_ipc_bind_unbind_msg msg; + struct skl_module_inst_id src_id = src_mcfg->id; + struct skl_module_inst_id dst_id = dst_mcfg->id; + int in_max = dst_mcfg->max_in_queue; + int out_max = src_mcfg->max_out_queue; + int src_index, dst_index; + + skl_dump_bind_info(ctx, src_mcfg, dst_mcfg); + + if (src_mcfg->m_state != SKL_MODULE_BIND_DONE) + return 0; + + /* + * if intra module unbind, check if both modules are BIND, + * then send unbind + */ + if ((src_mcfg->pipe->ppl_id != dst_mcfg->pipe->ppl_id) && + dst_mcfg->m_state != SKL_MODULE_BIND_DONE) + return 0; + else if (src_mcfg->m_state < SKL_MODULE_INIT_DONE && + dst_mcfg->m_state < SKL_MODULE_INIT_DONE) + return 0; + + /* get src queue index */ + src_index = skl_get_queue_index(src_mcfg->m_out_pin, dst_id, out_max); + if (src_index < 0) + return -EINVAL; + + msg.src_queue = src_mcfg->m_out_pin[src_index].pin_index; + + /* get dst queue index */ + dst_index = skl_get_queue_index(dst_mcfg->m_in_pin, src_id, in_max); + if (dst_index < 0) + return -EINVAL; + + msg.dst_queue = dst_mcfg->m_in_pin[dst_index].pin_index; + + msg.module_id = src_mcfg->id.module_id; + msg.instance_id = src_mcfg->id.instance_id; + msg.dst_module_id = dst_mcfg->id.module_id; + msg.dst_instance_id = dst_mcfg->id.instance_id; + msg.bind = false; + + ret = skl_ipc_bind_unbind(&ctx->ipc, &msg); + if (!ret) { + src_mcfg->m_state = SKL_MODULE_UNINIT; + /* free queue only if unbind is success */ + skl_free_queue(src_mcfg->m_out_pin, src_index); + skl_free_queue(dst_mcfg->m_in_pin, dst_index); + } + + return ret; +} + +/* + * Once a module is instantiated it need to be 'bind' with other modules in + * the pipeline. For binding we need to find the module pins which are bind + * together + * This function finds the pins and then sends bund_unbind IPC message to + * DSP using IPC helper + */ +int skl_bind_modules(struct skl_sst *ctx, + struct skl_module_cfg *src_mcfg, + struct skl_module_cfg *dst_mcfg) +{ + int ret; + struct skl_ipc_bind_unbind_msg msg; + struct skl_module_inst_id src_id = src_mcfg->id; + struct skl_module_inst_id dst_id = dst_mcfg->id; + int in_max = dst_mcfg->max_in_queue; + int out_max = src_mcfg->max_out_queue; + int src_index, dst_index; + + skl_dump_bind_info(ctx, src_mcfg, dst_mcfg); + + if (src_mcfg->m_state < SKL_MODULE_INIT_DONE && + dst_mcfg->m_state < SKL_MODULE_INIT_DONE) + return 0; + + src_index = skl_alloc_queue(src_mcfg->m_out_pin, dst_id, out_max); + if (src_index < 0) + return -EINVAL; + + msg.src_queue = src_mcfg->m_out_pin[src_index].pin_index; + dst_index = skl_alloc_queue(dst_mcfg->m_in_pin, src_id, in_max); + if (dst_index < 0) { + skl_free_queue(src_mcfg->m_out_pin, src_index); + return -EINVAL; + } + + msg.dst_queue = dst_mcfg->m_in_pin[dst_index].pin_index; + + dev_dbg(ctx->dev, "src queue = %d dst queue =%d\n", + msg.src_queue, msg.dst_queue); + + msg.module_id = src_mcfg->id.module_id; + msg.instance_id = src_mcfg->id.instance_id; + msg.dst_module_id = dst_mcfg->id.module_id; + msg.dst_instance_id = dst_mcfg->id.instance_id; + msg.bind = true; + + ret = skl_ipc_bind_unbind(&ctx->ipc, &msg); + + if (!ret) { + src_mcfg->m_state = SKL_MODULE_BIND_DONE; + } else { + /* error case , if IPC fails, clear the queue index */ + skl_free_queue(src_mcfg->m_out_pin, src_index); + skl_free_queue(dst_mcfg->m_in_pin, dst_index); + } + + return ret; +} + +static int skl_set_pipe_state(struct skl_sst *ctx, struct skl_pipe *pipe, + enum skl_ipc_pipeline_state state) +{ + dev_dbg(ctx->dev, "%s: pipe_satate = %d\n", __func__, state); + + return skl_ipc_set_pipeline_state(&ctx->ipc, pipe->ppl_id, state); +} + +/* + * A pipeline is a collection of modules. Before a module in instantiated a + * pipeline needs to be created for it. + * This function creates pipeline, by sending create pipeline IPC messages + * to FW + */ +int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe) +{ + int ret; + + dev_dbg(ctx->dev, "%s: pipe_id = %d\n", __func__, pipe->ppl_id); + + ret = skl_ipc_create_pipeline(&ctx->ipc, pipe->memory_pages, + pipe->pipe_priority, pipe->ppl_id); + if (ret < 0) { + dev_err(ctx->dev, "Failed to create pipeline\n"); + return ret; + } + + pipe->state = SKL_PIPE_CREATED; + + return 0; +} + +/* + * A pipeline needs to be deleted on cleanup. If a pipeline is running, then + * pause the pipeline first and then delete it + * The pipe delete is done by sending delete pipeline IPC. DSP will stop the + * DMA engines and releases resources + */ +int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe) +{ + int ret; + + dev_dbg(ctx->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id); + + /* If pipe is not started, do not try to stop the pipe in FW. */ + if (pipe->state > SKL_PIPE_STARTED) { + ret = skl_set_pipe_state(ctx, pipe, PPL_PAUSED); + if (ret < 0) { + dev_err(ctx->dev, "Failed to stop pipeline\n"); + return ret; + } + + pipe->state = SKL_PIPE_PAUSED; + } else { + /* If pipe was not created in FW, do not try to delete it */ + if (pipe->state < SKL_PIPE_CREATED) + return 0; + + ret = skl_ipc_delete_pipeline(&ctx->ipc, pipe->ppl_id); + if (ret < 0) + dev_err(ctx->dev, "Failed to delete pipeline\n"); + } + + return ret; +} + +/* + * A pipeline is also a scheduling entity in DSP which can be run, stopped + * For processing data the pipe need to be run by sending IPC set pipe state + * to DSP + */ +int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe) +{ + int ret; + + dev_dbg(ctx->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id); + + /* If pipe was not created in FW, do not try to pause or delete */ + if (pipe->state < SKL_PIPE_CREATED) + return 0; + + /* Pipe has to be paused before it is started */ + ret = skl_set_pipe_state(ctx, pipe, PPL_PAUSED); + if (ret < 0) { + dev_err(ctx->dev, "Failed to pause pipe\n"); + return ret; + } + + pipe->state = SKL_PIPE_PAUSED; + + ret = skl_set_pipe_state(ctx, pipe, PPL_RUNNING); + if (ret < 0) { + dev_err(ctx->dev, "Failed to start pipe\n"); + return ret; + } + + pipe->state = SKL_PIPE_STARTED; + + return 0; +} + +/* + * Stop the pipeline by sending set pipe state IPC + * DSP doesnt implement stop so we always send pause message + */ +int skl_stop_pipe(struct skl_sst *ctx, struct skl_pipe *pipe) +{ + int ret; + + dev_dbg(ctx->dev, "In %s pipe=%d\n", __func__, pipe->ppl_id); + + /* If pipe was not created in FW, do not try to pause or delete */ + if (pipe->state < SKL_PIPE_PAUSED) + return 0; + + ret = skl_set_pipe_state(ctx, pipe, PPL_PAUSED); + if (ret < 0) { + dev_dbg(ctx->dev, "Failed to stop pipe\n"); + return ret; + } + + pipe->state = SKL_PIPE_CREATED; + + return 0; +} diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c new file mode 100644 index 0000000..13036b1 --- /dev/null +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -0,0 +1,140 @@ +/* + * skl-nhlt.c - Intel SKL Platform NHLT parsing + * + * Copyright (C) 2015 Intel Corp + * Author: Sanjiv Kumar <sanjiv.kumar@intel.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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ +#include "skl.h" + +/* Unique identification for getting NHLT blobs */ +static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, + 0xA4, 0x1F, 0x7B, 0x5D, 0xCE, 0x24, 0xC5, 0x53}; + +#define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS" + +void __iomem *skl_nhlt_init(struct device *dev) +{ + acpi_handle handle; + union acpi_object *obj; + struct nhlt_resource_desc *nhlt_ptr = NULL; + + if (ACPI_FAILURE(acpi_get_handle(NULL, DSDT_NHLT_PATH, &handle))) { + dev_err(dev, "Requested NHLT device not found\n"); + return NULL; + } + + obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL); + if (obj && obj->type == ACPI_TYPE_BUFFER) { + nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; + + return ioremap_cache(nhlt_ptr->min_addr, nhlt_ptr->length); + } + + dev_err(dev, "device specific method to extract NHLT blob failed\n"); + return NULL; +} + +void skl_nhlt_free(void __iomem *addr) +{ + iounmap(addr); + addr = NULL; +} + +static struct nhlt_specific_cfg *skl_get_specific_cfg( + struct device *dev, struct nhlt_fmt *fmt, + u8 no_ch, u32 rate, u16 bps) +{ + struct nhlt_specific_cfg *sp_config; + struct wav_fmt *wfmt; + struct nhlt_fmt_cfg *fmt_config = fmt->fmt_config; + int i; + + dev_dbg(dev, "Format count =%d\n", fmt->fmt_count); + + for (i = 0; i < fmt->fmt_count; i++) { + wfmt = &fmt_config->fmt_ext.fmt; + dev_dbg(dev, "ch=%d fmt=%d s_rate=%d\n", wfmt->channels, + wfmt->bits_per_sample, wfmt->samples_per_sec); + if (wfmt->channels == no_ch && wfmt->samples_per_sec == rate && + wfmt->bits_per_sample == bps) { + sp_config = &fmt_config->config; + + return sp_config; + } + + fmt_config = (struct nhlt_fmt_cfg *)(fmt_config->config.caps + + fmt_config->config.size); + } + + return NULL; +} + +static void dump_config(struct device *dev, u32 instance_id, u8 linktype, + u8 s_fmt, u8 num_channels, u32 s_rate, u8 dirn, u16 bps) +{ + dev_dbg(dev, "Input configuration\n"); + dev_dbg(dev, "ch=%d fmt=%d s_rate=%d\n", num_channels, s_fmt, s_rate); + dev_dbg(dev, "vbus_id=%d link_type=%d\n", instance_id, linktype); + dev_dbg(dev, "bits_per_sample=%d\n", bps); +} + +static bool skl_check_ep_match(struct device *dev, struct nhlt_endpoint *epnt, + u32 instance_id, u8 link_type, u8 dirn) +{ + dev_dbg(dev, "vbus_id=%d link_type=%d dir=%d\n", + epnt->virtual_bus_id, epnt->linktype, epnt->direction); + + if ((epnt->virtual_bus_id == instance_id) && + (epnt->linktype == link_type) && + (epnt->direction == dirn)) + return true; + else + return false; +} + +struct nhlt_specific_cfg +*skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, + u8 s_fmt, u8 num_ch, u32 s_rate, u8 dirn) +{ + struct nhlt_fmt *fmt; + struct nhlt_endpoint *epnt; + struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct device *dev = bus->dev; + struct nhlt_specific_cfg *sp_config; + struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + u16 bps = num_ch * s_fmt; + u8 j; + + dump_config(dev, instance, link_type, s_fmt, num_ch, s_rate, dirn, bps); + + epnt = (struct nhlt_endpoint *)nhlt->desc; + + dev_dbg(dev, "endpoint count =%d\n", nhlt->endpoint_count); + + for (j = 0; j < nhlt->endpoint_count; j++) { + if (skl_check_ep_match(dev, epnt, instance, link_type, dirn)) { + fmt = (struct nhlt_fmt *)(epnt->config.caps + + epnt->config.size); + sp_config = skl_get_specific_cfg(dev, fmt, num_ch, s_rate, bps); + if (sp_config) + return sp_config; + } + + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + + return NULL; +} diff --git a/sound/soc/intel/skylake/skl-nhlt.h b/sound/soc/intel/skylake/skl-nhlt.h new file mode 100644 index 0000000..3769f9f --- /dev/null +++ b/sound/soc/intel/skylake/skl-nhlt.h @@ -0,0 +1,106 @@ +/* + * skl-nhlt.h - Intel HDA Platform NHLT header + * + * Copyright (C) 2015 Intel Corp + * Author: Sanjiv Kumar <sanjiv.kumar@intel.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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ +#ifndef __SKL_NHLT_H__ +#define __SKL_NHLT_H__ + +#include <linux/acpi.h> + +struct wav_fmt { + u16 fmt_tag; + u16 channels; + u32 samples_per_sec; + u32 avg_bytes_per_sec; + u16 block_align; + u16 bits_per_sample; + u16 cb_size; +} __packed; + +struct wav_fmt_ext { + struct wav_fmt fmt; + union samples { + u16 valid_bits_per_sample; + u16 samples_per_block; + u16 reserved; + } sample; + u32 channel_mask; + u8 sub_fmt[16]; +} __packed; + +enum nhlt_link_type { + NHLT_LINK_HDA = 0, + NHLT_LINK_DSP = 1, + NHLT_LINK_DMIC = 2, + NHLT_LINK_SSP = 3, + NHLT_LINK_INVALID +}; + +enum nhlt_device_type { + NHLT_DEVICE_BT = 0, + NHLT_DEVICE_DMIC = 1, + NHLT_DEVICE_I2S = 4, + NHLT_DEVICE_INVALID +}; + +struct nhlt_specific_cfg { + u32 size; + u8 caps[0]; +} __packed; + +struct nhlt_fmt_cfg { + struct wav_fmt_ext fmt_ext; + struct nhlt_specific_cfg config; +} __packed; + +struct nhlt_fmt { + u8 fmt_count; + struct nhlt_fmt_cfg fmt_config[0]; +} __packed; + +struct nhlt_endpoint { + u32 length; + u8 linktype; + u8 instance_id; + u16 vendor_id; + u16 device_id; + u16 revision_id; + u32 subsystem_id; + u8 device_type; + u8 direction; + u8 virtual_bus_id; + struct nhlt_specific_cfg config; +} __packed; + +struct nhlt_acpi_table { + struct acpi_table_header header; + u8 endpoint_count; + struct nhlt_endpoint desc[0]; +} __packed; + +struct nhlt_resource_desc { + u32 extra; + u16 flags; + u64 addr_spc_gra; + u64 min_addr; + u64 max_addr; + u64 addr_trans_offset; + u64 length; +} __packed; + +#endif diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c new file mode 100644 index 0000000..7d617bf --- /dev/null +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -0,0 +1,916 @@ +/* + * skl-pcm.c -ASoC HDA Platform driver file implementing PCM functionality + * + * Copyright (C) 2014-2015 Intel Corp + * Author: Jeeja KP <jeeja.kp@intel.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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#include <linux/pci.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "skl.h" + +#define HDA_MONO 1 +#define HDA_STEREO 2 + +static struct snd_pcm_hardware azx_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_HAS_WALL_CLOCK | /* legacy */ + SNDRV_PCM_INFO_HAS_LINK_ATIME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = AZX_MAX_BUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = AZX_MAX_BUF_SIZE / 2, + .periods_min = 2, + .periods_max = AZX_MAX_FRAG, + .fifo_size = 0, +}; + +static inline +struct hdac_ext_stream *get_hdac_ext_stream(struct snd_pcm_substream *substream) +{ + return substream->runtime->private_data; +} + +static struct hdac_ext_bus *get_bus_ctx(struct snd_pcm_substream *substream) +{ + struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + struct hdac_stream *hstream = hdac_stream(stream); + struct hdac_bus *bus = hstream->bus; + + return hbus_to_ebus(bus); +} + +static int skl_substream_alloc_pages(struct hdac_ext_bus *ebus, + struct snd_pcm_substream *substream, + size_t size) +{ + struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + + hdac_stream(stream)->bufsize = 0; + hdac_stream(stream)->period_bytes = 0; + hdac_stream(stream)->format_val = 0; + + return snd_pcm_lib_malloc_pages(substream, size); +} + +static int skl_substream_free_pages(struct hdac_bus *bus, + struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static void skl_set_pcm_constrains(struct hdac_ext_bus *ebus, + struct snd_pcm_runtime *runtime) +{ + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + + /* avoid wrap-around with wall-clock */ + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, + 20, 178000000); +} + +static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_ext_bus *ebus) +{ + if (ebus->ppcap) + return HDAC_EXT_STREAM_TYPE_HOST; + else + return HDAC_EXT_STREAM_TYPE_COUPLED; +} + +static int skl_pcm_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_ext_stream *stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct skl_dma_params *dma_params; + int ret; + + dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); + ret = pm_runtime_get_sync(dai->dev); + if (ret) + return ret; + + stream = snd_hdac_ext_stream_assign(ebus, substream, + skl_get_host_stream_type(ebus)); + if (stream == NULL) + return -EBUSY; + + skl_set_pcm_constrains(ebus, runtime); + + /* + * disable WALLCLOCK timestamps for capture streams + * until we figure out how to handle digital inputs + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */ + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME; + } + + runtime->private_data = stream; + + dma_params = kzalloc(sizeof(*dma_params), GFP_KERNEL); + if (!dma_params) + return -ENOMEM; + + dma_params->stream_tag = hdac_stream(stream)->stream_tag; + snd_soc_dai_set_dma_data(dai, substream, dma_params); + + dev_dbg(dai->dev, "stream tag set in dma params=%d\n", + dma_params->stream_tag); + snd_pcm_set_sync(substream); + + return 0; +} + +static int skl_get_format(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct skl_dma_params *dma_params; + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + int format_val = 0; + + if (ebus->ppcap) { + struct snd_pcm_runtime *runtime = substream->runtime; + + format_val = snd_hdac_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + 32, 0); + } else { + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + dma_params = snd_soc_dai_get_dma_data(codec_dai, substream); + if (dma_params) + format_val = dma_params->format; + } + + return format_val; +} + +static int skl_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + unsigned int format_val; + int err; + + dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); + if (hdac_stream(stream)->prepared) { + dev_dbg(dai->dev, "already stream is prepared - returning\n"); + return 0; + } + + format_val = skl_get_format(substream, dai); + dev_dbg(dai->dev, "stream_tag=%d formatvalue=%d\n", + hdac_stream(stream)->stream_tag, format_val); + snd_hdac_stream_reset(hdac_stream(stream)); + + err = snd_hdac_stream_set_params(hdac_stream(stream), format_val); + if (err < 0) + return err; + + err = snd_hdac_stream_setup(hdac_stream(stream)); + if (err < 0) + return err; + + hdac_stream(stream)->prepared = 1; + + return err; +} + +static int skl_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int ret, dma_id; + + dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); + ret = skl_substream_alloc_pages(ebus, substream, + params_buffer_bytes(params)); + if (ret < 0) + return ret; + + dev_dbg(dai->dev, "format_val, rate=%d, ch=%d, format=%d\n", + runtime->rate, runtime->channels, runtime->format); + + dma_id = hdac_stream(stream)->stream_tag - 1; + dev_dbg(dai->dev, "dma_id=%d\n", dma_id); + + return 0; +} + +static void skl_pcm_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct skl_dma_params *dma_params = NULL; + + dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); + + snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(ebus)); + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + /* + * now we should set this to NULL as we are freeing by the + * dma_params + */ + snd_soc_dai_set_dma_data(dai, substream, NULL); + + pm_runtime_mark_last_busy(dai->dev); + pm_runtime_put_autosuspend(dai->dev); + kfree(dma_params); +} + +static int skl_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + + dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); + + snd_hdac_stream_cleanup(hdac_stream(stream)); + hdac_stream(stream)->prepared = 0; + + return skl_substream_free_pages(ebus_to_hbus(ebus), substream); +} + +static int skl_link_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_ext_stream *link_dev; + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct skl_dma_params *dma_params; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int dma_id; + + pr_debug("%s\n", __func__); + link_dev = snd_hdac_ext_stream_assign(ebus, substream, + HDAC_EXT_STREAM_TYPE_LINK); + if (!link_dev) + return -EBUSY; + + snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); + + /* set the stream tag in the codec dai dma params */ + dma_params = (struct skl_dma_params *) + snd_soc_dai_get_dma_data(codec_dai, substream); + if (dma_params) + dma_params->stream_tag = hdac_stream(link_dev)->stream_tag; + snd_soc_dai_set_dma_data(codec_dai, substream, (void *)dma_params); + dma_id = hdac_stream(link_dev)->stream_tag - 1; + + return 0; +} + +static int skl_link_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct hdac_ext_stream *link_dev = + snd_soc_dai_get_dma_data(dai, substream); + unsigned int format_val = 0; + struct skl_dma_params *dma_params; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_pcm_hw_params *params; + struct snd_interval *channels, *rate; + struct hdac_ext_link *link; + + dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); + if (link_dev->link_prepared) { + dev_dbg(dai->dev, "already stream is prepared - returning\n"); + return 0; + } + params = devm_kzalloc(dai->dev, sizeof(*params), GFP_KERNEL); + if (params == NULL) + return -ENOMEM; + + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + channels->min = channels->max = substream->runtime->channels; + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->min = rate->max = substream->runtime->rate; + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + substream->runtime->format); + + + dma_params = (struct skl_dma_params *) + snd_soc_dai_get_dma_data(codec_dai, substream); + if (dma_params) + format_val = dma_params->format; + dev_dbg(dai->dev, "stream_tag=%d formatvalue=%d codec_dai_name=%s\n", + hdac_stream(link_dev)->stream_tag, format_val, codec_dai->name); + + snd_hdac_ext_link_stream_reset(link_dev); + + snd_hdac_ext_link_stream_setup(link_dev, format_val); + + link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name); + if (!link) + return -EINVAL; + + snd_hdac_ext_link_set_stream_id(link, hdac_stream(link_dev)->stream_tag); + link_dev->link_prepared = 1; + + return 0; +} + +static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *link_dev = + snd_soc_dai_get_dma_data(dai, substream); + + dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + snd_hdac_ext_link_stream_start(link_dev); + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + snd_hdac_ext_link_stream_clear(link_dev); + break; + + default: + return -EINVAL; + } + return 0; +} + +static int skl_link_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct hdac_ext_stream *link_dev = + snd_soc_dai_get_dma_data(dai, substream); + struct hdac_ext_link *link; + + dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); + + link_dev->link_prepared = 0; + + link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name); + if (!link) + return -EINVAL; + + snd_hdac_ext_link_clear_stream_id(link, hdac_stream(link_dev)->stream_tag); + snd_hdac_ext_stream_release(link_dev, HDAC_EXT_STREAM_TYPE_LINK); + return 0; +} + +static int skl_hda_be_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return pm_runtime_get_sync(dai->dev); +} + +static void skl_hda_be_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + pm_runtime_mark_last_busy(dai->dev); + pm_runtime_put_autosuspend(dai->dev); +} + +static struct snd_soc_dai_ops skl_pcm_dai_ops = { + .startup = skl_pcm_open, + .shutdown = skl_pcm_close, + .prepare = skl_pcm_prepare, + .hw_params = skl_pcm_hw_params, + .hw_free = skl_pcm_hw_free, +}; + +static struct snd_soc_dai_ops skl_dmic_dai_ops = { + .startup = skl_hda_be_startup, + .shutdown = skl_hda_be_shutdown, +}; + +static struct snd_soc_dai_ops skl_link_dai_ops = { + .startup = skl_hda_be_startup, + .prepare = skl_link_pcm_prepare, + .hw_params = skl_link_hw_params, + .hw_free = skl_link_hw_free, + .trigger = skl_link_pcm_trigger, + .shutdown = skl_hda_be_shutdown, +}; + +static struct snd_soc_dai_driver skl_platform_dai[] = { +{ + .name = "System Pin", + .ops = &skl_pcm_dai_ops, + .playback = { + .stream_name = "System Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "System Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Reference Pin", + .ops = &skl_pcm_dai_ops, + .capture = { + .stream_name = "Reference Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Deepbuffer Pin", + .ops = &skl_pcm_dai_ops, + .playback = { + .stream_name = "Deepbuffer Playback", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "LowLatency Pin", + .ops = &skl_pcm_dai_ops, + .playback = { + .stream_name = "Low Latency Playback", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +/* BE CPU Dais */ +{ + .name = "iDisp Pin", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "iDisp Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "DMIC01 Pin", + .ops = &skl_dmic_dai_ops, + .capture = { + .stream_name = "DMIC01 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "DMIC23 Pin", + .ops = &skl_dmic_dai_ops, + .capture = { + .stream_name = "DMIC23 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "HD-Codec Pin", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "HD-Codec Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "HD-Codec Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "HD-Codec-SPK Pin", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "HD-Codec-SPK Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "HD-Codec-AMIC Pin", + .ops = &skl_link_dai_ops, + .capture = { + .stream_name = "HD-Codec-AMIC Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +}; + +static int skl_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai_link = rtd->dai_link; + + dev_dbg(rtd->cpu_dai->dev, "In %s:%s\n", __func__, + dai_link->cpu_dai_name); + + runtime = substream->runtime; + snd_soc_set_runtime_hwparams(substream, &azx_pcm_hw); + + return 0; +} + +static int skl_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_stream *stream; + struct snd_pcm_substream *s; + bool start; + int sbits = 0; + unsigned long cookie; + struct hdac_stream *hstr; + + stream = get_hdac_ext_stream(substream); + hstr = hdac_stream(stream); + + dev_dbg(bus->dev, "In %s cmd=%d\n", __func__, cmd); + + if (!hstr->prepared) + return -EPIPE; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + start = true; + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + start = false; + break; + + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + stream = get_hdac_ext_stream(s); + sbits |= 1 << hdac_stream(stream)->index; + snd_pcm_trigger_done(s, substream); + } + + spin_lock_irqsave(&bus->reg_lock, cookie); + + /* first, set SYNC bits of corresponding streams */ + snd_hdac_stream_sync_trigger(hstr, true, sbits, AZX_REG_SSYNC); + + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + stream = get_hdac_ext_stream(s); + if (start) + snd_hdac_stream_start(hdac_stream(stream), true); + else + snd_hdac_stream_stop(hdac_stream(stream)); + } + spin_unlock_irqrestore(&bus->reg_lock, cookie); + + snd_hdac_stream_sync(hstr, start, sbits); + + spin_lock_irqsave(&bus->reg_lock, cookie); + + /* reset SYNC bits */ + snd_hdac_stream_sync_trigger(hstr, false, sbits, AZX_REG_SSYNC); + if (start) + snd_hdac_stream_timecounter_init(hstr, sbits); + spin_unlock_irqrestore(&bus->reg_lock, cookie); + + return 0; +} + +static int skl_dsp_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct hdac_ext_bus *ebus = get_bus_ctx(substream); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct hdac_ext_stream *stream; + int start; + unsigned long cookie; + struct hdac_stream *hstr; + + dev_dbg(bus->dev, "In %s cmd=%d streamname=%s\n", __func__, cmd, cpu_dai->name); + + stream = get_hdac_ext_stream(substream); + hstr = hdac_stream(stream); + + if (!hstr->prepared) + return -EPIPE; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + start = 1; + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + start = 0; + break; + + default: + return -EINVAL; + } + + spin_lock_irqsave(&bus->reg_lock, cookie); + + if (start) + snd_hdac_stream_start(hdac_stream(stream), true); + else + snd_hdac_stream_stop(hdac_stream(stream)); + + if (start) + snd_hdac_stream_timecounter_init(hstr, 0); + + spin_unlock_irqrestore(&bus->reg_lock, cookie); + + return 0; +} +static int skl_platform_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct hdac_ext_bus *ebus = get_bus_ctx(substream); + + if (ebus->ppcap) + return skl_dsp_trigger(substream, cmd); + else + return skl_pcm_trigger(substream, cmd); +} + +/* calculate runtime delay from LPIB */ +static int skl_get_delay_from_lpib(struct hdac_ext_bus *ebus, + struct hdac_ext_stream *sstream, + unsigned int pos) +{ + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_stream *hstream = hdac_stream(sstream); + struct snd_pcm_substream *substream = hstream->substream; + int stream = substream->stream; + unsigned int lpib_pos = snd_hdac_stream_get_pos_lpib(hstream); + int delay; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = pos - lpib_pos; + else + delay = lpib_pos - pos; + + if (delay < 0) { + if (delay >= hstream->delay_negative_threshold) + delay = 0; + else + delay += hstream->bufsize; + } + + if (delay >= hstream->period_bytes) { + dev_info(bus->dev, + "Unstable LPIB (%d >= %d); disabling LPIB delay counting\n", + delay, hstream->period_bytes); + delay = 0; + } + + return bytes_to_frames(substream->runtime, delay); +} + +static unsigned int skl_get_position(struct hdac_ext_stream *hstream, + int codec_delay) +{ + struct hdac_stream *hstr = hdac_stream(hstream); + struct snd_pcm_substream *substream = hstr->substream; + struct hdac_ext_bus *ebus = get_bus_ctx(substream); + unsigned int pos; + int delay; + + /* use the position buffer as default */ + pos = snd_hdac_stream_get_pos_posbuf(hdac_stream(hstream)); + + if (pos >= hdac_stream(hstream)->bufsize) + pos = 0; + + if (substream->runtime) { + delay = skl_get_delay_from_lpib(ebus, hstream, pos) + + codec_delay; + substream->runtime->delay += delay; + } + + return pos; +} + +static snd_pcm_uframes_t skl_platform_pcm_pointer + (struct snd_pcm_substream *substream) +{ + struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream); + + return bytes_to_frames(substream->runtime, + skl_get_position(hstream, 0)); +} + +static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + u64 codec_frames, codec_nsecs; + + if (!codec_dai->driver->ops->delay) + return nsec; + + codec_frames = codec_dai->driver->ops->delay(substream, codec_dai); + codec_nsecs = div_u64(codec_frames * 1000000000LL, + substream->runtime->rate); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return nsec + codec_nsecs; + + return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; +} + +static int skl_get_time_info(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report) +{ + struct hdac_ext_stream *sstream = get_hdac_ext_stream(substream); + struct hdac_stream *hstr = hdac_stream(sstream); + u64 nsec; + + if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) && + (audio_tstamp_config->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK)) { + + snd_pcm_gettime(substream->runtime, system_ts); + + nsec = timecounter_read(&hstr->tc); + nsec = div_u64(nsec, 3); /* can be optimized */ + if (audio_tstamp_config->report_delay) + nsec = skl_adjust_codec_delay(substream, nsec); + + *audio_ts = ns_to_timespec(nsec); + + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; + audio_tstamp_report->accuracy_report = 1; /* rest of struct is valid */ + audio_tstamp_report->accuracy = 42; /* 24MHzWallClk == 42ns resolution */ + + } else { + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; + } + + return 0; +} + +static struct snd_pcm_ops skl_platform_ops = { + .open = skl_platform_open, + .ioctl = snd_pcm_lib_ioctl, + .trigger = skl_platform_pcm_trigger, + .pointer = skl_platform_pcm_pointer, + .get_time_info = skl_get_time_info, + .mmap = snd_pcm_lib_default_mmap, + .page = snd_pcm_sgbuf_ops_page, +}; + +static void skl_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +#define MAX_PREALLOC_SIZE (32 * 1024 * 1024) + +static int skl_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *dai = rtd->cpu_dai; + struct hdac_ext_bus *ebus = dev_get_drvdata(dai->dev); + struct snd_pcm *pcm = rtd->pcm; + unsigned int size; + int retval = 0; + struct skl *skl = ebus_to_skl(ebus); + + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { + /* buffer pre-allocation */ + size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; + if (size > MAX_PREALLOC_SIZE) + size = MAX_PREALLOC_SIZE; + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV_SG, + snd_dma_pci_data(skl->pci), + size, MAX_PREALLOC_SIZE); + if (retval) { + dev_err(dai->dev, "dma buffer allocationf fail\n"); + return retval; + } + } + + return retval; +} + +static struct snd_soc_platform_driver skl_platform_drv = { + .ops = &skl_platform_ops, + .pcm_new = skl_pcm_new, + .pcm_free = skl_pcm_free, +}; + +static const struct snd_soc_component_driver skl_component = { + .name = "pcm", +}; + +int skl_platform_register(struct device *dev) +{ + int ret; + + ret = snd_soc_register_platform(dev, &skl_platform_drv); + if (ret) { + dev_err(dev, "soc platform registration failed %d\n", ret); + return ret; + } + ret = snd_soc_register_component(dev, &skl_component, + skl_platform_dai, + ARRAY_SIZE(skl_platform_dai)); + if (ret) { + dev_err(dev, "soc component registration failed %d\n", ret); + snd_soc_unregister_platform(dev); + } + + return ret; + +} + +int skl_platform_unregister(struct device *dev) +{ + snd_soc_unregister_component(dev); + snd_soc_unregister_platform(dev); + return 0; +} diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c new file mode 100644 index 0000000..44748ba --- /dev/null +++ b/sound/soc/intel/skylake/skl-sst-cldma.c @@ -0,0 +1,327 @@ +/* + * skl-sst-cldma.c - Code Loader DMA handler + * + * Copyright (C) 2015, Intel Corporation. + * Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ + +#include <linux/device.h> +#include <linux/mm.h> +#include <linux/kthread.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" + +static void skl_cldma_int_enable(struct sst_dsp *ctx) +{ + sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPIC, + SKL_ADSPIC_CL_DMA, SKL_ADSPIC_CL_DMA); +} + +void skl_cldma_int_disable(struct sst_dsp *ctx) +{ + sst_dsp_shim_update_bits_unlocked(ctx, + SKL_ADSP_REG_ADSPIC, SKL_ADSPIC_CL_DMA, 0); +} + +/* Code loader helper APIs */ +static void skl_cldma_setup_bdle(struct sst_dsp *ctx, + struct snd_dma_buffer *dmab_data, + u32 **bdlp, int size, int with_ioc) +{ + u32 *bdl = *bdlp; + + ctx->cl_dev.frags = 0; + while (size > 0) { + phys_addr_t addr = virt_to_phys(dmab_data->area + + (ctx->cl_dev.frags * ctx->cl_dev.bufsize)); + + bdl[0] = cpu_to_le32(lower_32_bits(addr)); + bdl[1] = cpu_to_le32(upper_32_bits(addr)); + + bdl[2] = cpu_to_le32(ctx->cl_dev.bufsize); + + size -= ctx->cl_dev.bufsize; + bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01); + + bdl += 4; + ctx->cl_dev.frags++; + } +} + +/* + * Setup controller + * Configure the registers to update the dma buffer address and + * enable interrupts. + * Note: Using the channel 1 for transfer + */ +static void skl_cldma_setup_controller(struct sst_dsp *ctx, + struct snd_dma_buffer *dmab_bdl, unsigned int max_size, + u32 count) +{ + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPL, + CL_SD_BDLPLBA(dmab_bdl->addr)); + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPU, + CL_SD_BDLPUBA(dmab_bdl->addr)); + + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_CBL, max_size); + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_LVI, count - 1); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_IOCE_MASK, CL_SD_CTL_IOCE(1)); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_FEIE_MASK, CL_SD_CTL_FEIE(1)); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_DEIE_MASK, CL_SD_CTL_DEIE(1)); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_STRM_MASK, CL_SD_CTL_STRM(FW_CL_STREAM_NUMBER)); +} + +static void skl_cldma_setup_spb(struct sst_dsp *ctx, + unsigned int size, bool enable) +{ + if (enable) + sst_dsp_shim_update_bits_unlocked(ctx, + SKL_ADSP_REG_CL_SPBFIFO_SPBFCCTL, + CL_SPBFIFO_SPBFCCTL_SPIBE_MASK, + CL_SPBFIFO_SPBFCCTL_SPIBE(1)); + + sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_CL_SPBFIFO_SPIB, size); +} + +static void skl_cldma_cleanup_spb(struct sst_dsp *ctx) +{ + sst_dsp_shim_update_bits_unlocked(ctx, + SKL_ADSP_REG_CL_SPBFIFO_SPBFCCTL, + CL_SPBFIFO_SPBFCCTL_SPIBE_MASK, + CL_SPBFIFO_SPBFCCTL_SPIBE(0)); + + sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_CL_SPBFIFO_SPIB, 0); +} + +static void skl_cldma_trigger(struct sst_dsp *ctx, bool enable) +{ + if (enable) + sst_dsp_shim_update_bits_unlocked(ctx, + SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_RUN_MASK, CL_SD_CTL_RUN(1)); + else + sst_dsp_shim_update_bits_unlocked(ctx, + SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_RUN_MASK, CL_SD_CTL_RUN(0)); +} + +static void skl_cldma_cleanup(struct sst_dsp *ctx) +{ + skl_cldma_cleanup_spb(ctx); + + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_IOCE_MASK, CL_SD_CTL_IOCE(0)); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_FEIE_MASK, CL_SD_CTL_FEIE(0)); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_DEIE_MASK, CL_SD_CTL_DEIE(0)); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, + CL_SD_CTL_STRM_MASK, CL_SD_CTL_STRM(0)); + + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPL, CL_SD_BDLPLBA(0)); + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPU, 0); + + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_CBL, 0); + sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_LVI, 0); +} + +static int skl_cldma_wait_interruptible(struct sst_dsp *ctx) +{ + int ret = 0; + + if (!wait_event_timeout(ctx->cl_dev.wait_queue, + ctx->cl_dev.wait_condition, + msecs_to_jiffies(SKL_WAIT_TIMEOUT))) { + dev_err(ctx->dev, "%s: Wait timeout\n", __func__); + ret = -EIO; + goto cleanup; + } + + dev_dbg(ctx->dev, "%s: Event wake\n", __func__); + if (ctx->cl_dev.wake_status != SKL_CL_DMA_BUF_COMPLETE) { + dev_err(ctx->dev, "%s: DMA Error\n", __func__); + ret = -EIO; + } + +cleanup: + ctx->cl_dev.wake_status = SKL_CL_DMA_STATUS_NONE; + return ret; +} + +static void skl_cldma_stop(struct sst_dsp *ctx) +{ + ctx->cl_dev.ops.cl_trigger(ctx, false); +} + +static void skl_cldma_fill_buffer(struct sst_dsp *ctx, unsigned int size, + const void *curr_pos, bool intr_enable, bool trigger) +{ + dev_dbg(ctx->dev, "Size: %x, intr_enable: %d\n", size, intr_enable); + dev_dbg(ctx->dev, "buf_pos_index:%d, trigger:%d\n", + ctx->cl_dev.dma_buffer_offset, trigger); + dev_dbg(ctx->dev, "spib position: %d\n", ctx->cl_dev.curr_spib_pos); + + memcpy(ctx->cl_dev.dmab_data.area + ctx->cl_dev.dma_buffer_offset, + curr_pos, size); + + if (ctx->cl_dev.curr_spib_pos == ctx->cl_dev.bufsize) + ctx->cl_dev.dma_buffer_offset = 0; + else + ctx->cl_dev.dma_buffer_offset = ctx->cl_dev.curr_spib_pos; + + ctx->cl_dev.wait_condition = false; + + if (intr_enable) + skl_cldma_int_enable(ctx); + + ctx->cl_dev.ops.cl_setup_spb(ctx, ctx->cl_dev.curr_spib_pos, trigger); + if (trigger) + ctx->cl_dev.ops.cl_trigger(ctx, true); +} + +/* + * The CL dma doesn't have any way to update the transfer status until a BDL + * buffer is fully transferred + * + * So Copying is divided in two parts. + * 1. Interrupt on buffer done where the size to be transferred is more than + * ring buffer size. + * 2. Polling on fw register to identify if data left to transferred doesn't + * fill the ring buffer. Caller takes care of polling the required status + * register to identify the transfer status. + */ +static int +skl_cldma_copy_to_buf(struct sst_dsp *ctx, const void *bin, u32 total_size) +{ + int ret = 0; + bool start = true; + unsigned int excess_bytes; + u32 size; + unsigned int bytes_left = total_size; + const void *curr_pos = bin; + + if (total_size <= 0) + return -EINVAL; + + dev_dbg(ctx->dev, "%s: Total binary size: %u\n", __func__, bytes_left); + + while (bytes_left) { + if (bytes_left > ctx->cl_dev.bufsize) { + + /* + * dma transfers only till the write pointer as + * updated in spib + */ + if (ctx->cl_dev.curr_spib_pos == 0) + ctx->cl_dev.curr_spib_pos = ctx->cl_dev.bufsize; + + size = ctx->cl_dev.bufsize; + skl_cldma_fill_buffer(ctx, size, curr_pos, true, start); + + start = false; + ret = skl_cldma_wait_interruptible(ctx); + if (ret < 0) { + skl_cldma_stop(ctx); + return ret; + } + + } else { + skl_cldma_int_disable(ctx); + + if ((ctx->cl_dev.curr_spib_pos + bytes_left) + <= ctx->cl_dev.bufsize) { + ctx->cl_dev.curr_spib_pos += bytes_left; + } else { + excess_bytes = bytes_left - + (ctx->cl_dev.bufsize - + ctx->cl_dev.curr_spib_pos); + ctx->cl_dev.curr_spib_pos = excess_bytes; + } + + size = bytes_left; + skl_cldma_fill_buffer(ctx, size, + curr_pos, false, start); + } + bytes_left -= size; + curr_pos = curr_pos + size; + } + + return ret; +} + +void skl_cldma_process_intr(struct sst_dsp *ctx) +{ + u8 cl_dma_intr_status; + + cl_dma_intr_status = + sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_CL_SD_STS); + + if (!(cl_dma_intr_status & SKL_CL_DMA_SD_INT_COMPLETE)) + ctx->cl_dev.wake_status = SKL_CL_DMA_ERR; + else + ctx->cl_dev.wake_status = SKL_CL_DMA_BUF_COMPLETE; + + ctx->cl_dev.wait_condition = true; + wake_up(&ctx->cl_dev.wait_queue); +} + +int skl_cldma_prepare(struct sst_dsp *ctx) +{ + int ret; + u32 *bdl; + + ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE; + + /* Allocate cl ops */ + ctx->cl_dev.ops.cl_setup_bdle = skl_cldma_setup_bdle; + ctx->cl_dev.ops.cl_setup_controller = skl_cldma_setup_controller; + ctx->cl_dev.ops.cl_setup_spb = skl_cldma_setup_spb; + ctx->cl_dev.ops.cl_cleanup_spb = skl_cldma_cleanup_spb; + ctx->cl_dev.ops.cl_trigger = skl_cldma_trigger; + ctx->cl_dev.ops.cl_cleanup_controller = skl_cldma_cleanup; + ctx->cl_dev.ops.cl_copy_to_dmabuf = skl_cldma_copy_to_buf; + ctx->cl_dev.ops.cl_stop_dma = skl_cldma_stop; + + /* Allocate buffer*/ + ret = ctx->dsp_ops.alloc_dma_buf(ctx->dev, + &ctx->cl_dev.dmab_data, ctx->cl_dev.bufsize); + if (ret < 0) { + dev_err(ctx->dev, "Alloc buffer for base fw failed: %x", ret); + return ret; + } + /* Setup Code loader BDL */ + ret = ctx->dsp_ops.alloc_dma_buf(ctx->dev, + &ctx->cl_dev.dmab_bdl, PAGE_SIZE); + if (ret < 0) { + dev_err(ctx->dev, "Alloc buffer for blde failed: %x", ret); + ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); + return ret; + } + bdl = (u32 *)ctx->cl_dev.dmab_bdl.area; + + /* Allocate BDLs */ + ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data, + &bdl, ctx->cl_dev.bufsize, 1); + ctx->cl_dev.ops.cl_setup_controller(ctx, &ctx->cl_dev.dmab_bdl, + ctx->cl_dev.bufsize, ctx->cl_dev.frags); + + ctx->cl_dev.curr_spib_pos = 0; + ctx->cl_dev.dma_buffer_offset = 0; + init_waitqueue_head(&ctx->cl_dev.wait_queue); + + return ret; +} diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h new file mode 100644 index 0000000..99e4c86 --- /dev/null +++ b/sound/soc/intel/skylake/skl-sst-cldma.h @@ -0,0 +1,251 @@ +/* + * Intel Code Loader DMA support + * + * Copyright (C) 2015, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ + +#ifndef SKL_SST_CLDMA_H_ +#define SKL_SST_CLDMA_H_ + +#define FW_CL_STREAM_NUMBER 0x1 + +#define DMA_ADDRESS_128_BITS_ALIGNMENT 7 +#define BDL_ALIGN(x) (x >> DMA_ADDRESS_128_BITS_ALIGNMENT) + +#define SKL_ADSPIC_CL_DMA 0x2 +#define SKL_ADSPIS_CL_DMA 0x2 +#define SKL_CL_DMA_SD_INT_DESC_ERR 0x10 /* Descriptor error interrupt */ +#define SKL_CL_DMA_SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ +#define SKL_CL_DMA_SD_INT_COMPLETE 0x04 /* Buffer completion interrupt */ + +/* Intel HD Audio Code Loader DMA Registers */ + +#define HDA_ADSP_LOADER_BASE 0x80 + +/* Stream Registers */ +#define SKL_ADSP_REG_CL_SD_CTL (HDA_ADSP_LOADER_BASE + 0x00) +#define SKL_ADSP_REG_CL_SD_STS (HDA_ADSP_LOADER_BASE + 0x03) +#define SKL_ADSP_REG_CL_SD_LPIB (HDA_ADSP_LOADER_BASE + 0x04) +#define SKL_ADSP_REG_CL_SD_CBL (HDA_ADSP_LOADER_BASE + 0x08) +#define SKL_ADSP_REG_CL_SD_LVI (HDA_ADSP_LOADER_BASE + 0x0c) +#define SKL_ADSP_REG_CL_SD_FIFOW (HDA_ADSP_LOADER_BASE + 0x0e) +#define SKL_ADSP_REG_CL_SD_FIFOSIZE (HDA_ADSP_LOADER_BASE + 0x10) +#define SKL_ADSP_REG_CL_SD_FORMAT (HDA_ADSP_LOADER_BASE + 0x12) +#define SKL_ADSP_REG_CL_SD_FIFOL (HDA_ADSP_LOADER_BASE + 0x14) +#define SKL_ADSP_REG_CL_SD_BDLPL (HDA_ADSP_LOADER_BASE + 0x18) +#define SKL_ADSP_REG_CL_SD_BDLPU (HDA_ADSP_LOADER_BASE + 0x1c) + +/* CL: Software Position Based FIFO Capability Registers */ +#define SKL_ADSP_REG_CL_SPBFIFO (HDA_ADSP_LOADER_BASE + 0x20) +#define SKL_ADSP_REG_CL_SPBFIFO_SPBFCH (SKL_ADSP_REG_CL_SPBFIFO + 0x0) +#define SKL_ADSP_REG_CL_SPBFIFO_SPBFCCTL (SKL_ADSP_REG_CL_SPBFIFO + 0x4) +#define SKL_ADSP_REG_CL_SPBFIFO_SPIB (SKL_ADSP_REG_CL_SPBFIFO + 0x8) +#define SKL_ADSP_REG_CL_SPBFIFO_MAXFIFOS (SKL_ADSP_REG_CL_SPBFIFO + 0xc) + +/* CL: Stream Descriptor x Control */ + +/* Stream Reset */ +#define CL_SD_CTL_SRST_SHIFT 0 +#define CL_SD_CTL_SRST_MASK (1 << CL_SD_CTL_SRST_SHIFT) +#define CL_SD_CTL_SRST(x) \ + ((x << CL_SD_CTL_SRST_SHIFT) & CL_SD_CTL_SRST_MASK) + +/* Stream Run */ +#define CL_SD_CTL_RUN_SHIFT 1 +#define CL_SD_CTL_RUN_MASK (1 << CL_SD_CTL_RUN_SHIFT) +#define CL_SD_CTL_RUN(x) \ + ((x << CL_SD_CTL_RUN_SHIFT) & CL_SD_CTL_RUN_MASK) + +/* Interrupt On Completion Enable */ +#define CL_SD_CTL_IOCE_SHIFT 2 +#define CL_SD_CTL_IOCE_MASK (1 << CL_SD_CTL_IOCE_SHIFT) +#define CL_SD_CTL_IOCE(x) \ + ((x << CL_SD_CTL_IOCE_SHIFT) & CL_SD_CTL_IOCE_MASK) + +/* FIFO Error Interrupt Enable */ +#define CL_SD_CTL_FEIE_SHIFT 3 +#define CL_SD_CTL_FEIE_MASK (1 << CL_SD_CTL_FEIE_SHIFT) +#define CL_SD_CTL_FEIE(x) \ + ((x << CL_SD_CTL_FEIE_SHIFT) & CL_SD_CTL_FEIE_MASK) + +/* Descriptor Error Interrupt Enable */ +#define CL_SD_CTL_DEIE_SHIFT 4 +#define CL_SD_CTL_DEIE_MASK (1 << CL_SD_CTL_DEIE_SHIFT) +#define CL_SD_CTL_DEIE(x) \ + ((x << CL_SD_CTL_DEIE_SHIFT) & CL_SD_CTL_DEIE_MASK) + +/* FIFO Limit Change */ +#define CL_SD_CTL_FIFOLC_SHIFT 5 +#define CL_SD_CTL_FIFOLC_MASK (1 << CL_SD_CTL_FIFOLC_SHIFT) +#define CL_SD_CTL_FIFOLC(x) \ + ((x << CL_SD_CTL_FIFOLC_SHIFT) & CL_SD_CTL_FIFOLC_MASK) + +/* Stripe Control */ +#define CL_SD_CTL_STRIPE_SHIFT 16 +#define CL_SD_CTL_STRIPE_MASK (0x3 << CL_SD_CTL_STRIPE_SHIFT) +#define CL_SD_CTL_STRIPE(x) \ + ((x << CL_SD_CTL_STRIPE_SHIFT) & CL_SD_CTL_STRIPE_MASK) + +/* Traffic Priority */ +#define CL_SD_CTL_TP_SHIFT 18 +#define CL_SD_CTL_TP_MASK (1 << CL_SD_CTL_TP_SHIFT) +#define CL_SD_CTL_TP(x) \ + ((x << CL_SD_CTL_TP_SHIFT) & CL_SD_CTL_TP_MASK) + +/* Bidirectional Direction Control */ +#define CL_SD_CTL_DIR_SHIFT 19 +#define CL_SD_CTL_DIR_MASK (1 << CL_SD_CTL_DIR_SHIFT) +#define CL_SD_CTL_DIR(x) \ + ((x << CL_SD_CTL_DIR_SHIFT) & CL_SD_CTL_DIR_MASK) + +/* Stream Number */ +#define CL_SD_CTL_STRM_SHIFT 20 +#define CL_SD_CTL_STRM_MASK (0xf << CL_SD_CTL_STRM_SHIFT) +#define CL_SD_CTL_STRM(x) \ + ((x << CL_SD_CTL_STRM_SHIFT) & CL_SD_CTL_STRM_MASK) + +/* CL: Stream Descriptor x Status */ + +/* Buffer Completion Interrupt Status */ +#define CL_SD_STS_BCIS(x) CL_SD_CTL_IOCE(x) + +/* FIFO Error */ +#define CL_SD_STS_FIFOE(x) CL_SD_CTL_FEIE(x) + +/* Descriptor Error */ +#define CL_SD_STS_DESE(x) CL_SD_CTL_DEIE(x) + +/* FIFO Ready */ +#define CL_SD_STS_FIFORDY(x) CL_SD_CTL_FIFOLC(x) + + +/* CL: Stream Descriptor x Last Valid Index */ +#define CL_SD_LVI_SHIFT 0 +#define CL_SD_LVI_MASK (0xff << CL_SD_LVI_SHIFT) +#define CL_SD_LVI(x) ((x << CL_SD_LVI_SHIFT) & CL_SD_LVI_MASK) + +/* CL: Stream Descriptor x FIFO Eviction Watermark */ +#define CL_SD_FIFOW_SHIFT 0 +#define CL_SD_FIFOW_MASK (0x7 << CL_SD_FIFOW_SHIFT) +#define CL_SD_FIFOW(x) \ + ((x << CL_SD_FIFOW_SHIFT) & CL_SD_FIFOW_MASK) + +/* CL: Stream Descriptor x Buffer Descriptor List Pointer Lower Base Address */ + +/* Protect Bits */ +#define CL_SD_BDLPLBA_PROT_SHIFT 0 +#define CL_SD_BDLPLBA_PROT_MASK (1 << CL_SD_BDLPLBA_PROT_SHIFT) +#define CL_SD_BDLPLBA_PROT(x) \ + ((x << CL_SD_BDLPLBA_PROT_SHIFT) & CL_SD_BDLPLBA_PROT_MASK) + +/* Buffer Descriptor List Lower Base Address */ +#define CL_SD_BDLPLBA_SHIFT 7 +#define CL_SD_BDLPLBA_MASK (0x1ffffff << CL_SD_BDLPLBA_SHIFT) +#define CL_SD_BDLPLBA(x) \ + ((BDL_ALIGN(lower_32_bits(x)) << CL_SD_BDLPLBA_SHIFT) & CL_SD_BDLPLBA_MASK) + +/* Buffer Descriptor List Upper Base Address */ +#define CL_SD_BDLPUBA_SHIFT 0 +#define CL_SD_BDLPUBA_MASK (0xffffffff << CL_SD_BDLPUBA_SHIFT) +#define CL_SD_BDLPUBA(x) \ + ((upper_32_bits(x) << CL_SD_BDLPUBA_SHIFT) & CL_SD_BDLPUBA_MASK) + +/* + * Code Loader - Software Position Based FIFO + * Capability Registers x Software Position Based FIFO Header + */ + +/* Next Capability Pointer */ +#define CL_SPBFIFO_SPBFCH_PTR_SHIFT 0 +#define CL_SPBFIFO_SPBFCH_PTR_MASK (0xff << CL_SPBFIFO_SPBFCH_PTR_SHIFT) +#define CL_SPBFIFO_SPBFCH_PTR(x) \ + ((x << CL_SPBFIFO_SPBFCH_PTR_SHIFT) & CL_SPBFIFO_SPBFCH_PTR_MASK) + +/* Capability Identifier */ +#define CL_SPBFIFO_SPBFCH_ID_SHIFT 16 +#define CL_SPBFIFO_SPBFCH_ID_MASK (0xfff << CL_SPBFIFO_SPBFCH_ID_SHIFT) +#define CL_SPBFIFO_SPBFCH_ID(x) \ + ((x << CL_SPBFIFO_SPBFCH_ID_SHIFT) & CL_SPBFIFO_SPBFCH_ID_MASK) + +/* Capability Version */ +#define CL_SPBFIFO_SPBFCH_VER_SHIFT 28 +#define CL_SPBFIFO_SPBFCH_VER_MASK (0xf << CL_SPBFIFO_SPBFCH_VER_SHIFT) +#define CL_SPBFIFO_SPBFCH_VER(x) \ + ((x << CL_SPBFIFO_SPBFCH_VER_SHIFT) & CL_SPBFIFO_SPBFCH_VER_MASK) + +/* Software Position in Buffer Enable */ +#define CL_SPBFIFO_SPBFCCTL_SPIBE_SHIFT 0 +#define CL_SPBFIFO_SPBFCCTL_SPIBE_MASK (1 << CL_SPBFIFO_SPBFCCTL_SPIBE_SHIFT) +#define CL_SPBFIFO_SPBFCCTL_SPIBE(x) \ + ((x << CL_SPBFIFO_SPBFCCTL_SPIBE_SHIFT) & CL_SPBFIFO_SPBFCCTL_SPIBE_MASK) + +/* SST IPC SKL defines */ +#define SKL_WAIT_TIMEOUT 500 /* 500 msec */ +#define SKL_MAX_BUFFER_SIZE (32 * PAGE_SIZE) + +enum skl_cl_dma_wake_states { + SKL_CL_DMA_STATUS_NONE = 0, + SKL_CL_DMA_BUF_COMPLETE, + SKL_CL_DMA_ERR, /* TODO: Expand the error states */ +}; + +struct sst_dsp; + +struct skl_cl_dev_ops { + void (*cl_setup_bdle)(struct sst_dsp *ctx, + struct snd_dma_buffer *dmab_data, + u32 **bdlp, int size, int with_ioc); + void (*cl_setup_controller)(struct sst_dsp *ctx, + struct snd_dma_buffer *dmab_bdl, + unsigned int max_size, u32 page_count); + void (*cl_setup_spb)(struct sst_dsp *ctx, + unsigned int size, bool enable); + void (*cl_cleanup_spb)(struct sst_dsp *ctx); + void (*cl_trigger)(struct sst_dsp *ctx, bool enable); + void (*cl_cleanup_controller)(struct sst_dsp *ctx); + int (*cl_copy_to_dmabuf)(struct sst_dsp *ctx, + const void *bin, u32 size); + void (*cl_stop_dma)(struct sst_dsp *ctx); +}; + +/** + * skl_cl_dev - holds information for code loader dma transfer + * + * @dmab_data: buffer pointer + * @dmab_bdl: buffer descriptor list + * @bufsize: ring buffer size + * @frags: Last valid buffer descriptor index in the BDL + * @curr_spib_pos: Current position in ring buffer + * @dma_buffer_offset: dma buffer offset + * @ops: operations supported on CL dma + * @wait_queue: wait queue to wake for wake event + * @wake_status: DMA wake status + * @wait_condition: condition to wait on wait queue + * @cl_dma_lock: for synchronized access to cldma + */ +struct skl_cl_dev { + struct snd_dma_buffer dmab_data; + struct snd_dma_buffer dmab_bdl; + + unsigned int bufsize; + unsigned int frags; + + unsigned int curr_spib_pos; + unsigned int dma_buffer_offset; + struct skl_cl_dev_ops ops; + + wait_queue_head_t wait_queue; + int wake_status; + bool wait_condition; +}; + +#endif /* SKL_SST_CLDMA_H_ */ diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c new file mode 100644 index 0000000..94875b0 --- /dev/null +++ b/sound/soc/intel/skylake/skl-sst-dsp.c @@ -0,0 +1,342 @@ +/* + * skl-sst-dsp.c - SKL SST library generic function + * + * Copyright (C) 2014-15, Intel Corporation. + * Author:Rafal Redzimski <rafal.f.redzimski@intel.com> + * Jeeja KP <jeeja.kp@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ +#include <sound/pcm.h> + +#include "../common/sst-dsp.h" +#include "../common/sst-ipc.h" +#include "../common/sst-dsp-priv.h" +#include "skl-sst-ipc.h" + +/* various timeout values */ +#define SKL_DSP_PU_TO 50 +#define SKL_DSP_PD_TO 50 +#define SKL_DSP_RESET_TO 50 + +void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state) +{ + mutex_lock(&ctx->mutex); + ctx->sst_state = state; + mutex_unlock(&ctx->mutex); +} + +static int skl_dsp_core_set_reset_state(struct sst_dsp *ctx) +{ + int ret; + + /* update bits */ + sst_dsp_shim_update_bits_unlocked(ctx, + SKL_ADSP_REG_ADSPCS, SKL_ADSPCS_CRST_MASK, + SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)); + + /* poll with timeout to check if operation successful */ + ret = sst_dsp_register_poll(ctx, + SKL_ADSP_REG_ADSPCS, + SKL_ADSPCS_CRST_MASK, + SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK), + SKL_DSP_RESET_TO, + "Set reset"); + if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & + SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) != + SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) { + dev_err(ctx->dev, "Set reset state failed\n"); + ret = -EIO; + } + + return ret; +} + +static int skl_dsp_core_unset_reset_state(struct sst_dsp *ctx) +{ + int ret; + + dev_dbg(ctx->dev, "In %s\n", __func__); + + /* update bits */ + sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, + SKL_ADSPCS_CRST_MASK, 0); + + /* poll with timeout to check if operation successful */ + ret = sst_dsp_register_poll(ctx, + SKL_ADSP_REG_ADSPCS, + SKL_ADSPCS_CRST_MASK, + 0, + SKL_DSP_RESET_TO, + "Unset reset"); + + if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & + SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) != 0) { + dev_err(ctx->dev, "Unset reset state failed\n"); + ret = -EIO; + } + + return ret; +} + +static bool is_skl_dsp_core_enable(struct sst_dsp *ctx) +{ + int val; + bool is_enable; + + val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS); + + is_enable = ((val & SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK)) && + (val & SKL_ADSPCS_SPA(SKL_DSP_CORES_MASK)) && + !(val & SKL_ADSPCS_CRST(SKL_DSP_CORES_MASK)) && + !(val & SKL_ADSPCS_CSTALL(SKL_DSP_CORES_MASK))); + + dev_dbg(ctx->dev, "DSP core is enabled=%d\n", is_enable); + return is_enable; +} + +static int skl_dsp_reset_core(struct sst_dsp *ctx) +{ + /* stall core */ + sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_ADSPCS, + sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & + SKL_ADSPCS_CSTALL(SKL_DSP_CORES_MASK)); + + /* set reset state */ + return skl_dsp_core_set_reset_state(ctx); +} + +static int skl_dsp_start_core(struct sst_dsp *ctx) +{ + int ret; + + /* unset reset state */ + ret = skl_dsp_core_unset_reset_state(ctx); + if (ret < 0) { + dev_dbg(ctx->dev, "dsp unset reset fails\n"); + return ret; + } + + /* run core */ + dev_dbg(ctx->dev, "run core...\n"); + sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_ADSPCS, + sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & + ~SKL_ADSPCS_CSTALL(SKL_DSP_CORES_MASK)); + + if (!is_skl_dsp_core_enable(ctx)) { + skl_dsp_reset_core(ctx); + dev_err(ctx->dev, "DSP core enable failed\n"); + ret = -EIO; + } + + return ret; +} + +static int skl_dsp_core_power_up(struct sst_dsp *ctx) +{ + int ret; + + /* update bits */ + sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, + SKL_ADSPCS_SPA_MASK, SKL_ADSPCS_SPA(SKL_DSP_CORES_MASK)); + + /* poll with timeout to check if operation successful */ + ret = sst_dsp_register_poll(ctx, + SKL_ADSP_REG_ADSPCS, + SKL_ADSPCS_CPA_MASK, + SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK), + SKL_DSP_PU_TO, + "Power up"); + + if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & + SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK)) != + SKL_ADSPCS_CPA(SKL_DSP_CORES_MASK)) { + dev_err(ctx->dev, "DSP core power up failed\n"); + ret = -EIO; + } + + return ret; +} + +static int skl_dsp_core_power_down(struct sst_dsp *ctx) +{ + /* update bits */ + sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, + SKL_ADSPCS_SPA_MASK, 0); + + /* poll with timeout to check if operation successful */ + return sst_dsp_register_poll(ctx, + SKL_ADSP_REG_ADSPCS, + SKL_ADSPCS_SPA_MASK, + 0, + SKL_DSP_PD_TO, + "Power down"); +} + +static int skl_dsp_enable_core(struct sst_dsp *ctx) +{ + int ret; + + /* power up */ + ret = skl_dsp_core_power_up(ctx); + if (ret < 0) { + dev_dbg(ctx->dev, "dsp core power up failed\n"); + return ret; + } + + return skl_dsp_start_core(ctx); +} + +int skl_dsp_disable_core(struct sst_dsp *ctx) +{ + int ret; + + ret = skl_dsp_reset_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "dsp core reset failed\n"); + return ret; + } + + /* power down core*/ + ret = skl_dsp_core_power_down(ctx); + if (ret < 0) { + dev_err(ctx->dev, "dsp core power down failed\n"); + return ret; + } + + if (is_skl_dsp_core_enable(ctx)) { + dev_err(ctx->dev, "DSP core disable failed\n"); + ret = -EIO; + } + + return ret; +} + +int skl_dsp_boot(struct sst_dsp *ctx) +{ + int ret; + + if (is_skl_dsp_core_enable(ctx)) { + dev_dbg(ctx->dev, "dsp core is already enabled, so reset the dap core\n"); + ret = skl_dsp_reset_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "dsp reset failed\n"); + return ret; + } + + ret = skl_dsp_start_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "dsp start failed\n"); + return ret; + } + } else { + dev_dbg(ctx->dev, "disable and enable to make sure DSP is invalid state\n"); + ret = skl_dsp_disable_core(ctx); + + if (ret < 0) { + dev_err(ctx->dev, "dsp disable core failes\n"); + return ret; + } + ret = skl_dsp_enable_core(ctx); + } + + return ret; +} + +irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id) +{ + struct sst_dsp *ctx = dev_id; + u32 val; + irqreturn_t result = IRQ_NONE; + + spin_lock(&ctx->spinlock); + + val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPIS); + ctx->intr_status = val; + + if (val & SKL_ADSPIS_IPC) { + skl_ipc_int_disable(ctx); + result = IRQ_WAKE_THREAD; + } + + if (val & SKL_ADSPIS_CL_DMA) { + skl_cldma_int_disable(ctx); + result = IRQ_WAKE_THREAD; + } + + spin_unlock(&ctx->spinlock); + + return result; +} + +int skl_dsp_wake(struct sst_dsp *ctx) +{ + return ctx->fw_ops.set_state_D0(ctx); +} +EXPORT_SYMBOL_GPL(skl_dsp_wake); + +int skl_dsp_sleep(struct sst_dsp *ctx) +{ + return ctx->fw_ops.set_state_D3(ctx); +} +EXPORT_SYMBOL_GPL(skl_dsp_sleep); + +struct sst_dsp *skl_dsp_ctx_init(struct device *dev, + struct sst_dsp_device *sst_dev, int irq) +{ + int ret; + struct sst_dsp *sst; + + sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); + if (sst == NULL) + return NULL; + + spin_lock_init(&sst->spinlock); + mutex_init(&sst->mutex); + sst->dev = dev; + sst->sst_dev = sst_dev; + sst->irq = irq; + sst->ops = sst_dev->ops; + sst->thread_context = sst_dev->thread_context; + + /* Initialise SST Audio DSP */ + if (sst->ops->init) { + ret = sst->ops->init(sst, NULL); + if (ret < 0) + return NULL; + } + + /* Register the ISR */ + ret = request_threaded_irq(sst->irq, sst->ops->irq_handler, + sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); + if (ret) { + dev_err(sst->dev, "unable to grab threaded IRQ %d, disabling device\n", + sst->irq); + return NULL; + } + + return sst; +} + +void skl_dsp_free(struct sst_dsp *dsp) +{ + skl_ipc_int_disable(dsp); + + free_irq(dsp->irq, dsp); + skl_dsp_disable_core(dsp); +} +EXPORT_SYMBOL_GPL(skl_dsp_free); + +bool is_skl_dsp_running(struct sst_dsp *ctx) +{ + return (ctx->sst_state == SKL_DSP_RUNNING); +} +EXPORT_SYMBOL_GPL(is_skl_dsp_running); diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h new file mode 100644 index 0000000..6bfcef4 --- /dev/null +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -0,0 +1,145 @@ +/* + * Skylake SST DSP Support + * + * Copyright (C) 2014-15, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ + +#ifndef __SKL_SST_DSP_H__ +#define __SKL_SST_DSP_H__ + +#include <linux/interrupt.h> +#include <sound/memalloc.h> +#include "skl-sst-cldma.h" + +struct sst_dsp; +struct skl_sst; +struct sst_dsp_device; + +/* Intel HD Audio General DSP Registers */ +#define SKL_ADSP_GEN_BASE 0x0 +#define SKL_ADSP_REG_ADSPCS (SKL_ADSP_GEN_BASE + 0x04) +#define SKL_ADSP_REG_ADSPIC (SKL_ADSP_GEN_BASE + 0x08) +#define SKL_ADSP_REG_ADSPIS (SKL_ADSP_GEN_BASE + 0x0C) +#define SKL_ADSP_REG_ADSPIC2 (SKL_ADSP_GEN_BASE + 0x10) +#define SKL_ADSP_REG_ADSPIS2 (SKL_ADSP_GEN_BASE + 0x14) + +/* Intel HD Audio Inter-Processor Communication Registers */ +#define SKL_ADSP_IPC_BASE 0x40 +#define SKL_ADSP_REG_HIPCT (SKL_ADSP_IPC_BASE + 0x00) +#define SKL_ADSP_REG_HIPCTE (SKL_ADSP_IPC_BASE + 0x04) +#define SKL_ADSP_REG_HIPCI (SKL_ADSP_IPC_BASE + 0x08) +#define SKL_ADSP_REG_HIPCIE (SKL_ADSP_IPC_BASE + 0x0C) +#define SKL_ADSP_REG_HIPCCTL (SKL_ADSP_IPC_BASE + 0x10) + +/* HIPCI */ +#define SKL_ADSP_REG_HIPCI_BUSY BIT(31) + +/* HIPCIE */ +#define SKL_ADSP_REG_HIPCIE_DONE BIT(30) + +/* HIPCCTL */ +#define SKL_ADSP_REG_HIPCCTL_DONE BIT(1) +#define SKL_ADSP_REG_HIPCCTL_BUSY BIT(0) + +/* HIPCT */ +#define SKL_ADSP_REG_HIPCT_BUSY BIT(31) + +/* Intel HD Audio SRAM Window 1 */ +#define SKL_ADSP_SRAM1_BASE 0xA000 + +#define SKL_ADSP_MMIO_LEN 0x10000 + +#define SKL_ADSP_W0_STAT_SZ 0x800 + +#define SKL_ADSP_W0_UP_SZ 0x800 + +#define SKL_ADSP_W1_SZ 0x1000 + +#define SKL_FW_STS_MASK 0xf + +#define SKL_FW_INIT 0x1 +#define SKL_FW_RFW_START 0xf + +#define SKL_ADSPIC_IPC 1 +#define SKL_ADSPIS_IPC 1 + +/* ADSPCS - Audio DSP Control & Status */ +#define SKL_DSP_CORES 1 +#define SKL_DSP_CORE0_MASK 1 +#define SKL_DSP_CORES_MASK ((1 << SKL_DSP_CORES) - 1) + +/* Core Reset - asserted high */ +#define SKL_ADSPCS_CRST_SHIFT 0 +#define SKL_ADSPCS_CRST_MASK (SKL_DSP_CORES_MASK << SKL_ADSPCS_CRST_SHIFT) +#define SKL_ADSPCS_CRST(x) ((x << SKL_ADSPCS_CRST_SHIFT) & SKL_ADSPCS_CRST_MASK) + +/* Core run/stall - when set to '1' core is stalled */ +#define SKL_ADSPCS_CSTALL_SHIFT 8 +#define SKL_ADSPCS_CSTALL_MASK (SKL_DSP_CORES_MASK << \ + SKL_ADSPCS_CSTALL_SHIFT) +#define SKL_ADSPCS_CSTALL(x) ((x << SKL_ADSPCS_CSTALL_SHIFT) & \ + SKL_ADSPCS_CSTALL_MASK) + +/* Set Power Active - when set to '1' turn cores on */ +#define SKL_ADSPCS_SPA_SHIFT 16 +#define SKL_ADSPCS_SPA_MASK (SKL_DSP_CORES_MASK << SKL_ADSPCS_SPA_SHIFT) +#define SKL_ADSPCS_SPA(x) ((x << SKL_ADSPCS_SPA_SHIFT) & SKL_ADSPCS_SPA_MASK) + +/* Current Power Active - power status of cores, set by hardware */ +#define SKL_ADSPCS_CPA_SHIFT 24 +#define SKL_ADSPCS_CPA_MASK (SKL_DSP_CORES_MASK << SKL_ADSPCS_CPA_SHIFT) +#define SKL_ADSPCS_CPA(x) ((x << SKL_ADSPCS_CPA_SHIFT) & SKL_ADSPCS_CPA_MASK) + +#define SST_DSP_POWER_D0 0x0 /* full On */ +#define SST_DSP_POWER_D3 0x3 /* Off */ + +enum skl_dsp_states { + SKL_DSP_RUNNING = 1, + SKL_DSP_RESET, +}; + +struct skl_dsp_fw_ops { + int (*load_fw)(struct sst_dsp *ctx); + /* FW module parser/loader */ + int (*parse_fw)(struct sst_dsp *ctx); + int (*set_state_D0)(struct sst_dsp *ctx); + int (*set_state_D3)(struct sst_dsp *ctx); + unsigned int (*get_fw_errcode)(struct sst_dsp *ctx); +}; + +struct skl_dsp_loader_ops { + int (*alloc_dma_buf)(struct device *dev, + struct snd_dma_buffer *dmab, size_t size); + int (*free_dma_buf)(struct device *dev, + struct snd_dma_buffer *dmab); +}; + +void skl_cldma_process_intr(struct sst_dsp *ctx); +void skl_cldma_int_disable(struct sst_dsp *ctx); +int skl_cldma_prepare(struct sst_dsp *ctx); + +void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state); +struct sst_dsp *skl_dsp_ctx_init(struct device *dev, + struct sst_dsp_device *sst_dev, int irq); +int skl_dsp_disable_core(struct sst_dsp *ctx); +bool is_skl_dsp_running(struct sst_dsp *ctx); +irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id); +int skl_dsp_wake(struct sst_dsp *ctx); +int skl_dsp_sleep(struct sst_dsp *ctx); +void skl_dsp_free(struct sst_dsp *dsp); + +int skl_dsp_boot(struct sst_dsp *ctx); +int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp); +void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); + +#endif /*__SKL_SST_DSP_H__*/ diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c new file mode 100644 index 0000000..937a0a3 --- /dev/null +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -0,0 +1,771 @@ +/* + * skl-sst-ipc.c - Intel skl IPC Support + * + * Copyright (C) 2014-15, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ +#include <linux/device.h> + +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-sst-dsp.h" +#include "skl-sst-ipc.h" + + +#define IPC_IXC_STATUS_BITS 24 + +/* Global Message - Generic */ +#define IPC_GLB_TYPE_SHIFT 24 +#define IPC_GLB_TYPE_MASK (0xf << IPC_GLB_TYPE_SHIFT) +#define IPC_GLB_TYPE(x) ((x) << IPC_GLB_TYPE_SHIFT) + +/* Global Message - Reply */ +#define IPC_GLB_REPLY_STATUS_SHIFT 24 +#define IPC_GLB_REPLY_STATUS_MASK ((0x1 << IPC_GLB_REPLY_STATUS_SHIFT) - 1) +#define IPC_GLB_REPLY_STATUS(x) ((x) << IPC_GLB_REPLY_STATUS_SHIFT) + +#define IPC_TIMEOUT_MSECS 3000 + +#define IPC_EMPTY_LIST_SIZE 8 + +#define IPC_MSG_TARGET_SHIFT 30 +#define IPC_MSG_TARGET_MASK 0x1 +#define IPC_MSG_TARGET(x) (((x) & IPC_MSG_TARGET_MASK) \ + << IPC_MSG_TARGET_SHIFT) + +#define IPC_MSG_DIR_SHIFT 29 +#define IPC_MSG_DIR_MASK 0x1 +#define IPC_MSG_DIR(x) (((x) & IPC_MSG_DIR_MASK) \ + << IPC_MSG_DIR_SHIFT) +/* Global Notification Message */ +#define IPC_GLB_NOTIFY_TYPE_SHIFT 16 +#define IPC_GLB_NOTIFY_TYPE_MASK 0xFF +#define IPC_GLB_NOTIFY_TYPE(x) (((x) >> IPC_GLB_NOTIFY_TYPE_SHIFT) \ + & IPC_GLB_NOTIFY_TYPE_MASK) + +#define IPC_GLB_NOTIFY_MSG_TYPE_SHIFT 24 +#define IPC_GLB_NOTIFY_MSG_TYPE_MASK 0x1F +#define IPC_GLB_NOTIFY_MSG_TYPE(x) (((x) >> IPC_GLB_NOTIFY_MSG_TYPE_SHIFT) \ + & IPC_GLB_NOTIFY_MSG_TYPE_MASK) + +#define IPC_GLB_NOTIFY_RSP_SHIFT 29 +#define IPC_GLB_NOTIFY_RSP_MASK 0x1 +#define IPC_GLB_NOTIFY_RSP_TYPE(x) (((x) >> IPC_GLB_NOTIFY_RSP_SHIFT) \ + & IPC_GLB_NOTIFY_RSP_MASK) + +/* Pipeline operations */ + +/* Create pipeline message */ +#define IPC_PPL_MEM_SIZE_SHIFT 0 +#define IPC_PPL_MEM_SIZE_MASK 0x7FF +#define IPC_PPL_MEM_SIZE(x) (((x) & IPC_PPL_MEM_SIZE_MASK) \ + << IPC_PPL_MEM_SIZE_SHIFT) + +#define IPC_PPL_TYPE_SHIFT 11 +#define IPC_PPL_TYPE_MASK 0x1F +#define IPC_PPL_TYPE(x) (((x) & IPC_PPL_TYPE_MASK) \ + << IPC_PPL_TYPE_SHIFT) + +#define IPC_INSTANCE_ID_SHIFT 16 +#define IPC_INSTANCE_ID_MASK 0xFF +#define IPC_INSTANCE_ID(x) (((x) & IPC_INSTANCE_ID_MASK) \ + << IPC_INSTANCE_ID_SHIFT) + +/* Set pipeline state message */ +#define IPC_PPL_STATE_SHIFT 0 +#define IPC_PPL_STATE_MASK 0x1F +#define IPC_PPL_STATE(x) (((x) & IPC_PPL_STATE_MASK) \ + << IPC_PPL_STATE_SHIFT) + +/* Module operations primary register */ +#define IPC_MOD_ID_SHIFT 0 +#define IPC_MOD_ID_MASK 0xFFFF +#define IPC_MOD_ID(x) (((x) & IPC_MOD_ID_MASK) \ + << IPC_MOD_ID_SHIFT) + +#define IPC_MOD_INSTANCE_ID_SHIFT 16 +#define IPC_MOD_INSTANCE_ID_MASK 0xFF +#define IPC_MOD_INSTANCE_ID(x) (((x) & IPC_MOD_INSTANCE_ID_MASK) \ + << IPC_MOD_INSTANCE_ID_SHIFT) + +/* Init instance message extension register */ +#define IPC_PARAM_BLOCK_SIZE_SHIFT 0 +#define IPC_PARAM_BLOCK_SIZE_MASK 0xFFFF +#define IPC_PARAM_BLOCK_SIZE(x) (((x) & IPC_PARAM_BLOCK_SIZE_MASK) \ + << IPC_PARAM_BLOCK_SIZE_SHIFT) + +#define IPC_PPL_INSTANCE_ID_SHIFT 16 +#define IPC_PPL_INSTANCE_ID_MASK 0xFF +#define IPC_PPL_INSTANCE_ID(x) (((x) & IPC_PPL_INSTANCE_ID_MASK) \ + << IPC_PPL_INSTANCE_ID_SHIFT) + +#define IPC_CORE_ID_SHIFT 24 +#define IPC_CORE_ID_MASK 0x1F +#define IPC_CORE_ID(x) (((x) & IPC_CORE_ID_MASK) \ + << IPC_CORE_ID_SHIFT) + +/* Bind/Unbind message extension register */ +#define IPC_DST_MOD_ID_SHIFT 0 +#define IPC_DST_MOD_ID(x) (((x) & IPC_MOD_ID_MASK) \ + << IPC_DST_MOD_ID_SHIFT) + +#define IPC_DST_MOD_INSTANCE_ID_SHIFT 16 +#define IPC_DST_MOD_INSTANCE_ID(x) (((x) & IPC_MOD_INSTANCE_ID_MASK) \ + << IPC_DST_MOD_INSTANCE_ID_SHIFT) + +#define IPC_DST_QUEUE_SHIFT 24 +#define IPC_DST_QUEUE_MASK 0x7 +#define IPC_DST_QUEUE(x) (((x) & IPC_DST_QUEUE_MASK) \ + << IPC_DST_QUEUE_SHIFT) + +#define IPC_SRC_QUEUE_SHIFT 27 +#define IPC_SRC_QUEUE_MASK 0x7 +#define IPC_SRC_QUEUE(x) (((x) & IPC_SRC_QUEUE_MASK) \ + << IPC_SRC_QUEUE_SHIFT) + +/* Save pipeline messgae extension register */ +#define IPC_DMA_ID_SHIFT 0 +#define IPC_DMA_ID_MASK 0x1F +#define IPC_DMA_ID(x) (((x) & IPC_DMA_ID_MASK) \ + << IPC_DMA_ID_SHIFT) +/* Large Config message extension register */ +#define IPC_DATA_OFFSET_SZ_SHIFT 0 +#define IPC_DATA_OFFSET_SZ_MASK 0xFFFFF +#define IPC_DATA_OFFSET_SZ(x) (((x) & IPC_DATA_OFFSET_SZ_MASK) \ + << IPC_DATA_OFFSET_SZ_SHIFT) +#define IPC_DATA_OFFSET_SZ_CLEAR ~(IPC_DATA_OFFSET_SZ_MASK \ + << IPC_DATA_OFFSET_SZ_SHIFT) + +#define IPC_LARGE_PARAM_ID_SHIFT 20 +#define IPC_LARGE_PARAM_ID_MASK 0xFF +#define IPC_LARGE_PARAM_ID(x) (((x) & IPC_LARGE_PARAM_ID_MASK) \ + << IPC_LARGE_PARAM_ID_SHIFT) + +#define IPC_FINAL_BLOCK_SHIFT 28 +#define IPC_FINAL_BLOCK_MASK 0x1 +#define IPC_FINAL_BLOCK(x) (((x) & IPC_FINAL_BLOCK_MASK) \ + << IPC_FINAL_BLOCK_SHIFT) + +#define IPC_INITIAL_BLOCK_SHIFT 29 +#define IPC_INITIAL_BLOCK_MASK 0x1 +#define IPC_INITIAL_BLOCK(x) (((x) & IPC_INITIAL_BLOCK_MASK) \ + << IPC_INITIAL_BLOCK_SHIFT) +#define IPC_INITIAL_BLOCK_CLEAR ~(IPC_INITIAL_BLOCK_MASK \ + << IPC_INITIAL_BLOCK_SHIFT) + +enum skl_ipc_msg_target { + IPC_FW_GEN_MSG = 0, + IPC_MOD_MSG = 1 +}; + +enum skl_ipc_msg_direction { + IPC_MSG_REQUEST = 0, + IPC_MSG_REPLY = 1 +}; + +/* Global Message Types */ +enum skl_ipc_glb_type { + IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ + IPC_GLB_LOAD_MULTIPLE_MODS = 15, + IPC_GLB_UNLOAD_MULTIPLE_MODS = 16, + IPC_GLB_CREATE_PPL = 17, + IPC_GLB_DELETE_PPL = 18, + IPC_GLB_SET_PPL_STATE = 19, + IPC_GLB_GET_PPL_STATE = 20, + IPC_GLB_GET_PPL_CONTEXT_SIZE = 21, + IPC_GLB_SAVE_PPL = 22, + IPC_GLB_RESTORE_PPL = 23, + IPC_GLB_NOTIFY = 26, + IPC_GLB_MAX_IPC_MSG_NUMBER = 31 /* Maximum message number */ +}; + +enum skl_ipc_glb_reply { + IPC_GLB_REPLY_SUCCESS = 0, + + IPC_GLB_REPLY_UNKNOWN_MSG_TYPE = 1, + IPC_GLB_REPLY_ERROR_INVALID_PARAM = 2, + + IPC_GLB_REPLY_BUSY = 3, + IPC_GLB_REPLY_PENDING = 4, + IPC_GLB_REPLY_FAILURE = 5, + IPC_GLB_REPLY_INVALID_REQUEST = 6, + + IPC_GLB_REPLY_OUT_OF_MEMORY = 7, + IPC_GLB_REPLY_OUT_OF_MIPS = 8, + + IPC_GLB_REPLY_INVALID_RESOURCE_ID = 9, + IPC_GLB_REPLY_INVALID_RESOURCE_STATE = 10, + + IPC_GLB_REPLY_MOD_MGMT_ERROR = 100, + IPC_GLB_REPLY_MOD_LOAD_CL_FAILED = 101, + IPC_GLB_REPLY_MOD_LOAD_INVALID_HASH = 102, + + IPC_GLB_REPLY_MOD_UNLOAD_INST_EXIST = 103, + IPC_GLB_REPLY_MOD_NOT_INITIALIZED = 104, + + IPC_GLB_REPLY_INVALID_CONFIG_PARAM_ID = 120, + IPC_GLB_REPLY_INVALID_CONFIG_DATA_LEN = 121, + IPC_GLB_REPLY_GATEWAY_NOT_INITIALIZED = 140, + IPC_GLB_REPLY_GATEWAY_NOT_EXIST = 141, + + IPC_GLB_REPLY_PPL_NOT_INITIALIZED = 160, + IPC_GLB_REPLY_PPL_NOT_EXIST = 161, + IPC_GLB_REPLY_PPL_SAVE_FAILED = 162, + IPC_GLB_REPLY_PPL_RESTORE_FAILED = 163, + + IPC_MAX_STATUS = ((1<<IPC_IXC_STATUS_BITS)-1) +}; + +enum skl_ipc_notification_type { + IPC_GLB_NOTIFY_GLITCH = 0, + IPC_GLB_NOTIFY_OVERRUN = 1, + IPC_GLB_NOTIFY_UNDERRUN = 2, + IPC_GLB_NOTIFY_END_STREAM = 3, + IPC_GLB_NOTIFY_PHRASE_DETECTED = 4, + IPC_GLB_NOTIFY_RESOURCE_EVENT = 5, + IPC_GLB_NOTIFY_LOG_BUFFER_STATUS = 6, + IPC_GLB_NOTIFY_TIMESTAMP_CAPTURED = 7, + IPC_GLB_NOTIFY_FW_READY = 8 +}; + +/* Module Message Types */ +enum skl_ipc_module_msg { + IPC_MOD_INIT_INSTANCE = 0, + IPC_MOD_CONFIG_GET = 1, + IPC_MOD_CONFIG_SET = 2, + IPC_MOD_LARGE_CONFIG_GET = 3, + IPC_MOD_LARGE_CONFIG_SET = 4, + IPC_MOD_BIND = 5, + IPC_MOD_UNBIND = 6, + IPC_MOD_SET_DX = 7 +}; + +static void skl_ipc_tx_data_copy(struct ipc_message *msg, char *tx_data, + size_t tx_size) +{ + if (tx_size) + memcpy(msg->tx_data, tx_data, tx_size); +} + +static bool skl_ipc_is_dsp_busy(struct sst_dsp *dsp) +{ + u32 hipci; + + hipci = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCI); + return (hipci & SKL_ADSP_REG_HIPCI_BUSY); +} + +/* Lock to be held by caller */ +static void skl_ipc_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg) +{ + struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->header); + + if (msg->tx_size) + sst_dsp_outbox_write(ipc->dsp, msg->tx_data, msg->tx_size); + sst_dsp_shim_write_unlocked(ipc->dsp, SKL_ADSP_REG_HIPCIE, + header->extension); + sst_dsp_shim_write_unlocked(ipc->dsp, SKL_ADSP_REG_HIPCI, + header->primary | SKL_ADSP_REG_HIPCI_BUSY); +} + +static struct ipc_message *skl_ipc_reply_get_msg(struct sst_generic_ipc *ipc, + u64 ipc_header) +{ + struct ipc_message *msg = NULL; + struct skl_ipc_header *header = (struct skl_ipc_header *)(&ipc_header); + + if (list_empty(&ipc->rx_list)) { + dev_err(ipc->dev, "ipc: rx list is empty but received 0x%x\n", + header->primary); + goto out; + } + + msg = list_first_entry(&ipc->rx_list, struct ipc_message, list); + +out: + return msg; + +} + +static int skl_ipc_process_notification(struct sst_generic_ipc *ipc, + struct skl_ipc_header header) +{ + struct skl_sst *skl = container_of(ipc, struct skl_sst, ipc); + + if (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { + switch (IPC_GLB_NOTIFY_TYPE(header.primary)) { + + case IPC_GLB_NOTIFY_UNDERRUN: + dev_err(ipc->dev, "FW Underrun %x\n", header.primary); + break; + + case IPC_GLB_NOTIFY_RESOURCE_EVENT: + dev_err(ipc->dev, "MCPS Budget Violation: %x\n", + header.primary); + break; + + case IPC_GLB_NOTIFY_FW_READY: + skl->boot_complete = true; + wake_up(&skl->boot_wait); + break; + + default: + dev_err(ipc->dev, "ipc: Unhandled error msg=%x", + header.primary); + break; + } + } + + return 0; +} + +static void skl_ipc_process_reply(struct sst_generic_ipc *ipc, + struct skl_ipc_header header) +{ + struct ipc_message *msg; + u32 reply = header.primary & IPC_GLB_REPLY_STATUS_MASK; + u64 *ipc_header = (u64 *)(&header); + + msg = skl_ipc_reply_get_msg(ipc, *ipc_header); + if (msg == NULL) { + dev_dbg(ipc->dev, "ipc: rx list is empty\n"); + return; + } + + /* first process the header */ + switch (reply) { + case IPC_GLB_REPLY_SUCCESS: + dev_info(ipc->dev, "ipc FW reply %x: success\n", header.primary); + break; + + case IPC_GLB_REPLY_OUT_OF_MEMORY: + dev_err(ipc->dev, "ipc fw reply: %x: no memory\n", header.primary); + msg->errno = -ENOMEM; + break; + + case IPC_GLB_REPLY_BUSY: + dev_err(ipc->dev, "ipc fw reply: %x: Busy\n", header.primary); + msg->errno = -EBUSY; + break; + + default: + dev_err(ipc->dev, "Unknown ipc reply: 0x%x", reply); + msg->errno = -EINVAL; + break; + } + + if (reply != IPC_GLB_REPLY_SUCCESS) { + dev_err(ipc->dev, "ipc FW reply: reply=%d", reply); + dev_err(ipc->dev, "FW Error Code: %u\n", + ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); + } + + list_del(&msg->list); + sst_ipc_tx_msg_reply_complete(ipc, msg); +} + +irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context) +{ + struct sst_dsp *dsp = context; + struct skl_sst *skl = sst_dsp_get_thread_context(dsp); + struct sst_generic_ipc *ipc = &skl->ipc; + struct skl_ipc_header header = {0}; + u32 hipcie, hipct, hipcte; + int ipc_irq = 0; + + if (dsp->intr_status & SKL_ADSPIS_CL_DMA) + skl_cldma_process_intr(dsp); + + /* Here we handle IPC interrupts only */ + if (!(dsp->intr_status & SKL_ADSPIS_IPC)) + return IRQ_NONE; + + hipcie = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCIE); + hipct = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCT); + + /* reply message from DSP */ + if (hipcie & SKL_ADSP_REG_HIPCIE_DONE) { + sst_dsp_shim_update_bits(dsp, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_DONE, 0); + + /* clear DONE bit - tell DSP we have completed the operation */ + sst_dsp_shim_update_bits_forced(dsp, SKL_ADSP_REG_HIPCIE, + SKL_ADSP_REG_HIPCIE_DONE, SKL_ADSP_REG_HIPCIE_DONE); + + ipc_irq = 1; + + /* unmask Done interrupt */ + sst_dsp_shim_update_bits(dsp, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_DONE, SKL_ADSP_REG_HIPCCTL_DONE); + } + + /* New message from DSP */ + if (hipct & SKL_ADSP_REG_HIPCT_BUSY) { + hipcte = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCTE); + header.primary = hipct; + header.extension = hipcte; + dev_dbg(dsp->dev, "IPC irq: Firmware respond primary:%x", + header.primary); + dev_dbg(dsp->dev, "IPC irq: Firmware respond extension:%x", + header.extension); + + if (IPC_GLB_NOTIFY_RSP_TYPE(header.primary)) { + /* Handle Immediate reply from DSP Core */ + skl_ipc_process_reply(ipc, header); + } else { + dev_dbg(dsp->dev, "IPC irq: Notification from firmware\n"); + skl_ipc_process_notification(ipc, header); + } + /* clear busy interrupt */ + sst_dsp_shim_update_bits_forced(dsp, SKL_ADSP_REG_HIPCT, + SKL_ADSP_REG_HIPCT_BUSY, SKL_ADSP_REG_HIPCT_BUSY); + ipc_irq = 1; + } + + if (ipc_irq == 0) + return IRQ_NONE; + + skl_ipc_int_enable(dsp); + + /* continue to send any remaining messages... */ + queue_kthread_work(&ipc->kworker, &ipc->kwork); + + return IRQ_HANDLED; +} + +void skl_ipc_int_enable(struct sst_dsp *ctx) +{ + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_ADSPIC, + SKL_ADSPIC_IPC, SKL_ADSPIC_IPC); +} + +void skl_ipc_int_disable(struct sst_dsp *ctx) +{ + sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPIC, + SKL_ADSPIC_IPC, 0); +} + +void skl_ipc_op_int_enable(struct sst_dsp *ctx) +{ + /* enable IPC DONE interrupt */ + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_DONE, SKL_ADSP_REG_HIPCCTL_DONE); + + /* Enable IPC BUSY interrupt */ + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_BUSY, SKL_ADSP_REG_HIPCCTL_BUSY); +} + +bool skl_ipc_int_status(struct sst_dsp *ctx) +{ + return sst_dsp_shim_read_unlocked(ctx, + SKL_ADSP_REG_ADSPIS) & SKL_ADSPIS_IPC; +} + +int skl_ipc_init(struct device *dev, struct skl_sst *skl) +{ + struct sst_generic_ipc *ipc; + int err; + + ipc = &skl->ipc; + ipc->dsp = skl->dsp; + ipc->dev = dev; + + ipc->tx_data_max_size = SKL_ADSP_W1_SZ; + ipc->rx_data_max_size = SKL_ADSP_W0_UP_SZ; + + err = sst_ipc_init(ipc); + if (err) + return err; + + ipc->ops.tx_msg = skl_ipc_tx_msg; + ipc->ops.tx_data_copy = skl_ipc_tx_data_copy; + ipc->ops.is_dsp_busy = skl_ipc_is_dsp_busy; + + return 0; +} + +void skl_ipc_free(struct sst_generic_ipc *ipc) +{ + /* Disable IPC DONE interrupt */ + sst_dsp_shim_update_bits(ipc->dsp, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_DONE, 0); + + /* Disable IPC BUSY interrupt */ + sst_dsp_shim_update_bits(ipc->dsp, SKL_ADSP_REG_HIPCCTL, + SKL_ADSP_REG_HIPCCTL_BUSY, 0); + + sst_ipc_fini(ipc); +} + +int skl_ipc_create_pipeline(struct sst_generic_ipc *ipc, + u16 ppl_mem_size, u8 ppl_type, u8 instance_id) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret; + + header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_GLB_CREATE_PPL); + header.primary |= IPC_INSTANCE_ID(instance_id); + header.primary |= IPC_PPL_TYPE(ppl_type); + header.primary |= IPC_PPL_MEM_SIZE(ppl_mem_size); + + dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, "ipc: create pipeline fail, err: %d\n", ret); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_create_pipeline); + +int skl_ipc_delete_pipeline(struct sst_generic_ipc *ipc, u8 instance_id) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret; + + header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_GLB_DELETE_PPL); + header.primary |= IPC_INSTANCE_ID(instance_id); + + dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, "ipc: delete pipeline failed, err %d\n", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(skl_ipc_delete_pipeline); + +int skl_ipc_set_pipeline_state(struct sst_generic_ipc *ipc, + u8 instance_id, enum skl_ipc_pipeline_state state) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret; + + header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_GLB_SET_PPL_STATE); + header.primary |= IPC_INSTANCE_ID(instance_id); + header.primary |= IPC_PPL_STATE(state); + + dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, "ipc: set pipeline state failed, err: %d\n", ret); + return ret; + } + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_set_pipeline_state); + +int +skl_ipc_save_pipeline(struct sst_generic_ipc *ipc, u8 instance_id, int dma_id) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret; + + header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_GLB_SAVE_PPL); + header.primary |= IPC_INSTANCE_ID(instance_id); + + header.extension = IPC_DMA_ID(dma_id); + dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, "ipc: save pipeline failed, err: %d\n", ret); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_save_pipeline); + +int skl_ipc_restore_pipeline(struct sst_generic_ipc *ipc, u8 instance_id) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret; + + header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_GLB_RESTORE_PPL); + header.primary |= IPC_INSTANCE_ID(instance_id); + + dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, "ipc: restore pipeline failed, err: %d\n", ret); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_restore_pipeline); + +int skl_ipc_set_dx(struct sst_generic_ipc *ipc, u8 instance_id, + u16 module_id, struct skl_ipc_dxstate_info *dx) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret; + + header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_MOD_SET_DX); + header.primary |= IPC_MOD_INSTANCE_ID(instance_id); + header.primary |= IPC_MOD_ID(module_id); + + dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__, + header.primary, header.extension); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, + dx, sizeof(dx), NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, "ipc: set dx failed, err %d\n", ret); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_set_dx); + +int skl_ipc_init_instance(struct sst_generic_ipc *ipc, + struct skl_ipc_init_instance_msg *msg, void *param_data) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret; + u32 *buffer = (u32 *)param_data; + /* param_block_size must be in dwords */ + u16 param_block_size = msg->param_data_size / sizeof(u32); + + print_hex_dump(KERN_DEBUG, NULL, DUMP_PREFIX_NONE, + 16, 4, buffer, param_block_size, false); + + header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_MOD_INIT_INSTANCE); + header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); + header.primary |= IPC_MOD_ID(msg->module_id); + + header.extension = IPC_CORE_ID(msg->core_id); + header.extension |= IPC_PPL_INSTANCE_ID(msg->ppl_instance_id); + header.extension |= IPC_PARAM_BLOCK_SIZE(param_block_size); + + dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__, + header.primary, header.extension); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, param_data, + msg->param_data_size, NULL, 0); + + if (ret < 0) { + dev_err(ipc->dev, "ipc: init instance failed\n"); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_init_instance); + +int skl_ipc_bind_unbind(struct sst_generic_ipc *ipc, + struct skl_ipc_bind_unbind_msg *msg) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + u8 bind_unbind = msg->bind ? IPC_MOD_BIND : IPC_MOD_UNBIND; + int ret; + + header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(bind_unbind); + header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); + header.primary |= IPC_MOD_ID(msg->module_id); + + header.extension = IPC_DST_MOD_ID(msg->dst_module_id); + header.extension |= IPC_DST_MOD_INSTANCE_ID(msg->dst_instance_id); + header.extension |= IPC_DST_QUEUE(msg->dst_queue); + header.extension |= IPC_SRC_QUEUE(msg->src_queue); + + dev_dbg(ipc->dev, "In %s hdr=%x ext=%x\n", __func__, header.primary, + header.extension); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, "ipc: bind/unbind faileden"); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_bind_unbind); + +int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, + struct skl_ipc_large_config_msg *msg, u32 *param) +{ + struct skl_ipc_header header = {0}; + u64 *ipc_header = (u64 *)(&header); + int ret = 0; + size_t sz_remaining, tx_size, data_offset; + + header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); + header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); + header.primary |= IPC_GLB_TYPE(IPC_MOD_LARGE_CONFIG_SET); + header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); + header.primary |= IPC_MOD_ID(msg->module_id); + + header.extension = IPC_DATA_OFFSET_SZ(msg->param_data_size); + header.extension |= IPC_LARGE_PARAM_ID(msg->large_param_id); + header.extension |= IPC_FINAL_BLOCK(0); + header.extension |= IPC_INITIAL_BLOCK(1); + + sz_remaining = msg->param_data_size; + data_offset = 0; + while (sz_remaining != 0) { + tx_size = sz_remaining > SKL_ADSP_W1_SZ + ? SKL_ADSP_W1_SZ : sz_remaining; + if (tx_size == sz_remaining) + header.extension |= IPC_FINAL_BLOCK(1); + + dev_dbg(ipc->dev, "In %s primary=%#x ext=%#x\n", __func__, + header.primary, header.extension); + dev_dbg(ipc->dev, "transmitting offset: %#x, size: %#x\n", + (unsigned)data_offset, (unsigned)tx_size); + ret = sst_ipc_tx_message_wait(ipc, *ipc_header, + ((char *)param) + data_offset, + tx_size, NULL, 0); + if (ret < 0) { + dev_err(ipc->dev, + "ipc: set large config fail, err: %d\n", ret); + return ret; + } + sz_remaining -= tx_size; + data_offset = msg->param_data_size - sz_remaining; + + /* clear the fields */ + header.extension &= IPC_INITIAL_BLOCK_CLEAR; + header.extension &= IPC_DATA_OFFSET_SZ_CLEAR; + /* fill the fields */ + header.extension |= IPC_INITIAL_BLOCK(0); + header.extension |= IPC_DATA_OFFSET_SZ(data_offset); + } + + return ret; +} +EXPORT_SYMBOL_GPL(skl_ipc_set_large_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h new file mode 100644 index 0000000..9f5f672 --- /dev/null +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -0,0 +1,125 @@ +/* + * Intel SKL IPC Support + * + * Copyright (C) 2014-15, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ + +#ifndef __SKL_IPC_H +#define __SKL_IPC_H + +#include <linux/kthread.h> +#include <linux/irqreturn.h> +#include "../common/sst-ipc.h" + +struct sst_dsp; +struct skl_sst; +struct sst_generic_ipc; + +enum skl_ipc_pipeline_state { + PPL_INVALID_STATE = 0, + PPL_UNINITIALIZED = 1, + PPL_RESET = 2, + PPL_PAUSED = 3, + PPL_RUNNING = 4, + PPL_ERROR_STOP = 5, + PPL_SAVED = 6, + PPL_RESTORED = 7 +}; + +struct skl_ipc_dxstate_info { + u32 core_mask; + u32 dx_mask; +}; + +struct skl_ipc_header { + u32 primary; + u32 extension; +}; + +struct skl_sst { + struct device *dev; + struct sst_dsp *dsp; + + /* boot */ + wait_queue_head_t boot_wait; + bool boot_complete; + + /* IPC messaging */ + struct sst_generic_ipc ipc; +}; + +struct skl_ipc_init_instance_msg { + u32 module_id; + u32 instance_id; + u16 param_data_size; + u8 ppl_instance_id; + u8 core_id; +}; + +struct skl_ipc_bind_unbind_msg { + u32 module_id; + u32 instance_id; + u32 dst_module_id; + u32 dst_instance_id; + u8 src_queue; + u8 dst_queue; + bool bind; +}; + +struct skl_ipc_large_config_msg { + u32 module_id; + u32 instance_id; + u32 large_param_id; + u32 param_data_size; +}; + +#define SKL_IPC_BOOT_MSECS 3000 + +#define SKL_IPC_D3_MASK 0 +#define SKL_IPC_D0_MASK 3 + +irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context); + +int skl_ipc_create_pipeline(struct sst_generic_ipc *sst_ipc, + u16 ppl_mem_size, u8 ppl_type, u8 instance_id); + +int skl_ipc_delete_pipeline(struct sst_generic_ipc *sst_ipc, u8 instance_id); + +int skl_ipc_set_pipeline_state(struct sst_generic_ipc *sst_ipc, + u8 instance_id, enum skl_ipc_pipeline_state state); + +int skl_ipc_save_pipeline(struct sst_generic_ipc *ipc, + u8 instance_id, int dma_id); + +int skl_ipc_restore_pipeline(struct sst_generic_ipc *ipc, u8 instance_id); + +int skl_ipc_init_instance(struct sst_generic_ipc *sst_ipc, + struct skl_ipc_init_instance_msg *msg, void *param_data); + +int skl_ipc_bind_unbind(struct sst_generic_ipc *sst_ipc, + struct skl_ipc_bind_unbind_msg *msg); + +int skl_ipc_set_dx(struct sst_generic_ipc *ipc, + u8 instance_id, u16 module_id, struct skl_ipc_dxstate_info *dx); + +int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, + struct skl_ipc_large_config_msg *msg, u32 *param); + +void skl_ipc_int_enable(struct sst_dsp *dsp); +void skl_ipc_op_int_enable(struct sst_dsp *ctx); +void skl_ipc_int_disable(struct sst_dsp *dsp); + +bool skl_ipc_int_status(struct sst_dsp *dsp); +void skl_ipc_free(struct sst_generic_ipc *ipc); +int skl_ipc_init(struct device *dev, struct skl_sst *skl); + +#endif /* __SKL_IPC_H */ diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c new file mode 100644 index 0000000..c18ea51 --- /dev/null +++ b/sound/soc/intel/skylake/skl-sst.c @@ -0,0 +1,280 @@ +/* + * skl-sst.c - HDA DSP library functions for SKL platform + * + * Copyright (C) 2014-15, Intel Corporation. + * Author:Rafal Redzimski <rafal.f.redzimski@intel.com> + * Jeeja KP <jeeja.kp@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/device.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "../common/sst-ipc.h" +#include "skl-sst-ipc.h" + +#define SKL_BASEFW_TIMEOUT 300 +#define SKL_INIT_TIMEOUT 1000 + +/* Intel HD Audio SRAM Window 0*/ +#define SKL_ADSP_SRAM0_BASE 0x8000 + +/* Firmware status window */ +#define SKL_ADSP_FW_STATUS SKL_ADSP_SRAM0_BASE +#define SKL_ADSP_ERROR_CODE (SKL_ADSP_FW_STATUS + 0x4) + +#define SKL_INSTANCE_ID 0 +#define SKL_BASE_FW_MODULE_ID 0 + +static bool skl_check_fw_status(struct sst_dsp *ctx, u32 status) +{ + u32 cur_sts; + + cur_sts = sst_dsp_shim_read(ctx, SKL_ADSP_FW_STATUS) & SKL_FW_STS_MASK; + + return (cur_sts == status); +} + +static int skl_transfer_firmware(struct sst_dsp *ctx, + const void *basefw, u32 base_fw_size) +{ + int ret = 0; + + ret = ctx->cl_dev.ops.cl_copy_to_dmabuf(ctx, basefw, base_fw_size); + if (ret < 0) + return ret; + + ret = sst_dsp_register_poll(ctx, + SKL_ADSP_FW_STATUS, + SKL_FW_STS_MASK, + SKL_FW_RFW_START, + SKL_BASEFW_TIMEOUT, + "Firmware boot"); + + ctx->cl_dev.ops.cl_stop_dma(ctx); + + return ret; +} + +static int skl_load_base_firmware(struct sst_dsp *ctx) +{ + int ret = 0, i; + const struct firmware *fw = NULL; + struct skl_sst *skl = ctx->thread_context; + u32 reg; + + ret = request_firmware(&fw, "dsp_fw_release.bin", ctx->dev); + if (ret < 0) { + dev_err(ctx->dev, "Request firmware failed %d\n", ret); + skl_dsp_disable_core(ctx); + return -EIO; + } + + /* enable Interrupt */ + skl_ipc_int_enable(ctx); + skl_ipc_op_int_enable(ctx); + + /* check ROM Status */ + for (i = SKL_INIT_TIMEOUT; i > 0; --i) { + if (skl_check_fw_status(ctx, SKL_FW_INIT)) { + dev_dbg(ctx->dev, + "ROM loaded, we can continue with FW loading\n"); + break; + } + mdelay(1); + } + if (!i) { + reg = sst_dsp_shim_read(ctx, SKL_ADSP_FW_STATUS); + dev_err(ctx->dev, + "Timeout waiting for ROM init done, reg:0x%x\n", reg); + ret = -EIO; + goto skl_load_base_firmware_failed; + } + + ret = skl_transfer_firmware(ctx, fw->data, fw->size); + if (ret < 0) { + dev_err(ctx->dev, "Transfer firmware failed%d\n", ret); + goto skl_load_base_firmware_failed; + } else { + ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, + msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(ctx->dev, "DSP boot failed, FW Ready timed-out\n"); + ret = -EIO; + goto skl_load_base_firmware_failed; + } + + dev_dbg(ctx->dev, "Download firmware successful%d\n", ret); + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + } + release_firmware(fw); + + return 0; + +skl_load_base_firmware_failed: + skl_dsp_disable_core(ctx); + release_firmware(fw); + return ret; +} + +static int skl_set_dsp_D0(struct sst_dsp *ctx) +{ + int ret; + + ret = skl_load_base_firmware(ctx); + if (ret < 0) { + dev_err(ctx->dev, "unable to load firmware\n"); + return ret; + } + + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + + return ret; +} + +static int skl_set_dsp_D3(struct sst_dsp *ctx) +{ + int ret; + struct skl_ipc_dxstate_info dx; + struct skl_sst *skl = ctx->thread_context; + + dev_dbg(ctx->dev, "In %s:\n", __func__); + mutex_lock(&ctx->mutex); + if (!is_skl_dsp_running(ctx)) { + mutex_unlock(&ctx->mutex); + return 0; + } + mutex_unlock(&ctx->mutex); + + dx.core_mask = SKL_DSP_CORE0_MASK; + dx.dx_mask = SKL_IPC_D3_MASK; + ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID, SKL_BASE_FW_MODULE_ID, &dx); + if (ret < 0) { + dev_err(ctx->dev, "Failed to set DSP to D3 state\n"); + return ret; + } + + ret = skl_dsp_disable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "disable dsp core failed ret: %d\n", ret); + ret = -EIO; + } + skl_dsp_set_state_locked(ctx, SKL_DSP_RESET); + + return ret; +} + +static unsigned int skl_get_errorcode(struct sst_dsp *ctx) +{ + return sst_dsp_shim_read(ctx, SKL_ADSP_ERROR_CODE); +} + +static struct skl_dsp_fw_ops skl_fw_ops = { + .set_state_D0 = skl_set_dsp_D0, + .set_state_D3 = skl_set_dsp_D3, + .load_fw = skl_load_base_firmware, + .get_fw_errcode = skl_get_errorcode, +}; + +static struct sst_ops skl_ops = { + .irq_handler = skl_dsp_sst_interrupt, + .write = sst_shim32_write, + .read = sst_shim32_read, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .free = skl_dsp_free, +}; + +static struct sst_dsp_device skl_dev = { + .thread = skl_dsp_irq_thread_handler, + .ops = &skl_ops, +}; + +int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp) +{ + struct skl_sst *skl; + struct sst_dsp *sst; + int ret; + + skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL); + if (skl == NULL) + return -ENOMEM; + + skl->dev = dev; + skl_dev.thread_context = skl; + + skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq); + if (!skl->dsp) { + dev_err(skl->dev, "%s: no device\n", __func__); + return -ENODEV; + } + + sst = skl->dsp; + + sst->addr.lpe = mmio_base; + sst->addr.shim = mmio_base; + sst_dsp_mailbox_init(sst, (SKL_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), + SKL_ADSP_W0_UP_SZ, SKL_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); + + sst->dsp_ops = dsp_ops; + sst->fw_ops = skl_fw_ops; + + ret = skl_ipc_init(dev, skl); + if (ret) + return ret; + + skl->boot_complete = false; + init_waitqueue_head(&skl->boot_wait); + + ret = skl_dsp_boot(sst); + if (ret < 0) { + dev_err(skl->dev, "Boot dsp core failed ret: %d", ret); + goto free_ipc; + } + + ret = skl_cldma_prepare(sst); + if (ret < 0) { + dev_err(dev, "CL dma prepare failed : %d", ret); + goto free_ipc; + } + + + ret = sst->fw_ops.load_fw(sst); + if (ret < 0) { + dev_err(dev, "Load base fw failed : %d", ret); + return ret; + } + + if (dsp) + *dsp = skl; + + return 0; + +free_ipc: + skl_ipc_free(&skl->ipc); + return ret; +} +EXPORT_SYMBOL_GPL(skl_sst_dsp_init); + +void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) +{ + skl_ipc_free(&ctx->ipc); + ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp); + ctx->dsp->ops->free(ctx->dsp); +} +EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Skylake IPC driver"); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h new file mode 100644 index 0000000..8c7767b --- /dev/null +++ b/sound/soc/intel/skylake/skl-topology.h @@ -0,0 +1,286 @@ +/* + * skl_topology.h - Intel HDA Platform topology header file + * + * Copyright (C) 2014-15 Intel Corp + * Author: Jeeja KP <jeeja.kp@intel.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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SKL_TOPOLOGY_H__ +#define __SKL_TOPOLOGY_H__ + +#include <linux/types.h> + +#include <sound/hdaudio_ext.h> +#include <sound/soc.h> +#include "skl.h" +#include "skl-tplg-interface.h" + +#define BITS_PER_BYTE 8 +#define MAX_TS_GROUPS 8 +#define MAX_DMIC_TS_GROUPS 4 +#define MAX_FIXED_DMIC_PARAMS_SIZE 727 + +/* Maximum number of coefficients up down mixer module */ +#define UP_DOWN_MIXER_MAX_COEFF 6 + +enum skl_channel_index { + SKL_CHANNEL_LEFT = 0, + SKL_CHANNEL_RIGHT = 1, + SKL_CHANNEL_CENTER = 2, + SKL_CHANNEL_LEFT_SURROUND = 3, + SKL_CHANNEL_CENTER_SURROUND = 3, + SKL_CHANNEL_RIGHT_SURROUND = 4, + SKL_CHANNEL_LFE = 7, + SKL_CHANNEL_INVALID = 0xF, +}; + +enum skl_bitdepth { + SKL_DEPTH_8BIT = 8, + SKL_DEPTH_16BIT = 16, + SKL_DEPTH_24BIT = 24, + SKL_DEPTH_32BIT = 32, + SKL_DEPTH_INVALID +}; + +enum skl_interleaving { + /* [s1_ch1...s1_chN,...,sM_ch1...sM_chN] */ + SKL_INTERLEAVING_PER_CHANNEL = 0, + /* [s1_ch1...sM_ch1,...,s1_chN...sM_chN] */ + SKL_INTERLEAVING_PER_SAMPLE = 1, +}; + +enum skl_s_freq { + SKL_FS_8000 = 8000, + SKL_FS_11025 = 11025, + SKL_FS_12000 = 12000, + SKL_FS_16000 = 16000, + SKL_FS_22050 = 22050, + SKL_FS_24000 = 24000, + SKL_FS_32000 = 32000, + SKL_FS_44100 = 44100, + SKL_FS_48000 = 48000, + SKL_FS_64000 = 64000, + SKL_FS_88200 = 88200, + SKL_FS_96000 = 96000, + SKL_FS_128000 = 128000, + SKL_FS_176400 = 176400, + SKL_FS_192000 = 192000, + SKL_FS_INVALID +}; + +enum skl_widget_type { + SKL_WIDGET_VMIXER = 1, + SKL_WIDGET_MIXER = 2, + SKL_WIDGET_PGA = 3, + SKL_WIDGET_MUX = 4 +}; + +struct skl_audio_data_format { + enum skl_s_freq s_freq; + enum skl_bitdepth bit_depth; + u32 channel_map; + enum skl_ch_cfg ch_cfg; + enum skl_interleaving interleaving; + u8 number_of_channels; + u8 valid_bit_depth; + u8 sample_type; + u8 reserved[1]; +} __packed; + +struct skl_base_cfg { + u32 cps; + u32 ibs; + u32 obs; + u32 is_pages; + struct skl_audio_data_format audio_fmt; +}; + +struct skl_cpr_gtw_cfg { + u32 node_id; + u32 dma_buffer_size; + u32 config_length; + /* not mandatory; required only for DMIC/I2S */ + u32 config_data[1]; +} __packed; + +struct skl_cpr_cfg { + struct skl_base_cfg base_cfg; + struct skl_audio_data_format out_fmt; + u32 cpr_feature_mask; + struct skl_cpr_gtw_cfg gtw_cfg; +} __packed; + + +struct skl_src_module_cfg { + struct skl_base_cfg base_cfg; + enum skl_s_freq src_cfg; +} __packed; + +struct skl_up_down_mixer_cfg { + struct skl_base_cfg base_cfg; + enum skl_ch_cfg out_ch_cfg; + /* This should be set to 1 if user coefficients are required */ + u32 coeff_sel; + /* Pass the user coeff in this array */ + s32 coeff[UP_DOWN_MIXER_MAX_COEFF]; +} __packed; + +enum skl_dma_type { + SKL_DMA_HDA_HOST_OUTPUT_CLASS = 0, + SKL_DMA_HDA_HOST_INPUT_CLASS = 1, + SKL_DMA_HDA_HOST_INOUT_CLASS = 2, + SKL_DMA_HDA_LINK_OUTPUT_CLASS = 8, + SKL_DMA_HDA_LINK_INPUT_CLASS = 9, + SKL_DMA_HDA_LINK_INOUT_CLASS = 0xA, + SKL_DMA_DMIC_LINK_INPUT_CLASS = 0xB, + SKL_DMA_I2S_LINK_OUTPUT_CLASS = 0xC, + SKL_DMA_I2S_LINK_INPUT_CLASS = 0xD, +}; + +union skl_ssp_dma_node { + u8 val; + struct { + u8 dual_mono:1; + u8 time_slot:3; + u8 i2s_instance:4; + } dma_node; +}; + +union skl_connector_node_id { + u32 val; + struct { + u32 vindex:8; + u32 dma_type:4; + u32 rsvd:20; + } node; +}; + +struct skl_module_fmt { + u32 channels; + u32 s_freq; + u32 bit_depth; + u32 valid_bit_depth; + u32 ch_cfg; +}; + +struct skl_module_inst_id { + u32 module_id; + u32 instance_id; +}; + +struct skl_module_pin { + struct skl_module_inst_id id; + u8 pin_index; + bool is_dynamic; + bool in_use; +}; + +struct skl_specific_cfg { + u32 caps_size; + u32 *caps; +}; + +enum skl_pipe_state { + SKL_PIPE_INVALID = 0, + SKL_PIPE_CREATED = 1, + SKL_PIPE_PAUSED = 2, + SKL_PIPE_STARTED = 3 +}; + +struct skl_pipe_module { + struct snd_soc_dapm_widget *w; + struct list_head node; +}; + +struct skl_pipe_params { + u8 host_dma_id; + u8 link_dma_id; + u32 ch; + u32 s_freq; + u32 s_fmt; + u8 linktype; + int stream; +}; + +struct skl_pipe { + u8 ppl_id; + u8 pipe_priority; + u16 conn_type; + u32 memory_pages; + struct skl_pipe_params *p_params; + enum skl_pipe_state state; + struct list_head w_list; +}; + +enum skl_module_state { + SKL_MODULE_UNINIT = 0, + SKL_MODULE_INIT_DONE = 1, + SKL_MODULE_LOADED = 2, + SKL_MODULE_UNLOADED = 3, + SKL_MODULE_BIND_DONE = 4 +}; + +struct skl_module_cfg { + struct skl_module_inst_id id; + struct skl_module_fmt in_fmt; + struct skl_module_fmt out_fmt; + u8 max_in_queue; + u8 max_out_queue; + u8 in_queue_mask; + u8 out_queue_mask; + u8 in_queue; + u8 out_queue; + u32 mcps; + u32 ibs; + u32 obs; + u8 is_loadable; + u8 core_id; + u8 dev_type; + u8 dma_id; + u8 time_slot; + u32 params_fixup; + u32 converter; + u32 vbus_id; + struct skl_module_pin *m_in_pin; + struct skl_module_pin *m_out_pin; + enum skl_module_type m_type; + enum skl_hw_conn_type hw_conn_type; + enum skl_module_state m_state; + struct skl_pipe *pipe; + struct skl_specific_cfg formats_config; +}; + +int skl_create_pipeline(struct skl_sst *ctx, struct skl_pipe *pipe); + +int skl_run_pipe(struct skl_sst *ctx, struct skl_pipe *pipe); + +int skl_pause_pipe(struct skl_sst *ctx, struct skl_pipe *pipe); + +int skl_delete_pipe(struct skl_sst *ctx, struct skl_pipe *pipe); + +int skl_stop_pipe(struct skl_sst *ctx, struct skl_pipe *pipe); + +int skl_init_module(struct skl_sst *ctx, struct skl_module_cfg *module_config, + char *param); + +int skl_bind_modules(struct skl_sst *ctx, struct skl_module_cfg + *src_module, struct skl_module_cfg *dst_module); + +int skl_unbind_modules(struct skl_sst *ctx, struct skl_module_cfg + *src_module, struct skl_module_cfg *dst_module); + +enum skl_bitdepth skl_get_bit_depth(int params); +#endif diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h new file mode 100644 index 0000000..a506898 --- /dev/null +++ b/sound/soc/intel/skylake/skl-tplg-interface.h @@ -0,0 +1,88 @@ +/* + * skl-tplg-interface.h - Intel DSP FW private data interface + * + * Copyright (C) 2015 Intel Corp + * Author: Jeeja KP <jeeja.kp@intel.com> + * Nilofer, Samreen <samreen.nilofer@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as 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. + */ + +#ifndef __HDA_TPLG_INTERFACE_H__ +#define __HDA_TPLG_INTERFACE_H__ + +/** + * enum skl_ch_cfg - channel configuration + * + * @SKL_CH_CFG_MONO: One channel only + * @SKL_CH_CFG_STEREO: L & R + * @SKL_CH_CFG_2_1: L, R & LFE + * @SKL_CH_CFG_3_0: L, C & R + * @SKL_CH_CFG_3_1: L, C, R & LFE + * @SKL_CH_CFG_QUATRO: L, R, Ls & Rs + * @SKL_CH_CFG_4_0: L, C, R & Cs + * @SKL_CH_CFG_5_0: L, C, R, Ls & Rs + * @SKL_CH_CFG_5_1: L, C, R, Ls, Rs & LFE + * @SKL_CH_CFG_DUAL_MONO: One channel replicated in two + * @SKL_CH_CFG_I2S_DUAL_STEREO_0: Stereo(L,R) in 4 slots, 1st stream:[ L, R, -, - ] + * @SKL_CH_CFG_I2S_DUAL_STEREO_1: Stereo(L,R) in 4 slots, 2nd stream:[ -, -, L, R ] + * @SKL_CH_CFG_INVALID: Invalid + */ +enum skl_ch_cfg { + SKL_CH_CFG_MONO = 0, + SKL_CH_CFG_STEREO = 1, + SKL_CH_CFG_2_1 = 2, + SKL_CH_CFG_3_0 = 3, + SKL_CH_CFG_3_1 = 4, + SKL_CH_CFG_QUATRO = 5, + SKL_CH_CFG_4_0 = 6, + SKL_CH_CFG_5_0 = 7, + SKL_CH_CFG_5_1 = 8, + SKL_CH_CFG_DUAL_MONO = 9, + SKL_CH_CFG_I2S_DUAL_STEREO_0 = 10, + SKL_CH_CFG_I2S_DUAL_STEREO_1 = 11, + SKL_CH_CFG_INVALID +}; + +enum skl_module_type { + SKL_MODULE_TYPE_MIXER = 0, + SKL_MODULE_TYPE_COPIER, + SKL_MODULE_TYPE_UPDWMIX, + SKL_MODULE_TYPE_SRCINT +}; + +enum skl_core_affinity { + SKL_AFFINITY_CORE_0 = 0, + SKL_AFFINITY_CORE_1, + SKL_AFFINITY_CORE_MAX +}; + +enum skl_pipe_conn_type { + SKL_PIPE_CONN_TYPE_NONE = 0, + SKL_PIPE_CONN_TYPE_FE, + SKL_PIPE_CONN_TYPE_BE +}; + +enum skl_hw_conn_type { + SKL_CONN_NONE = 0, + SKL_CONN_SOURCE = 1, + SKL_CONN_SINK = 2 +}; + +enum skl_dev_type { + SKL_DEVICE_BT = 0x0, + SKL_DEVICE_DMIC = 0x1, + SKL_DEVICE_I2S = 0x2, + SKL_DEVICE_SLIMBUS = 0x3, + SKL_DEVICE_HDALINK = 0x4, + SKL_DEVICE_NONE +}; +#endif diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c new file mode 100644 index 0000000..348d094 --- /dev/null +++ b/sound/soc/intel/skylake/skl.c @@ -0,0 +1,536 @@ +/* + * skl.c - Implementation of ASoC Intel SKL HD Audio driver + * + * Copyright (C) 2014-2015 Intel Corp + * Author: Jeeja KP <jeeja.kp@intel.com> + * + * Derived mostly from Intel HDA driver with following copyrights: + * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de> + * PeiSen Hou <pshou@realtek.com.tw> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/pm_runtime.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include "skl.h" + +/* + * initialize the PCI registers + */ +static void skl_update_pci_byte(struct pci_dev *pci, unsigned int reg, + unsigned char mask, unsigned char val) +{ + unsigned char data; + + pci_read_config_byte(pci, reg, &data); + data &= ~mask; + data |= (val & mask); + pci_write_config_byte(pci, reg, data); +} + +static void skl_init_pci(struct skl *skl) +{ + struct hdac_ext_bus *ebus = &skl->ebus; + + /* + * Clear bits 0-2 of PCI register TCSEL (at offset 0x44) + * TCSEL == Traffic Class Select Register, which sets PCI express QOS + * Ensuring these bits are 0 clears playback static on some HD Audio + * codecs. + * The PCI register TCSEL is defined in the Intel manuals. + */ + dev_dbg(ebus_to_hbus(ebus)->dev, "Clearing TCSEL\n"); + skl_update_pci_byte(skl->pci, AZX_PCIREG_TCSEL, 0x07, 0); +} + +/* called from IRQ */ +static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr) +{ + snd_pcm_period_elapsed(hstr->substream); +} + +static irqreturn_t skl_interrupt(int irq, void *dev_id) +{ + struct hdac_ext_bus *ebus = dev_id; + struct hdac_bus *bus = ebus_to_hbus(ebus); + u32 status; + + if (!pm_runtime_active(bus->dev)) + return IRQ_NONE; + + spin_lock(&bus->reg_lock); + + status = snd_hdac_chip_readl(bus, INTSTS); + if (status == 0 || status == 0xffffffff) { + spin_unlock(&bus->reg_lock); + return IRQ_NONE; + } + + /* clear rirb int */ + status = snd_hdac_chip_readb(bus, RIRBSTS); + if (status & RIRB_INT_MASK) { + if (status & RIRB_INT_RESPONSE) + snd_hdac_bus_update_rirb(bus); + snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); + } + + spin_unlock(&bus->reg_lock); + + return snd_hdac_chip_readl(bus, INTSTS) ? IRQ_WAKE_THREAD : IRQ_HANDLED; +} + +static irqreturn_t skl_threaded_handler(int irq, void *dev_id) +{ + struct hdac_ext_bus *ebus = dev_id; + struct hdac_bus *bus = ebus_to_hbus(ebus); + u32 status; + + status = snd_hdac_chip_readl(bus, INTSTS); + + snd_hdac_bus_handle_stream_irq(bus, status, skl_stream_update); + + return IRQ_HANDLED; +} + +static int skl_acquire_irq(struct hdac_ext_bus *ebus, int do_disconnect) +{ + struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = ebus_to_hbus(ebus); + int ret; + + ret = request_threaded_irq(skl->pci->irq, skl_interrupt, + skl_threaded_handler, + IRQF_SHARED, + KBUILD_MODNAME, ebus); + if (ret) { + dev_err(bus->dev, + "unable to grab IRQ %d, disabling device\n", + skl->pci->irq); + return ret; + } + + bus->irq = skl->pci->irq; + pci_intx(skl->pci, 1); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +/* + * power management + */ +static int skl_suspend(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct hdac_ext_bus *ebus = pci_get_drvdata(pci); + struct hdac_bus *bus = ebus_to_hbus(ebus); + + snd_hdac_bus_stop_chip(bus); + snd_hdac_bus_enter_link_reset(bus); + + return 0; +} + +static int skl_resume(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct hdac_ext_bus *ebus = pci_get_drvdata(pci); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *hda = ebus_to_skl(ebus); + + skl_init_pci(hda); + + snd_hdac_bus_init_chip(bus, 1); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +#ifdef CONFIG_PM +static int skl_runtime_suspend(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct hdac_ext_bus *ebus = pci_get_drvdata(pci); + struct hdac_bus *bus = ebus_to_hbus(ebus); + + dev_dbg(bus->dev, "in %s\n", __func__); + + /* enable controller wake up event */ + snd_hdac_chip_updatew(bus, WAKEEN, 0, STATESTS_INT_MASK); + + snd_hdac_bus_stop_chip(bus); + snd_hdac_bus_enter_link_reset(bus); + + return 0; +} + +static int skl_runtime_resume(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct hdac_ext_bus *ebus = pci_get_drvdata(pci); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct skl *hda = ebus_to_skl(ebus); + int status; + + dev_dbg(bus->dev, "in %s\n", __func__); + + /* Read STATESTS before controller reset */ + status = snd_hdac_chip_readw(bus, STATESTS); + + skl_init_pci(hda); + snd_hdac_bus_init_chip(bus, true); + /* disable controller Wake Up event */ + snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, 0); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops skl_pm = { + SET_SYSTEM_SLEEP_PM_OPS(skl_suspend, skl_resume) + SET_RUNTIME_PM_OPS(skl_runtime_suspend, skl_runtime_resume, NULL) +}; + +/* + * destructor + */ +static int skl_free(struct hdac_ext_bus *ebus) +{ + struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = ebus_to_hbus(ebus); + + skl->init_failed = 1; /* to be sure */ + + snd_hdac_ext_stop_streams(ebus); + + if (bus->irq >= 0) + free_irq(bus->irq, (void *)bus); + if (bus->remap_addr) + iounmap(bus->remap_addr); + + snd_hdac_bus_free_stream_pages(bus); + snd_hdac_stream_free_all(ebus); + snd_hdac_link_free_all(ebus); + pci_release_regions(skl->pci); + pci_disable_device(skl->pci); + + snd_hdac_ext_bus_exit(ebus); + + return 0; +} + +static int skl_dmic_device_register(struct skl *skl) +{ + struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct platform_device *pdev; + int ret; + + /* SKL has one dmic port, so allocate dmic device for this */ + pdev = platform_device_alloc("dmic-codec", -1); + if (!pdev) { + dev_err(bus->dev, "failed to allocate dmic device\n"); + return -ENOMEM; + } + + ret = platform_device_add(pdev); + if (ret) { + dev_err(bus->dev, "failed to add dmic device: %d\n", ret); + platform_device_put(pdev); + return ret; + } + skl->dmic_dev = pdev; + + return 0; +} + +static void skl_dmic_device_unregister(struct skl *skl) +{ + if (skl->dmic_dev) + platform_device_unregister(skl->dmic_dev); +} + +/* + * Probe the given codec address + */ +static int probe_codec(struct hdac_ext_bus *ebus, int addr) +{ + struct hdac_bus *bus = ebus_to_hbus(ebus); + unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | + (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; + unsigned int res; + + mutex_lock(&bus->cmd_mutex); + snd_hdac_bus_send_cmd(bus, cmd); + snd_hdac_bus_get_response(bus, addr, &res); + mutex_unlock(&bus->cmd_mutex); + if (res == -1) + return -EIO; + dev_dbg(bus->dev, "codec #%d probed OK\n", addr); + + return snd_hdac_ext_bus_device_init(ebus, addr); +} + +/* Codec initialization */ +static int skl_codec_create(struct hdac_ext_bus *ebus) +{ + struct hdac_bus *bus = ebus_to_hbus(ebus); + int c, max_slots; + + max_slots = HDA_MAX_CODECS; + + /* First try to probe all given codec slots */ + for (c = 0; c < max_slots; c++) { + if ((bus->codec_mask & (1 << c))) { + if (probe_codec(ebus, c) < 0) { + /* + * Some BIOSen give you wrong codec addresses + * that don't exist + */ + dev_warn(bus->dev, + "Codec #%d probe error; disabling it...\n", c); + bus->codec_mask &= ~(1 << c); + /* + * More badly, accessing to a non-existing + * codec often screws up the controller bus, + * and disturbs the further communications. + * Thus if an error occurs during probing, + * better to reset the controller bus to get + * back to the sanity state. + */ + snd_hdac_bus_stop_chip(bus); + snd_hdac_bus_init_chip(bus, true); + } + } + } + + return 0; +} + +static const struct hdac_bus_ops bus_core_ops = { + .command = snd_hdac_bus_send_cmd, + .get_response = snd_hdac_bus_get_response, +}; + +/* + * constructor + */ +static int skl_create(struct pci_dev *pci, + const struct hdac_io_ops *io_ops, + struct skl **rskl) +{ + struct skl *skl; + struct hdac_ext_bus *ebus; + + int err; + + *rskl = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + skl = devm_kzalloc(&pci->dev, sizeof(*skl), GFP_KERNEL); + if (!skl) { + pci_disable_device(pci); + return -ENOMEM; + } + ebus = &skl->ebus; + snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); + ebus->bus.use_posbuf = 1; + skl->pci = pci; + + ebus->bus.bdl_pos_adj = 0; + + *rskl = skl; + + return 0; +} + +static int skl_first_init(struct hdac_ext_bus *ebus) +{ + struct skl *skl = ebus_to_skl(ebus); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct pci_dev *pci = skl->pci; + int err; + unsigned short gcap; + int cp_streams, pb_streams, start_idx; + + err = pci_request_regions(pci, "Skylake HD audio"); + if (err < 0) + return err; + + bus->addr = pci_resource_start(pci, 0); + bus->remap_addr = pci_ioremap_bar(pci, 0); + if (bus->remap_addr == NULL) { + dev_err(bus->dev, "ioremap error\n"); + return -ENXIO; + } + + snd_hdac_ext_bus_parse_capabilities(ebus); + + if (skl_acquire_irq(ebus, 0) < 0) + return -EBUSY; + + pci_set_master(pci); + synchronize_irq(bus->irq); + + gcap = snd_hdac_chip_readw(bus, GCAP); + dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); + + /* allow 64bit DMA address if supported by H/W */ + if (!dma_set_mask(bus->dev, DMA_BIT_MASK(64))) { + dma_set_coherent_mask(bus->dev, DMA_BIT_MASK(64)); + } else { + dma_set_mask(bus->dev, DMA_BIT_MASK(32)); + dma_set_coherent_mask(bus->dev, DMA_BIT_MASK(32)); + } + + /* read number of streams from GCAP register */ + cp_streams = (gcap >> 8) & 0x0f; + pb_streams = (gcap >> 12) & 0x0f; + + if (!pb_streams && !cp_streams) + return -EIO; + + ebus->num_streams = cp_streams + pb_streams; + + /* initialize streams */ + snd_hdac_ext_stream_init_all + (ebus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); + start_idx = cp_streams; + snd_hdac_ext_stream_init_all + (ebus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); + + err = snd_hdac_bus_alloc_stream_pages(bus); + if (err < 0) + return err; + + /* initialize chip */ + skl_init_pci(skl); + + snd_hdac_bus_init_chip(bus, true); + + /* codec detection */ + if (!bus->codec_mask) { + dev_err(bus->dev, "no codecs found!\n"); + return -ENODEV; + } + + return 0; +} + +static int skl_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct skl *skl; + struct hdac_ext_bus *ebus = NULL; + struct hdac_bus *bus = NULL; + int err; + + /* we use ext core ops, so provide NULL for ops here */ + err = skl_create(pci, NULL, &skl); + if (err < 0) + return err; + + ebus = &skl->ebus; + bus = ebus_to_hbus(ebus); + + err = skl_first_init(ebus); + if (err < 0) + goto out_free; + + pci_set_drvdata(skl->pci, ebus); + + /* check if dsp is there */ + if (ebus->ppcap) { + /* TODO register with dsp IPC */ + dev_dbg(bus->dev, "Register dsp\n"); + } + + if (ebus->mlcap) + snd_hdac_ext_bus_get_ml_capabilities(ebus); + + /* create device for soc dmic */ + err = skl_dmic_device_register(skl); + if (err < 0) + goto out_free; + + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) + goto out_dmic_free; + + /* create codec instances */ + err = skl_codec_create(ebus); + if (err < 0) + goto out_unregister; + + /*configure PM */ + pm_runtime_set_autosuspend_delay(bus->dev, SKL_SUSPEND_DELAY); + pm_runtime_use_autosuspend(bus->dev); + pm_runtime_put_noidle(bus->dev); + pm_runtime_allow(bus->dev); + + return 0; + +out_unregister: + skl_platform_unregister(bus->dev); +out_dmic_free: + skl_dmic_device_unregister(skl); +out_free: + skl->init_failed = 1; + skl_free(ebus); + + return err; +} + +static void skl_remove(struct pci_dev *pci) +{ + struct hdac_ext_bus *ebus = pci_get_drvdata(pci); + struct skl *skl = ebus_to_skl(ebus); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + pci_dev_put(pci); + skl_platform_unregister(&pci->dev); + skl_dmic_device_unregister(skl); + skl_free(ebus); + dev_set_drvdata(&pci->dev, NULL); +} + +/* PCI IDs */ +static const struct pci_device_id skl_ids[] = { + /* Sunrise Point-LP */ + { PCI_DEVICE(0x8086, 0x9d70), 0}, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, skl_ids); + +/* pci_driver definition */ +static struct pci_driver skl_driver = { + .name = KBUILD_MODNAME, + .id_table = skl_ids, + .probe = skl_probe, + .remove = skl_remove, + .driver = { + .pm = &skl_pm, + }, +}; +module_pci_driver(skl_driver); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Skylake ASoC HDA driver"); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h new file mode 100644 index 0000000..f7fdbb0 --- /dev/null +++ b/sound/soc/intel/skylake/skl.h @@ -0,0 +1,84 @@ +/* + * skl.h - HD Audio skylake defintions. + * + * Copyright (C) 2015 Intel Corp + * Author: Jeeja KP <jeeja.kp@intel.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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SOUND_SOC_SKL_H +#define __SOUND_SOC_SKL_H + +#include <sound/hda_register.h> +#include <sound/hdaudio_ext.h> +#include "skl-nhlt.h" + +#define SKL_SUSPEND_DELAY 2000 + +/* Vendor Specific Registers */ +#define AZX_REG_VS_EM1 0x1000 +#define AZX_REG_VS_INRC 0x1004 +#define AZX_REG_VS_OUTRC 0x1008 +#define AZX_REG_VS_FIFOTRK 0x100C +#define AZX_REG_VS_FIFOTRK2 0x1010 +#define AZX_REG_VS_EM2 0x1030 +#define AZX_REG_VS_EM3L 0x1038 +#define AZX_REG_VS_EM3U 0x103C +#define AZX_REG_VS_EM4L 0x1040 +#define AZX_REG_VS_EM4U 0x1044 +#define AZX_REG_VS_LTRC 0x1048 +#define AZX_REG_VS_D0I3C 0x104A +#define AZX_REG_VS_PCE 0x104B +#define AZX_REG_VS_L2MAGC 0x1050 +#define AZX_REG_VS_L2LAHPT 0x1054 +#define AZX_REG_VS_SDXDPIB_XBASE 0x1084 +#define AZX_REG_VS_SDXDPIB_XINTERVAL 0x20 +#define AZX_REG_VS_SDXEFIFOS_XBASE 0x1094 +#define AZX_REG_VS_SDXEFIFOS_XINTERVAL 0x20 + +struct skl { + struct hdac_ext_bus ebus; + struct pci_dev *pci; + + unsigned int init_failed:1; /* delayed init failed */ + struct platform_device *dmic_dev; + + void __iomem *nhlt; /* nhlt ptr */ + struct skl_sst *skl_sst; /* sst skl ctx */ +}; + +#define skl_to_ebus(s) (&(s)->ebus) +#define ebus_to_skl(sbus) \ + container_of(sbus, struct skl, sbus) + +/* to pass dai dma data */ +struct skl_dma_params { + u32 format; + u8 stream_tag; +}; + +int skl_platform_unregister(struct device *dev); +int skl_platform_register(struct device *dev); + +void __iomem *skl_nhlt_init(struct device *dev); +void skl_nhlt_free(void __iomem *addr); +struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, + u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); + +int skl_init_dsp(struct skl *skl); +void skl_free_dsp(struct skl *skl); +int skl_suspend_dsp(struct skl *skl); +int skl_resume_dsp(struct skl *skl); +#endif /* __SOUND_SOC_SKL_H */ diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 4cf2245..dbfdfe9 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -148,10 +148,14 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) dram = mv_mbus_dram_info(); addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (priv->substream_play) + return -EBUSY; priv->substream_play = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { + if (priv->substream_rec) + return -EBUSY; priv->substream_rec = substream; kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_RECORD_WIN, addr, dram); diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 4d44b58..fee0c74 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -103,7 +103,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Playback", .stream_name = "MAX98090 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -114,7 +113,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Capture", .stream_name = "MAX98090 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -125,7 +123,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codec_dai_name = "HiFi", .init = mt8173_max98090_init, @@ -139,6 +136,7 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { static struct snd_soc_card mt8173_max98090_card = { .name = "mt8173-max98090", + .owner = THIS_MODULE, .dai_link = mt8173_max98090_dais, .num_links = ARRAY_SIZE(mt8173_max98090_dais), .controls = mt8173_max98090_controls, @@ -152,9 +150,21 @@ static struct snd_soc_card mt8173_max98090_card = { static int mt8173_max98090_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_max98090_card; - struct device_node *codec_node; + struct device_node *codec_node, *platform_node; int ret, i; + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + for (i = 0; i < card->num_links; i++) { + if (mt8173_max98090_dais[i].platform_name) + continue; + mt8173_max98090_dais[i].platform_of_node = platform_node; + } + codec_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); if (!codec_node) { diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 0940553..c1f8803 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -138,7 +138,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Playback", .stream_name = "rt5650_rt5676 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -149,7 +148,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Capture", .stream_name = "rt5650_rt5676 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -161,7 +159,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codecs = mt8173_rt5650_rt5676_codecs, .num_codecs = 2, @@ -194,6 +191,7 @@ static struct snd_soc_codec_conf mt8173_rt5650_rt5676_codec_conf[] = { static struct snd_soc_card mt8173_rt5650_rt5676_card = { .name = "mtk-rt5650-rt5676", + .owner = THIS_MODULE, .dai_link = mt8173_rt5650_rt5676_dais, .num_links = ARRAY_SIZE(mt8173_rt5650_rt5676_dais), .codec_conf = mt8173_rt5650_rt5676_codec_conf, @@ -209,7 +207,21 @@ static struct snd_soc_card mt8173_rt5650_rt5676_card = { static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5676_card; - int ret; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5676_dais[i].platform_name) + continue; + mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node; + } mt8173_rt5650_rt5676_codecs[0].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index cc228db..9863da7 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -1199,6 +1199,8 @@ err_pm_disable: static int mtk_afe_pcm_dev_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + mtk_afe_runtime_suspend(&pdev->dev); snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); return 0; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 076bec6..732e749 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -154,8 +154,7 @@ static const struct snd_soc_dapm_route omap3pandora_map[] = { static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; /* All TWL4030 output pins are floating */ snd_soc_dapm_nc_pin(dapm, "EARPIECE"); @@ -174,8 +173,7 @@ static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd) static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; /* Not comnnected */ snd_soc_dapm_nc_pin(dapm, "HSMIC"); diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index e181826..58bae8e 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -14,3 +14,22 @@ config SND_SOC_ROCKCHIP_I2S Say Y or M if you want to add support for I2S driver for Rockchip I2S device. The device supports upto maximum of 8 channels each for play and record. + +config SND_SOC_ROCKCHIP_MAX98090 + tristate "ASoC support for Rockchip boards using a MAX98090 codec" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98090 + select SND_SOC_TS3A227E + help + Say Y or M here if you want to add support for SoC audio on Rockchip + boards using the MAX98090 codec, such as Veyron. + +config SND_SOC_ROCKCHIP_RT5645 + tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_RT5645 + help + Say Y or M here if you want to add support for SoC audio on Rockchip + boards using the RT5645/RT5650 codec, such as Veyron. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index b921909..1bc1dc3 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -2,3 +2,9 @@ snd-soc-i2s-objs := rockchip_i2s.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o + +snd-soc-rockchip-max98090-objs := rockchip_max98090.o +snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o + +obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index acb5be5..b936102 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -483,16 +483,14 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pcm_register; + return ret; } return 0; -err_pcm_register: - snd_dmaengine_pcm_unregister(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) i2s_runtime_suspend(&pdev->dev); @@ -512,8 +510,6 @@ static int rockchip_i2s_remove(struct platform_device *pdev) clk_disable_unprepare(i2s->mclk); clk_disable_unprepare(i2s->hclk); - snd_dmaengine_pcm_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c new file mode 100644 index 0000000..cc26f81 --- /dev/null +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -0,0 +1,237 @@ +/* + * Rockchip machine ASoC driver for boards using a MAX90809 CODEC. + * + * Copyright (c) 2014, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "rockchip_i2s.h" +#include "../codecs/ts3a227e.h" + +#define DRV_NAME "rockchip-snd-max98090" + +static struct snd_soc_jack headset_jack; +static struct snd_soc_jack_pin headset_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + }, +}; + +static const struct snd_soc_dapm_widget rk_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route rk_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"IN34", NULL, "MICBIAS"}, + {"MICBIAS", NULL, "Headset Mic"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Speaker", NULL, "SPKL"}, + {"Speaker", NULL, "SPKR"}, +}; + +static const struct snd_kcontrol_new rk_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int rk_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int ret = 0; + 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 mclk; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = 12288000; + break; + case 44100: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + return ret; +} + +static int rk_init(struct snd_soc_pcm_runtime *runtime) +{ + /* Enable Headset and 4 Buttons Jack detection */ + return snd_soc_card_jack_new(runtime->card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &headset_jack, + headset_jack_pins, + ARRAY_SIZE(headset_jack_pins)); +} + +static int rk_98090_headset_init(struct snd_soc_component *component) +{ + return ts3a227e_enable_jack_detect(component, &headset_jack); +} + +static struct snd_soc_ops rk_aif1_ops = { + .hw_params = rk_aif1_hw_params, +}; + +static struct snd_soc_aux_dev rk_98090_headset_dev = { + .name = "Headset Chip", + .init = rk_98090_headset_init, +}; + +static struct snd_soc_dai_link rk_dailink = { + .name = "max98090", + .stream_name = "Audio", + .codec_dai_name = "HiFi", + .init = rk_init, + .ops = &rk_aif1_ops, + /* set max98090 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_card_rk = { + .name = "ROCKCHIP-I2S", + .owner = THIS_MODULE, + .dai_link = &rk_dailink, + .num_links = 1, + .aux_dev = &rk_98090_headset_dev, + .num_aux_devs = 1, + .dapm_widgets = rk_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk_dapm_widgets), + .dapm_routes = rk_audio_map, + .num_dapm_routes = ARRAY_SIZE(rk_audio_map), + .controls = rk_mc_controls, + .num_controls = ARRAY_SIZE(rk_mc_controls), +}; + +static int snd_rk_mc_probe(struct platform_device *pdev) +{ + int ret = 0; + struct snd_soc_card *card = &snd_soc_card_rk; + struct device_node *np = pdev->dev.of_node; + + /* register the soc card */ + card->dev = &pdev->dev; + + rk_dailink.codec_of_node = of_parse_phandle(np, + "rockchip,audio-codec", 0); + if (!rk_dailink.codec_of_node) { + dev_err(&pdev->dev, + "Property 'rockchip,audio-codec' missing or invalid\n"); + return -EINVAL; + } + + rk_dailink.cpu_of_node = of_parse_phandle(np, + "rockchip,i2s-controller", 0); + if (!rk_dailink.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'rockchip,i2s-controller' missing or invalid\n"); + return -EINVAL; + } + + rk_dailink.platform_of_node = rk_dailink.cpu_of_node; + + rk_98090_headset_dev.codec_of_node = of_parse_phandle(np, + "rockchip,headset-codec", 0); + if (!rk_98090_headset_dev.codec_of_node) { + dev_err(&pdev->dev, + "Property 'rockchip,headset-codec' missing/invalid\n"); + return -EINVAL; + } + + ret = snd_soc_of_parse_card_name(card, "rockchip,model"); + if (ret) { + dev_err(&pdev->dev, + "Soc parse card name failed %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err(&pdev->dev, + "Soc register card failed %d\n", ret); + return ret; + } + + return ret; +} + +static const struct of_device_id rockchip_max98090_of_match[] = { + { .compatible = "rockchip,rockchip-audio-max98090", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, rockchip_max98090_of_match); + +static struct platform_driver snd_rk_mc_driver = { + .probe = snd_rk_mc_probe, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = rockchip_max98090_of_match, + }, +}; + +module_platform_driver(snd_rk_mc_driver); + +MODULE_AUTHOR("jianqun <jay.xu@rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip max98090 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c new file mode 100644 index 0000000..0940279 --- /dev/null +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -0,0 +1,226 @@ +/* + * Rockchip machine ASoC driver for boards using a RT5645/RT5650 CODEC. + * + * Copyright (c) 2015, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" + +#define DRV_NAME "rockchip-snd-rt5645" + +static struct snd_soc_jack headset_jack; + +/* Jack detect via rt5645 driver. */ +extern int rt5645_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, + struct snd_soc_jack *btn_jack); + +static const struct snd_soc_dapm_widget rk_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rk_audio_map[] = { + /* Input Lines */ + {"DMIC L2", NULL, "Int Mic"}, + {"DMIC R2", NULL, "Int Mic"}, + {"RECMIXL", NULL, "Headset Mic"}, + {"RECMIXR", NULL, "Headset Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPOR"}, + {"Headphones", NULL, "HPOL"}, + {"Speakers", NULL, "SPOL"}, + {"Speakers", NULL, "SPOR"}, +}; + +static const struct snd_kcontrol_new rk_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rk_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int ret = 0; + 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 mclk; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = 12288000; + break; + case 44100: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + return ret; +} + +static int rk_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &headset_jack, NULL, 0); + if (!ret) { + dev_err(card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + return rt5645_set_jack_detect(runtime->codec, + &headset_jack, + &headset_jack, + &headset_jack); +} + +static struct snd_soc_ops rk_aif1_ops = { + .hw_params = rk_aif1_hw_params, +}; + +static struct snd_soc_dai_link rk_dailink = { + .name = "rt5645", + .stream_name = "rt5645 PCM", + .codec_dai_name = "rt5645-aif1", + .init = rk_init, + .ops = &rk_aif1_ops, + /* set rt5645 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_card_rk = { + .name = "I2S-RT5650", + .owner = THIS_MODULE, + .dai_link = &rk_dailink, + .num_links = 1, + .dapm_widgets = rk_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk_dapm_widgets), + .dapm_routes = rk_audio_map, + .num_dapm_routes = ARRAY_SIZE(rk_audio_map), + .controls = rk_mc_controls, + .num_controls = ARRAY_SIZE(rk_mc_controls), +}; + +static int snd_rk_mc_probe(struct platform_device *pdev) +{ + int ret = 0; + struct snd_soc_card *card = &snd_soc_card_rk; + struct device_node *np = pdev->dev.of_node; + + /* register the soc card */ + card->dev = &pdev->dev; + + rk_dailink.codec_of_node = of_parse_phandle(np, + "rockchip,audio-codec", 0); + if (!rk_dailink.codec_of_node) { + dev_err(&pdev->dev, + "Property 'rockchip,audio-codec' missing or invalid\n"); + return -EINVAL; + } + + rk_dailink.cpu_of_node = of_parse_phandle(np, + "rockchip,i2s-controller", 0); + if (!rk_dailink.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'rockchip,i2s-controller' missing or invalid\n"); + return -EINVAL; + } + + rk_dailink.platform_of_node = rk_dailink.cpu_of_node; + + ret = snd_soc_of_parse_card_name(card, "rockchip,model"); + if (ret) { + dev_err(&pdev->dev, + "Soc parse card name failed %d\n", ret); + return ret; + } + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err(&pdev->dev, + "Soc register card failed %d\n", ret); + return ret; + } + + return ret; +} + +static const struct of_device_id rockchip_rt5645_of_match[] = { + { .compatible = "rockchip,rockchip-audio-rt5645", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); + +static struct platform_driver snd_rk_mc_driver = { + .probe = snd_rk_mc_probe, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = rockchip_rt5645_of_match, + }, +}; + +module_platform_driver(snd_rk_mc_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing@rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip rt5645 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 8bf2e2c..ee1fda9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -71,6 +71,7 @@ static struct snd_soc_dai_link arndale_rt5631_dai[] = { static struct snd_soc_card arndale_rt5631 = { .name = "Arndale RT5631", + .owner = THIS_MODULE, .dai_link = arndale_rt5631_dai, .num_links = ARRAY_SIZE(arndale_rt5631_dai), }; @@ -116,15 +117,6 @@ static int arndale_audio_probe(struct platform_device *pdev) return ret; } -static int arndale_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { { .compatible = "samsung,arndale-rt5631", }, { .compatible = "samsung,arndale-alc5631", }, @@ -139,7 +131,6 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), }, .probe = arndale_audio_probe, - .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 7651dc9..07ce2cf 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -56,6 +56,7 @@ static int snow_late_probe(struct snd_soc_card *card) static struct snd_soc_card snow_snd = { .name = "Snow-I2S", + .owner = THIS_MODULE, .dai_link = snow_dai, .num_links = ARRAY_SIZE(snow_dai), diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 142c066..0215c78 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1911,7 +1911,6 @@ MODULE_DEVICE_TABLE(of, fsi_of_match); static const struct platform_device_id fsi_id_table[] = { { "sh_fsi", (kernel_ulong_t)&fsi1_core }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, {}, }; MODULE_DEVICE_TABLE(platform, fsi_id_table); diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 08d7259..d40efc9 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -85,10 +85,19 @@ EXPORT_SYMBOL(snd_soc_alloc_ac97_codec); /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec + * @id: The expected device ID + * @id_mask: Mask that is applied to the device ID before comparing with @id * * Initialises AC97 codec resources for use by ad-hoc devices only. + * + * If @id is not 0 this function will reset the device, then read the ID from + * the device and check if it matches the expected ID. If it doesn't match an + * error will be returned and device will not be registered. + * + * Returns: A PTR_ERR() on failure or a valid snd_ac97 struct on success. */ -struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec, + unsigned int id, unsigned int id_mask) { struct snd_ac97 *ac97; int ret; @@ -97,13 +106,24 @@ struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) if (IS_ERR(ac97)) return ac97; - ret = device_add(&ac97->dev); - if (ret) { - put_device(&ac97->dev); - return ERR_PTR(ret); + if (id) { + ret = snd_ac97_reset(ac97, false, id, id_mask); + if (ret < 0) { + dev_err(codec->dev, "Failed to reset AC97 device: %d\n", + ret); + goto err_put_device; + } } + ret = device_add(&ac97->dev); + if (ret) + goto err_put_device; + return ac97; + +err_put_device: + put_device(&ac97->dev); + return ERR_PTR(ret); } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4a5c0..0c0ac01 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -654,10 +654,12 @@ int snd_soc_suspend(struct device *dev) /* suspend all CODECs */ list_for_each_entry(codec, &card->codec_dev_list, card_list) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + /* If there are paths active then the CODEC will be held with * bias _ON and should not be suspended. */ if (!codec->suspended) { - switch (codec->dapm.bias_level) { + switch (snd_soc_dapm_get_bias_level(dapm)) { case SND_SOC_BIAS_STANDBY: /* * If the CODEC is capable of idle @@ -665,7 +667,7 @@ int snd_soc_suspend(struct device *dev) * means it's doing something, * otherwise fall through. */ - if (codec->dapm.idle_bias_off) { + if (dapm->idle_bias_off) { dev_dbg(codec->dev, "ASoC: idle_bias_off CODEC on over suspend\n"); break; @@ -978,7 +980,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) static void soc_remove_component(struct snd_soc_component *component) { - if (!component->probed) + if (!component->card) return; /* This is a HACK and will be removed soon */ @@ -991,7 +993,7 @@ static void soc_remove_component(struct snd_soc_component *component) snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); - component->probed = 0; + component->card = NULL; module_put(component->dev->driver->owner); } @@ -1102,16 +1104,26 @@ static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_dai *dai; int ret; - if (component->probed) + if (!strcmp(component->name, "snd-soc-dummy")) return 0; - component->card = card; - dapm->card = card; - soc_set_name_prefix(card, component); + if (component->card) { + if (component->card != card) { + dev_err(component->dev, + "Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n", + card->name, component->card->name); + return -ENODEV; + } + return 0; + } if (!try_module_get(component->dev->driver->owner)) return -ENODEV; + component->card = card; + dapm->card = card; + soc_set_name_prefix(card, component); + soc_init_component_debugfs(component); if (component->dapm_widgets) { @@ -1155,7 +1167,6 @@ static int soc_probe_component(struct snd_soc_card *card, snd_soc_dapm_add_routes(dapm, component->dapm_routes, component->num_dapm_routes); - component->probed = 1; list_add(&dapm->list, &card->dapm_list); /* This is a HACK and will be removed soon */ @@ -1166,6 +1177,7 @@ static int soc_probe_component(struct snd_soc_card *card, err_probe: soc_cleanup_component_debugfs(component); + component->card = NULL; module_put(component->dev->driver->owner); return ret; @@ -1449,7 +1461,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (component && component->probed) + if (component) soc_remove_component(component); } @@ -1716,6 +1728,7 @@ card_probe_error: if (card->remove) card->remove(card); + snd_soc_dapm_free(&card->dapm); soc_cleanup_card_debugfs(card); snd_card_free(card->snd_card); @@ -2127,7 +2140,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -2651,10 +2664,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->probe = component->driver->probe; component->remove = component->driver->remove; - if (!component->dapm_ptr) - component->dapm_ptr = &component->dapm; - - dapm = component->dapm_ptr; + dapm = &component->dapm; dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; @@ -2798,6 +2808,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core * + * @dev: The device to unregister */ void snd_soc_unregister_component(struct device *dev) { @@ -2838,7 +2849,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -2877,7 +2888,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: The device for the platform + * @platform_drv: The driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -2938,7 +2950,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -3029,13 +3041,17 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: The parent device for this codec + * @codec_drv: Codec driver + * @dai_drv: The associated DAI driver + * @num_dai: Number of DAIs */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai) { + struct snd_soc_dapm_context *dapm; struct snd_soc_codec *codec; struct snd_soc_dai *dai; int ret, i; @@ -3046,7 +3062,6 @@ int snd_soc_register_codec(struct device *dev, if (codec == NULL) return -ENOMEM; - codec->component.dapm_ptr = &codec->dapm; codec->component.codec = codec; ret = snd_soc_component_initialize(&codec->component, @@ -3076,12 +3091,14 @@ int snd_soc_register_codec(struct device *dev, if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.idle_bias_off = codec_drv->idle_bias_off; - codec->dapm.suspend_bias_off = codec_drv->suspend_bias_off; + + dapm = snd_soc_codec_get_dapm(codec); + dapm->idle_bias_off = codec_drv->idle_bias_off; + dapm->suspend_bias_off = codec_drv->suspend_bias_off; if (codec_drv->seq_notifier) - codec->dapm.seq_notifier = codec_drv->seq_notifier; + dapm->seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) - codec->dapm.set_bias_level = snd_soc_codec_set_bias_level; + dapm->set_bias_level = snd_soc_codec_set_bias_level; codec->dev = dev; codec->driver = codec_drv; codec->component.val_bytes = codec_drv->reg_word_size; @@ -3128,7 +3145,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..f4bf21a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -47,6 +47,13 @@ #define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; +#define SND_SOC_DAPM_DIR_REVERSE(x) ((x == SND_SOC_DAPM_DIR_IN) ? \ + SND_SOC_DAPM_DIR_OUT : SND_SOC_DAPM_DIR_IN) + +#define snd_soc_dapm_for_each_direction(dir) \ + for ((dir) = SND_SOC_DAPM_DIR_IN; (dir) <= SND_SOC_DAPM_DIR_OUT; \ + (dir)++) + static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, const char *control, @@ -167,45 +174,59 @@ static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) } /* - * dapm_widget_invalidate_input_paths() - Invalidate the cached number of input - * paths - * @w: The widget for which to invalidate the cached number of input paths - * - * The function resets the cached number of inputs for the specified widget and - * all widgets that can be reached via outgoing paths from the widget. - * - * This function must be called if the number of input paths for a widget might - * have changed. E.g. if the source state of a widget changes or a path is added - * or activated with the widget as the sink. + * Common implementation for dapm_widget_invalidate_input_paths() and + * dapm_widget_invalidate_output_paths(). The function is inlined since the + * combined size of the two specialized functions is only marginally larger then + * the size of the generic function and at the same time the fast path of the + * specialized functions is significantly smaller than the generic function. */ -static void dapm_widget_invalidate_input_paths(struct snd_soc_dapm_widget *w) +static __always_inline void dapm_widget_invalidate_paths( + struct snd_soc_dapm_widget *w, enum snd_soc_dapm_direction dir) { - struct snd_soc_dapm_widget *sink; + enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); + struct snd_soc_dapm_widget *node; struct snd_soc_dapm_path *p; LIST_HEAD(list); dapm_assert_locked(w->dapm); - if (w->inputs == -1) + if (w->endpoints[dir] == -1) return; - w->inputs = -1; list_add_tail(&w->work_list, &list); + w->endpoints[dir] = -1; list_for_each_entry(w, &list, work_list) { - list_for_each_entry(p, &w->sinks, list_source) { + snd_soc_dapm_widget_for_each_path(w, dir, p) { if (p->is_supply || p->weak || !p->connect) continue; - sink = p->sink; - if (sink->inputs != -1) { - sink->inputs = -1; - list_add_tail(&sink->work_list, &list); + node = p->node[rdir]; + if (node->endpoints[dir] != -1) { + node->endpoints[dir] = -1; + list_add_tail(&node->work_list, &list); } } } } /* + * dapm_widget_invalidate_input_paths() - Invalidate the cached number of + * input paths + * @w: The widget for which to invalidate the cached number of input paths + * + * Resets the cached number of inputs for the specified widget and all widgets + * that can be reached via outcoming paths from the widget. + * + * This function must be called if the number of output paths for a widget might + * have changed. E.g. if the source state of a widget changes or a path is added + * or activated with the widget as the sink. + */ +static void dapm_widget_invalidate_input_paths(struct snd_soc_dapm_widget *w) +{ + dapm_widget_invalidate_paths(w, SND_SOC_DAPM_DIR_IN); +} + +/* * dapm_widget_invalidate_output_paths() - Invalidate the cached number of * output paths * @w: The widget for which to invalidate the cached number of output paths @@ -219,29 +240,7 @@ static void dapm_widget_invalidate_input_paths(struct snd_soc_dapm_widget *w) */ static void dapm_widget_invalidate_output_paths(struct snd_soc_dapm_widget *w) { - struct snd_soc_dapm_widget *source; - struct snd_soc_dapm_path *p; - LIST_HEAD(list); - - dapm_assert_locked(w->dapm); - - if (w->outputs == -1) - return; - - w->outputs = -1; - list_add_tail(&w->work_list, &list); - - list_for_each_entry(w, &list, work_list) { - list_for_each_entry(p, &w->sources, list_sink) { - if (p->is_supply || p->weak || !p->connect) - continue; - source = p->source; - if (source->outputs != -1) { - source->outputs = -1; - list_add_tail(&source->work_list, &list); - } - } - } + dapm_widget_invalidate_paths(w, SND_SOC_DAPM_DIR_OUT); } /* @@ -270,9 +269,9 @@ static void dapm_path_invalidate(struct snd_soc_dapm_path *p) * endpoints is either connected or disconnected that sum won't change, * so there is no need to re-check the path. */ - if (p->source->inputs != 0) + if (p->source->endpoints[SND_SOC_DAPM_DIR_IN] != 0) dapm_widget_invalidate_input_paths(p->sink); - if (p->sink->outputs != 0) + if (p->sink->endpoints[SND_SOC_DAPM_DIR_OUT] != 0) dapm_widget_invalidate_output_paths(p->source); } @@ -283,11 +282,11 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { - if (w->is_sink || w->is_source) { + if (w->is_ep) { dapm_mark_dirty(w, "Rechecking endpoints"); - if (w->is_sink) + if (w->is_ep & SND_SOC_DAPM_EP_SINK) dapm_widget_invalidate_output_paths(w); - if (w->is_source) + if (w->is_ep & SND_SOC_DAPM_EP_SOURCE) dapm_widget_invalidate_input_paths(w); } } @@ -358,9 +357,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked(widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } } break; @@ -389,11 +389,12 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->value = template.on_val; - data->widget = snd_soc_dapm_new_control(widget->dapm, - &template); + data->widget = snd_soc_dapm_new_control_unlocked( + widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } snd_soc_dapm_add_path(widget->dapm, data->widget, @@ -408,8 +409,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, return 0; -err_name: - kfree(name); err_data: kfree(data); return ret; @@ -418,8 +417,6 @@ err_data: static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); - if (data->widget) - kfree(data->widget->name); kfree(data->wlist); kfree(data); } @@ -896,7 +893,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { /* match name */ - list_for_each_entry(path, &w->sources, list_sink) { + snd_soc_dapm_widget_for_each_source_path(w, path) { /* mixer/mux paths name must match control name */ if (path->name != (char *)w->kcontrol_news[i].name) continue; @@ -925,18 +922,18 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) static int dapm_new_mux(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_context *dapm = w->dapm; + enum snd_soc_dapm_direction dir; struct snd_soc_dapm_path *path; - struct list_head *paths; const char *type; int ret; switch (w->id) { case snd_soc_dapm_mux: - paths = &w->sources; + dir = SND_SOC_DAPM_DIR_OUT; type = "mux"; break; case snd_soc_dapm_demux: - paths = &w->sinks; + dir = SND_SOC_DAPM_DIR_IN; type = "demux"; break; default: @@ -950,7 +947,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - if (list_empty(paths)) { + if (list_empty(&w->edges[dir])) { dev_err(dapm->dev, "ASoC: %s %s has no paths\n", type, w->name); return -EINVAL; } @@ -959,16 +956,9 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) if (ret < 0) return ret; - if (w->id == snd_soc_dapm_mux) { - list_for_each_entry(path, &w->sources, list_sink) { - if (path->name) - dapm_kcontrol_add_path(w->kcontrols[0], path); - } - } else { - list_for_each_entry(path, &w->sinks, list_source) { - if (path->name) - dapm_kcontrol_add_path(w->kcontrols[0], path); - } + snd_soc_dapm_widget_for_each_path(w, dir, path) { + if (path->name) + dapm_kcontrol_add_path(w->kcontrols[0], path); } return 0; @@ -1034,66 +1024,59 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } -/* add widget to list if it's not already in the list */ -static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, - struct snd_soc_dapm_widget *w) +static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, + struct list_head *widgets) { - struct snd_soc_dapm_widget_list *wlist; - int wlistsize, wlistentries, i; - - if (*list == NULL) - return -EINVAL; - - wlist = *list; + struct snd_soc_dapm_widget *w; + struct list_head *it; + unsigned int size = 0; + unsigned int i = 0; - /* is this widget already in the list */ - for (i = 0; i < wlist->num_widgets; i++) { - if (wlist->widgets[i] == w) - return 0; - } + list_for_each(it, widgets) + size++; - /* allocate some new space */ - wlistentries = wlist->num_widgets + 1; - wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - wlistentries * sizeof(struct snd_soc_dapm_widget *); - *list = krealloc(wlist, wlistsize, GFP_KERNEL); - if (*list == NULL) { - dev_err(w->dapm->dev, "ASoC: can't allocate widget list for %s\n", - w->name); + *list = kzalloc(sizeof(**list) + size * sizeof(*w), GFP_KERNEL); + if (*list == NULL) return -ENOMEM; - } - wlist = *list; - /* insert the widget */ - dev_dbg(w->dapm->dev, "ASoC: added %s in widget list pos %d\n", - w->name, wlist->num_widgets); + list_for_each_entry(w, widgets, work_list) + (*list)->widgets[i++] = w; - wlist->widgets[wlist->num_widgets] = w; - wlist->num_widgets++; - return 1; + (*list)->num_widgets = i; + + return 0; } /* - * Recursively check for a completed path to an active or physically connected - * output widget. Returns number of complete paths. + * Common implementation for is_connected_output_ep() and + * is_connected_input_ep(). The function is inlined since the combined size of + * the two specialized functions is only marginally larger then the size of the + * generic function and at the same time the fast path of the specialized + * functions is significantly smaller than the generic function. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct snd_soc_dapm_widget_list **list) +static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, + struct list_head *list, enum snd_soc_dapm_direction dir, + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) { + enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; int con = 0; - if (widget->outputs >= 0) - return widget->outputs; + if (widget->endpoints[dir] >= 0) + return widget->endpoints[dir]; DAPM_UPDATE_STAT(widget, path_checks); - if (widget->is_sink && widget->connected) { - widget->outputs = snd_soc_dapm_suspend_check(widget); - return widget->outputs; + /* do we need to add this widget to the list ? */ + if (list) + list_add_tail(&widget->work_list, list); + + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { + widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); + return widget->endpoints[dir]; } - list_for_each_entry(path, &widget->sinks, list_source) { + snd_soc_dapm_widget_for_each_path(widget, rdir, path) { DAPM_UPDATE_STAT(widget, neighbour_checks); if (path->weak || path->is_supply) @@ -1102,91 +1085,40 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->walking) return 1; - trace_snd_soc_dapm_output_path(widget, path); + trace_snd_soc_dapm_path(widget, dir, path); if (path->connect) { path->walking = 1; - - /* do we need to add this widget to the list ? */ - if (list) { - int err; - err = dapm_list_add_widget(list, path->sink); - if (err < 0) { - dev_err(widget->dapm->dev, - "ASoC: could not add widget %s\n", - widget->name); - path->walking = 0; - return con; - } - } - - con += is_connected_output_ep(path->sink, list); - + con += fn(path->node[dir], list); path->walking = 0; } } - widget->outputs = con; + widget->endpoints[dir] = con; return con; } /* * Recursively check for a completed path to an active or physically connected + * output widget. Returns number of complete paths. + */ +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct list_head *list) +{ + return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, + is_connected_output_ep); +} + +/* + * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct snd_soc_dapm_widget_list **list) + struct list_head *list) { - struct snd_soc_dapm_path *path; - int con = 0; - - if (widget->inputs >= 0) - return widget->inputs; - - DAPM_UPDATE_STAT(widget, path_checks); - - if (widget->is_source && widget->connected) { - widget->inputs = snd_soc_dapm_suspend_check(widget); - return widget->inputs; - } - - list_for_each_entry(path, &widget->sources, list_sink) { - DAPM_UPDATE_STAT(widget, neighbour_checks); - - if (path->weak || path->is_supply) - continue; - - if (path->walking) - return 1; - - trace_snd_soc_dapm_input_path(widget, path); - - if (path->connect) { - path->walking = 1; - - /* do we need to add this widget to the list ? */ - if (list) { - int err; - err = dapm_list_add_widget(list, path->source); - if (err < 0) { - dev_err(widget->dapm->dev, - "ASoC: could not add widget %s\n", - widget->name); - path->walking = 0; - return con; - } - } - - con += is_connected_input_ep(path->source, list); - - path->walking = 0; - } - } - - widget->inputs = con; - - return con; + return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, + is_connected_input_ep); } /** @@ -1206,7 +1138,9 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; + LIST_HEAD(widgets); int paths; + int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); @@ -1215,14 +1149,21 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, * to reset the cached number of inputs and outputs. */ list_for_each_entry(w, &card->widgets, list) { - w->inputs = -1; - w->outputs = -1; + w->endpoints[SND_SOC_DAPM_DIR_IN] = -1; + w->endpoints[SND_SOC_DAPM_DIR_OUT] = -1; } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, list); + paths = is_connected_output_ep(dai->playback_widget, &widgets); else - paths = is_connected_input_ep(dai->capture_widget, list); + paths = is_connected_input_ep(dai->capture_widget, &widgets); + + /* Drop starting point */ + list_del(widgets.next); + + ret = dapm_widget_list_create(list, &widgets); + if (ret) + paths = ret; trace_snd_soc_dapm_connected(paths, stream); mutex_unlock(&card->dapm_mutex); @@ -1323,7 +1264,7 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); /* Check if one of our outputs is connected */ - list_for_each_entry(path, &w->sinks, list_source) { + snd_soc_dapm_widget_for_each_sink_path(w, path) { DAPM_UPDATE_STAT(w, neighbour_checks); if (path->weak) @@ -1747,12 +1688,12 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, /* If we changed our power state perhaps our neigbours changed * also. */ - list_for_each_entry(path, &w->sources, list_sink) + snd_soc_dapm_widget_for_each_source_path(w, path) dapm_widget_set_peer_power(path->source, power, path->connect); /* Supplies can't affect their outputs, only their inputs */ if (!w->is_supply) { - list_for_each_entry(path, &w->sinks, list_source) + snd_soc_dapm_widget_for_each_sink_path(w, path) dapm_widget_set_peer_power(path->sink, power, path->connect); } @@ -1952,6 +1893,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, size_t count, loff_t *ppos) { struct snd_soc_dapm_widget *w = file->private_data; + struct snd_soc_card *card = w->dapm->card; + enum snd_soc_dapm_direction dir, rdir; char *buf; int in, out; ssize_t ret; @@ -1961,6 +1904,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; + mutex_lock(&card->dapm_mutex); + /* Supply widgets are not handled by is_connected_{input,output}_ep() */ if (w->is_supply) { in = 0; @@ -1986,27 +1931,25 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->sname, w->active ? "active" : "inactive"); - list_for_each_entry(p, &w->sources, list_sink) { - if (p->connected && !p->connected(w, p->source)) - continue; + snd_soc_dapm_for_each_direction(dir) { + rdir = SND_SOC_DAPM_DIR_REVERSE(dir); + snd_soc_dapm_widget_for_each_path(w, dir, p) { + if (p->connected && !p->connected(w, p->node[rdir])) + continue; - if (p->connect) - ret += snprintf(buf + ret, PAGE_SIZE - ret, - " in \"%s\" \"%s\"\n", - p->name ? p->name : "static", - p->source->name); - } - list_for_each_entry(p, &w->sinks, list_source) { - if (p->connected && !p->connected(w, p->sink)) - continue; + if (!p->connect) + continue; - if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, - " out \"%s\" \"%s\"\n", + " %s \"%s\" \"%s\"\n", + (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", - p->sink->name); + p->node[rdir]->name); + } } + mutex_unlock(&card->dapm_mutex); + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -2220,14 +2163,16 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); -static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf) +static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, + char *buf) { + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt); struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; - list_for_each_entry(w, &codec->component.card->widgets, list) { - if (w->dapm != &codec->dapm) + list_for_each_entry(w, &cmpnt->card->widgets, list) { + if (w->dapm != dapm) continue; /* only display widgets that burnm power */ @@ -2255,7 +2200,7 @@ static ssize_t dapm_widget_show_codec(struct snd_soc_codec *codec, char *buf) } } - switch (codec->dapm.bias_level) { + switch (snd_soc_dapm_get_bias_level(dapm)) { case SND_SOC_BIAS_ON: state = "On"; break; @@ -2281,11 +2226,16 @@ static ssize_t dapm_widget_show(struct device *dev, struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int i, count = 0; + mutex_lock(&rtd->card->dapm_mutex); + for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_codec *codec = rtd->codec_dais[i]->codec; - count += dapm_widget_show_codec(codec, buf + count); + struct snd_soc_component *cmpnt = rtd->codec_dais[i]->component; + + count += dapm_widget_show_component(cmpnt, buf + count); } + mutex_unlock(&rtd->card->dapm_mutex); + return count; } @@ -2298,37 +2248,43 @@ struct attribute *soc_dapm_dev_attrs[] = { static void dapm_free_path(struct snd_soc_dapm_path *path) { - list_del(&path->list_sink); - list_del(&path->list_source); + list_del(&path->list_node[SND_SOC_DAPM_DIR_IN]); + list_del(&path->list_node[SND_SOC_DAPM_DIR_OUT]); list_del(&path->list_kcontrol); list_del(&path->list); kfree(path); } +void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p, *next_p; + enum snd_soc_dapm_direction dir; + + list_del(&w->list); + /* + * remove source and sink paths associated to this widget. + * While removing the path, remove reference to it from both + * source and sink widgets so that path is removed only once. + */ + snd_soc_dapm_for_each_direction(dir) { + snd_soc_dapm_widget_for_each_path_safe(w, dir, p, next_p) + dapm_free_path(p); + } + + kfree(w->kcontrols); + kfree_const(w->name); + kfree(w); +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w, *next_w; - struct snd_soc_dapm_path *p, *next_p; list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { if (w->dapm != dapm) continue; - list_del(&w->list); - /* - * remove source and sink paths associated to this widget. - * While removing the path, remove reference to it from both - * source and sink widgets so that path is removed only once. - */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) - dapm_free_path(p); - - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) - dapm_free_path(p); - - kfree(w->kcontrols); - kfree(w->name); - kfree(w); + snd_soc_dapm_free_widget(w); } } @@ -2434,20 +2390,22 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); */ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w) { + enum snd_soc_dapm_direction dir; struct snd_soc_dapm_path *p; + unsigned int ep; switch (w->id) { case snd_soc_dapm_input: /* On a fully routed card a input is never a source */ if (w->dapm->card->fully_routed) - break; - w->is_source = 1; - list_for_each_entry(p, &w->sources, list_sink) { + return; + ep = SND_SOC_DAPM_EP_SOURCE; + snd_soc_dapm_widget_for_each_source_path(w, p) { if (p->source->id == snd_soc_dapm_micbias || p->source->id == snd_soc_dapm_mic || p->source->id == snd_soc_dapm_line || p->source->id == snd_soc_dapm_output) { - w->is_source = 0; + ep = 0; break; } } @@ -2455,25 +2413,30 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w) case snd_soc_dapm_output: /* On a fully routed card a output is never a sink */ if (w->dapm->card->fully_routed) - break; - w->is_sink = 1; - list_for_each_entry(p, &w->sinks, list_source) { + return; + ep = SND_SOC_DAPM_EP_SINK; + snd_soc_dapm_widget_for_each_sink_path(w, p) { if (p->sink->id == snd_soc_dapm_spk || p->sink->id == snd_soc_dapm_hp || p->sink->id == snd_soc_dapm_line || p->sink->id == snd_soc_dapm_input) { - w->is_sink = 0; + ep = 0; break; } } break; case snd_soc_dapm_line: - w->is_sink = !list_empty(&w->sources); - w->is_source = !list_empty(&w->sinks); + ep = 0; + snd_soc_dapm_for_each_direction(dir) { + if (!list_empty(&w->edges[dir])) + ep |= SND_SOC_DAPM_DIR_TO_EP(dir); + } break; default: - break; + return; } + + w->is_ep = ep; } static int snd_soc_dapm_check_dynamic_path(struct snd_soc_dapm_context *dapm, @@ -2526,6 +2489,8 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink)) { + struct snd_soc_dapm_widget *widgets[2]; + enum snd_soc_dapm_direction dir; struct snd_soc_dapm_path *path; int ret; @@ -2558,13 +2523,14 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, if (!path) return -ENOMEM; - path->source = wsource; - path->sink = wsink; + path->node[SND_SOC_DAPM_DIR_IN] = wsource; + path->node[SND_SOC_DAPM_DIR_OUT] = wsink; + widgets[SND_SOC_DAPM_DIR_IN] = wsource; + widgets[SND_SOC_DAPM_DIR_OUT] = wsink; + path->connected = connected; INIT_LIST_HEAD(&path->list); INIT_LIST_HEAD(&path->list_kcontrol); - INIT_LIST_HEAD(&path->list_source); - INIT_LIST_HEAD(&path->list_sink); if (wsource->is_supply || wsink->is_supply) path->is_supply = 1; @@ -2602,14 +2568,13 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, } list_add(&path->list, &dapm->card->paths); - list_add(&path->list_sink, &wsink->sources); - list_add(&path->list_source, &wsource->sinks); - - dapm_update_widget_flags(wsource); - dapm_update_widget_flags(wsink); + snd_soc_dapm_for_each_direction(dir) + list_add(&path->list_node[dir], &widgets[dir]->edges[dir]); - dapm_mark_dirty(wsource, "Route added"); - dapm_mark_dirty(wsink, "Route added"); + snd_soc_dapm_for_each_direction(dir) { + dapm_update_widget_flags(widgets[dir]); + dapm_mark_dirty(widgets[dir], "Route added"); + } if (dapm->card->instantiated && path->connect) dapm_path_invalidate(path); @@ -2857,7 +2822,7 @@ static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm, dev_warn(dapm->dev, "ASoC: Ignoring control for weak route %s->%s\n", route->source, route->sink); - list_for_each_entry(path, &source->sinks, list_source) { + snd_soc_dapm_widget_for_each_sink_path(source, path) { if (path->sink == sink) { path->weak = 1; count++; @@ -2911,7 +2876,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card to be checked for new dapm widgets * * Checks the codec for any new dapm widgets and creates them if found. * @@ -3291,6 +3256,7 @@ struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { + enum snd_soc_dapm_direction dir; struct snd_soc_dapm_widget *w; const char *prefix; int ret; @@ -3334,16 +3300,10 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } prefix = soc_dapm_prefix(dapm); - if (prefix) { + if (prefix) w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix, - widget->sname); - } else { - w->name = kasprintf(GFP_KERNEL, "%s", widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname); - } + else + w->name = kstrdup_const(widget->name, GFP_KERNEL); if (w->name == NULL) { kfree(w); return NULL; @@ -3351,27 +3311,27 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_mic: - w->is_source = 1; + w->is_ep = SND_SOC_DAPM_EP_SOURCE; w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_input: if (!dapm->card->fully_routed) - w->is_source = 1; + w->is_ep = SND_SOC_DAPM_EP_SOURCE; w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_spk: case snd_soc_dapm_hp: - w->is_sink = 1; + w->is_ep = SND_SOC_DAPM_EP_SINK; w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_output: if (!dapm->card->fully_routed) - w->is_sink = 1; + w->is_ep = SND_SOC_DAPM_EP_SINK; w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_vmid: case snd_soc_dapm_siggen: - w->is_source = 1; + w->is_ep = SND_SOC_DAPM_EP_SOURCE; w->power_check = dapm_always_on_check_power; break; case snd_soc_dapm_mux: @@ -3405,14 +3365,14 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - INIT_LIST_HEAD(&w->sources); - INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); INIT_LIST_HEAD(&w->dirty); list_add_tail(&w->list, &dapm->card->widgets); - w->inputs = -1; - w->outputs = -1; + snd_soc_dapm_for_each_direction(dir) { + INIT_LIST_HEAD(&w->edges[dir]); + w->endpoints[dir] = -1; + } /* machine layer set ups unconnected pins and insertions */ w->connected = 1; @@ -3466,19 +3426,17 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, int ret; if (WARN_ON(!config) || - WARN_ON(list_empty(&w->sources) || list_empty(&w->sinks))) + WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) || + list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) return -EINVAL; /* We only support a single source and sink, pick the first */ - source_p = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, - list_source); - - if (WARN_ON(!source_p || !sink_p) || - WARN_ON(!sink_p->source || !source_p->sink) || - WARN_ON(!source_p->source || !sink_p->sink)) - return -EINVAL; + source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT], + struct snd_soc_dapm_path, + list_node[SND_SOC_DAPM_DIR_OUT]); + sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN], + struct snd_soc_dapm_path, + list_node[SND_SOC_DAPM_DIR_IN]); source = source_p->source->priv; sink = sink_p->sink->priv; @@ -3792,7 +3750,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || !strstr(w->sname, dai_w->sname)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { @@ -3820,11 +3778,6 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - /* there is no point in connecting BE DAI links with dummies */ - if (snd_soc_dai_is_dummy(codec_dai) || - snd_soc_dai_is_dummy(cpu_dai)) - continue; - /* connect BE DAI playback if widgets are valid */ if (codec_dai->playback_widget && cpu_dai->playback_widget) { source = cpu_dai->playback_widget; @@ -3855,6 +3808,7 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, int event) { struct snd_soc_dapm_widget *w; + unsigned int ep; if (stream == SNDRV_PCM_STREAM_PLAYBACK) w = dai->playback_widget; @@ -3864,12 +3818,22 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, if (w) { dapm_mark_dirty(w, "stream event"); + if (w->id == snd_soc_dapm_dai_in) { + ep = SND_SOC_DAPM_EP_SOURCE; + dapm_widget_invalidate_input_paths(w); + } else { + ep = SND_SOC_DAPM_EP_SINK; + dapm_widget_invalidate_output_paths(w); + } + switch (event) { case SND_SOC_DAPM_STREAM_START: w->active = 1; + w->is_ep = ep; break; case SND_SOC_DAPM_STREAM_STOP: w->active = 0; + w->is_ep = 0; break; case SND_SOC_DAPM_STREAM_SUSPEND: case SND_SOC_DAPM_STREAM_RESUME: @@ -3877,14 +3841,6 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: break; } - - if (w->id == snd_soc_dapm_dai_in) { - w->is_source = w->active; - dapm_widget_invalidate_input_paths(w); - } else { - w->is_sink = w->active; - dapm_widget_invalidate_output_paths(w); - } } } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 256b9c9..70e4b9d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1231,24 +1231,17 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, } int dpcm_path_get(struct snd_soc_pcm_runtime *fe, - int stream, struct snd_soc_dapm_widget_list **list_) + int stream, struct snd_soc_dapm_widget_list **list) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; - struct snd_soc_dapm_widget_list *list; int paths; - list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) + - sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL); - if (list == NULL) - return -ENOMEM; - /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); - *list_ = list; return paths; } @@ -1306,7 +1299,12 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, switch (list->widgets[i]->id) { case snd_soc_dapm_dai_in: + if (stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; + break; case snd_soc_dapm_dai_out: + if (stream != SNDRV_PCM_STREAM_CAPTURE) + continue; break; default: continue; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index d096068..f4e92d3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -33,6 +33,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/soc-topology.h> +#include <sound/tlv.h> /* * We make several passes over the data (since it wont necessarily be ordered) @@ -144,7 +145,7 @@ static const struct snd_soc_tplg_kcontrol_ops io_ops[] = { {SND_SOC_TPLG_CTL_STROBE, snd_soc_get_strobe, snd_soc_put_strobe, NULL}, {SND_SOC_TPLG_DAPM_CTL_VOLSW, snd_soc_dapm_get_volsw, - snd_soc_dapm_put_volsw, NULL}, + snd_soc_dapm_put_volsw, snd_soc_info_volsw}, {SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE, snd_soc_dapm_get_enum_double, snd_soc_dapm_put_enum_double, snd_soc_info_enum_double}, {SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT, snd_soc_dapm_get_enum_double, @@ -534,7 +535,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, k->put = bops[i].put; if (k->get == NULL && bops[i].id == hdr->ops.get) k->get = bops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && bops[i].id == hdr->ops.info) k->info = bops[i].info; } @@ -579,29 +580,51 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, return 0; } + +static int soc_tplg_create_tlv_db_scale(struct soc_tplg *tplg, + struct snd_kcontrol_new *kc, struct snd_soc_tplg_tlv_dbscale *scale) +{ + unsigned int item_len = 2 * sizeof(unsigned int); + unsigned int *p; + + p = kzalloc(item_len + 2 * sizeof(unsigned int), GFP_KERNEL); + if (!p) + return -ENOMEM; + + p[0] = SNDRV_CTL_TLVT_DB_SCALE; + p[1] = item_len; + p[2] = scale->min; + p[3] = (scale->step & TLV_DB_SCALE_MASK) + | (scale->mute ? TLV_DB_SCALE_MUTE : 0); + + kc->tlv.p = (void *)p; + return 0; +} + static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, u32 tlv_size) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_hdr *tc) { struct snd_soc_tplg_ctl_tlv *tplg_tlv; - struct snd_ctl_tlv *tlv; - if (tlv_size == 0) + if (!(tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)) return 0; - tplg_tlv = (struct snd_soc_tplg_ctl_tlv *) tplg->pos; - tplg->pos += tlv_size; - - tlv = kzalloc(sizeof(*tlv) + tlv_size, GFP_KERNEL); - if (tlv == NULL) - return -ENOMEM; - - dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, tplg_tlv->size); + if (tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + kc->tlv.c = snd_soc_bytes_tlv_callback; + } else { + tplg_tlv = &tc->tlv; + switch (tplg_tlv->type) { + case SNDRV_CTL_TLVT_DB_SCALE: + return soc_tplg_create_tlv_db_scale(tplg, kc, + &tplg_tlv->scale); - tlv->numid = tplg_tlv->numid; - tlv->length = tplg_tlv->size; - memcpy(tlv->tlv, tplg_tlv + 1, tplg_tlv->size); - kc->tlv.p = (void *)tlv; + /* TODO: add support for other TLV types */ + default: + dev_dbg(tplg->dev, "Unsupported TLV type %d\n", + tplg_tlv->type); + return -EINVAL; + } + } return 0; } @@ -773,7 +796,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, mc->hdr.tlv_size); + soc_tplg_create_tlv(tplg, &kc, &mc->hdr); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, @@ -1351,6 +1374,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, template.reg = w->reg; template.shift = w->shift; template.mask = w->mask; + template.subseq = w->subseq; template.on_val = w->invert ? 0 : 1; template.off_val = w->invert ? 1 : 0; template.ignore_suspend = w->ignore_suspend; @@ -1734,7 +1758,6 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, u32 index) { struct snd_soc_dapm_widget *w, *next_w; - struct snd_soc_dapm_path *p, *next_p; list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { @@ -1746,31 +1769,9 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, if (w->dobj.index != index && w->dobj.index != SND_SOC_TPLG_INDEX_ALL) continue; - - list_del(&w->list); - - /* - * remove source and sink paths associated to this widget. - * While removing the path, remove reference to it from both - * source and sink widgets so that path is removed only once. - */ - list_for_each_entry_safe(p, next_p, &w->sources, list_sink) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p); - } - list_for_each_entry_safe(p, next_p, &w->sinks, list_source) { - list_del(&p->list_sink); - list_del(&p->list_source); - list_del(&p->list); - kfree(p); - } /* check and free and dynamic widget kcontrols */ snd_soc_tplg_widget_remove(w); - kfree(w->kcontrols); - kfree(w->name); - kfree(w); + snd_soc_dapm_free_widget(w); } } EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c index 98d96e1..1930c42 100644 --- a/sound/soc/zte/zx296702-i2s.c +++ b/sound/soc/zte/zx296702-i2s.c @@ -393,9 +393,9 @@ static int zx_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); zx_i2s->mapbase = res->start; zx_i2s->reg_base = devm_ioremap_resource(&pdev->dev, res); - if (!zx_i2s->reg_base) { + if (IS_ERR(zx_i2s->reg_base)) { dev_err(&pdev->dev, "ioremap failed!\n"); - return -EIO; + return PTR_ERR(zx_i2s->reg_base); } writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx296702-spdif.c index 11a0e46..26265ce 100644 --- a/sound/soc/zte/zx296702-spdif.c +++ b/sound/soc/zte/zx296702-spdif.c @@ -322,9 +322,9 @@ static int zx_spdif_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); zx_spdif->mapbase = res->start; zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res); - if (!zx_spdif->reg_base) { + if (IS_ERR(zx_spdif->reg_base)) { dev_err(&pdev->dev, "ioremap failed!\n"); - return -EIO; + return PTR_ERR(zx_spdif->reg_base); } zx_spdif_dev_init(zx_spdif->reg_base); |
