diff options
Diffstat (limited to 'sound/soc')
| -rw-r--r-- | sound/soc/au1x/Kconfig | 10 | ||||
| -rw-r--r-- | sound/soc/au1x/Makefile | 4 | ||||
| -rw-r--r-- | sound/soc/au1x/db1200.c | 141 | ||||
| -rw-r--r-- | sound/soc/au1x/dbdma2.c | 14 | ||||
| -rw-r--r-- | sound/soc/au1x/sample-ac97.c | 144 | ||||
| -rw-r--r-- | sound/soc/codecs/uda1380.c | 2 | ||||
| -rw-r--r-- | sound/soc/codecs/wm8350.c | 8 | ||||
| -rw-r--r-- | sound/soc/fsl/efika-audio-fabric.c | 2 | ||||
| -rw-r--r-- | sound/soc/fsl/pcm030-audio-fabric.c | 2 | ||||
| -rw-r--r-- | sound/soc/omap/omap-mcbsp.c | 144 | ||||
| -rw-r--r-- | sound/soc/omap/omap-mcbsp.h | 4 | ||||
| -rw-r--r-- | sound/soc/sh/siu.h | 2 | ||||
| -rw-r--r-- | sound/soc/sh/siu_pcm.c | 2 |
13 files changed, 308 insertions, 171 deletions
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 410a893aa66..4b67140fdec 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97 ## ## Boards ## -config SND_SOC_SAMPLE_PSC_AC97 - tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" +config SND_SOC_DB1200 + tristate "DB1200 AC97+I2S audio support" depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC + select SND_SOC_AU1XPSC_I2S + select SND_SOC_WM8731 help - This is a sample AC97 sound machine for use in Au12x0/Au1550 - based systems which have audio on PSC1 (e.g. Db1200 demoboard). + Select this option to enable audio (AC97 or I2S) on the + Alchemy/AMD/RMI DB1200 demoboard. diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 6c6950b8003..16873076e8c 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o # Boards -snd-soc-sample-ac97-objs := sample-ac97.o +snd-soc-db1200-objs := db1200.o -obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o +obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c new file mode 100644 index 00000000000..cdf7be1b9b9 --- /dev/null +++ b/sound/soc/au1x/db1200.c @@ -0,0 +1,141 @@ +/* + * DB1200 ASoC audio fabric support code. + * + * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_psc.h> +#include <asm/mach-au1x00/au1xxx_dbdma.h> +#include <asm/mach-db1x00/bcsr.h> + +#include "../codecs/ac97.h" +#include "../codecs/wm8731.h" +#include "psc.h" + +/*------------------------- AC97 PART ---------------------------*/ + +static struct snd_soc_dai_link db1200_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card db1200_ac97_machine = { + .name = "DB1200_AC97", + .dai_link = &db1200_ac97_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_ac97_devdata = { + .card = &db1200_ac97_machine, + .codec_dev = &soc_codec_dev_ac97, +}; + +/*------------------------- I2S PART ---------------------------*/ + +static int db1200_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* WM8731 has its own 12MHz crystal */ + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + + /* codec is bitclock and lrclk master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = 0; +out: + return ret; +} + +static struct snd_soc_ops db1200_i2s_wm8731_ops = { + .startup = db1200_i2s_startup, +}; + +static struct snd_soc_dai_link db1200_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &au1xpsc_i2s_dai, + .codec_dai = &wm8731_dai, + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1200_i2s_machine = { + .name = "DB1200_I2S", + .dai_link = &db1200_i2s_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_i2s_devdata = { + .card = &db1200_i2s_machine, + .codec_dev = &soc_codec_dev_wm8731, +}; + +/*------------------------- COMMON PART ---------------------------*/ + +static struct platform_device *db1200_asoc_dev; + +static int __init db1200_audio_load(void) +{ + int ret; + + ret = -ENOMEM; + db1200_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!db1200_asoc_dev) + goto out; + + /* DB1200 board setup set PSC1MUX to preferred audio device */ + if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX) + platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata); + else + platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata); + + db1200_ac97_devdata.dev = &db1200_asoc_dev->dev; + db1200_i2s_devdata.dev = &db1200_asoc_dev->dev; + ret = platform_device_add(db1200_asoc_dev); + + if (ret) { + platform_device_put(db1200_asoc_dev); + db1200_asoc_dev = NULL; + } +out: + return ret; +} + +static void __exit db1200_audio_unload(void) +{ + platform_device_unregister(db1200_asoc_dev); +} + +module_init(db1200_audio_load); +module_exit(db1200_audio_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1200 ASoC audio support"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 19e4d37eba1..6d9f4c62494 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -51,8 +51,8 @@ struct au1xpsc_audio_dmadata { struct snd_pcm_substream *substream; unsigned long curr_period; /* current segment DDMA is working on */ unsigned long q_period; /* queue period(s) */ - unsigned long dma_area; /* address of queued DMA area */ - unsigned long dma_area_s; /* start address of DMA area */ + dma_addr_t dma_area; /* address of queued DMA area */ + dma_addr_t dma_area_s; /* start address of DMA area */ unsigned long pos; /* current byte position being played */ unsigned long periods; /* number of SG segments in total */ unsigned long period_bytes; /* size in bytes of one SG segment */ @@ -94,8 +94,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_source_flags(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), + au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area, cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ @@ -109,9 +108,8 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_dest_flags(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), - cd->period_bytes, DDMA_FLAGS_IE); + au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area, + cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ ++cd->q_period; @@ -233,7 +231,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, pcd->substream = substream; pcd->period_bytes = params_period_bytes(params); pcd->periods = params_periods(params); - pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr; + pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; pcd->q_period = 0; pcd->curr_period = 0; pcd->pos = 0; diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c deleted file mode 100644 index 27683eb7905..00000000000 --- a/sound/soc/au1x/sample-ac97.c +++ /dev/null @@ -1,144 +0,0 @@ -/* - * Sample Au12x0/Au1550 PSC AC97 sound machine. - * - * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms outlined in the file COPYING at the root of this - * source archive. - * - * This is a very generic AC97 sound machine driver for boards which - * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1xxx_psc.h> -#include <asm/mach-au1x00/au1xxx_dbdma.h> - -#include "../codecs/ac97.h" -#include "psc.h" - -static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_sync(codec); - return 0; -} - -static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ - .codec_dai = &ac97_dai, /* see codecs/ac97.c */ - .init = au1xpsc_sample_ac97_init, - .ops = NULL, -}; - -static struct snd_soc_card au1xpsc_sample_ac97_machine = { - .name = "Au1xxx PSC AC97 Audio", - .dai_link = &au1xpsc_sample_ac97_dai, - .num_links = 1, -}; - -static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct resource au1xpsc_psc1_res[] = { - [0] = { - .start = CPHYSADDR(PSC1_BASE_ADDR), - .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, - .flags = IORESOURCE_MEM, - }, - [1] = { -#ifdef CONFIG_SOC_AU1200 - .start = AU1200_PSC1_INT, - .end = AU1200_PSC1_INT, -#elif defined(CONFIG_SOC_AU1550) - .start = AU1550_PSC1_INT, - .end = AU1550_PSC1_INT, -#endif - .flags = IORESOURCE_IRQ, - }, - [2] = { - .start = DSCR_CMD0_PSC1_TX, - .end = DSCR_CMD0_PSC1_TX, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = DSCR_CMD0_PSC1_RX, - .end = DSCR_CMD0_PSC1_RX, - .flags = IORESOURCE_DMA, - }, -}; - -static struct platform_device *au1xpsc_sample_ac97_dev; - -static int __init au1xpsc_sample_ac97_load(void) -{ - int ret; - -#ifdef CONFIG_SOC_AU1200 - unsigned long io; - - /* modify sys_pinfunc for AC97 on PSC1 */ - io = au_readl(SYS_PINFUNC); - io |= SYS_PINFUNC_P1C; - io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); - au_writel(io, SYS_PINFUNC); - au_sync(); -#endif - - ret = -ENOMEM; - - /* setup PSC clock source for AC97 part: external clock provided - * by codec. The psc-ac97.c driver depends on this setting! - */ - au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); - au_sync(); - - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; - - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; - - platform_set_drvdata(au1xpsc_sample_ac97_dev, - &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); - - if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; - } - -out: - return ret; -} - -static void __exit au1xpsc_sample_ac97_exit(void) -{ - platform_device_unregister(au1xpsc_sample_ac97_dev); -} - -module_init(au1xpsc_sample_ac97_load); -module_exit(au1xpsc_sample_ac97_exit); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a2763c2e734..9cd0a66b766 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -137,7 +137,7 @@ static void uda1380_flush_work(struct work_struct *work) { int bit, reg; - for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) { + for_each_set_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) { reg = 0x10 + bit; pr_debug("uda1380: flush reg %x val %x:\n", reg, uda1380_read_reg_cache(uda1380_codec, reg)); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 718ef912e75..df2c6d9617f 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1349,7 +1349,7 @@ static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) int mask; struct wm8350_jack_data *jack = NULL; - switch (irq) { + switch (irq - wm8350->irq_base) { case WM8350_IRQ_CODEC_JCK_DET_L: jack = &priv->hpl; mask = WM8350_JACK_L_LVL; @@ -1424,7 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); /* Sync status */ - wm8350_hp_jack_handler(irq, priv); + wm8350_hp_jack_handler(irq + wm8350->irq_base, priv); return 0; } @@ -1521,8 +1521,8 @@ static int wm8350_remove(struct platform_device *pdev) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); - wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); - wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); priv->hpl.jack = NULL; priv->hpr.jack = NULL; diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 3326e2a1e86..1a5b8e0d6a3 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -55,7 +55,7 @@ static __init int efika_fabric_init(void) struct platform_device *pdev; int rc; - if (!machine_is_compatible("bplan,efika")) + if (!of_machine_is_compatible("bplan,efika")) return -ENODEV; card.platform = &mpc5200_audio_dma_platform; diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index b928ef7d28e..6644cba7cbf 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -55,7 +55,7 @@ static __init int pcm030_fabric_init(void) struct platform_device *pdev; int rc; - if (!machine_is_compatible("phytec,pcm030")) + if (!of_machine_is_compatible("phytec,pcm030")) return -ENODEV; card.platform = &mpc5200_audio_dma_platform; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d2972566418..e814a9591f7 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,6 +39,14 @@ #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = omap_mcbsp_st_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long) &(struct soc_mixer_control) \ + {.min = xmin, .max = xmax} } + struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; @@ -82,11 +90,11 @@ static const int omap1_dma_reqs[][2] = {}; static const unsigned long omap1_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3) static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, @@ -124,7 +132,7 @@ static const unsigned long omap2430_mcbsp_port[][2] = { static const unsigned long omap2430_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP3) static const unsigned long omap34xx_mcbsp_port[][2] = { { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, @@ -639,6 +647,136 @@ struct snd_soc_dai omap_mcbsp_dai[] = { EXPORT_SYMBOL_GPL(omap_mcbsp_dai); +int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int max = mc->max; + int min = mc->min; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = min; + uinfo->value.integer.max = max; + return 0; +} + +#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(id, channel) \ +static int \ +omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + struct soc_mixer_control *mc = \ + (struct soc_mixer_control *)kc->private_value; \ + int max = mc->max; \ + int min = mc->min; \ + int val = uc->value.integer.value[0]; \ + \ + if (val < min || val > max) \ + return -EINVAL; \ + \ + /* OMAP McBSP implementation uses index values 0..4 */ \ + return omap_st_set_chgain((id)-1, channel, val); \ +} + +#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(id, channel) \ +static int \ +omap_mcbsp##id##_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ + struct snd_ctl_elem_value *uc) \ +{ \ + s16 chgain; \ + \ + if (omap_st_get_chgain((id)-1, channel, &chgain)) \ + return -EAGAIN; \ + \ + uc->value.integer.value[0] = chgain; \ + return 0; \ +} + +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 0) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 1) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 0) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 1) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 0) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 1) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 0) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 1) + +static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u8 value = ucontrol->value.integer.value[0]; + + if (value == omap_st_is_enabled(mc->reg)) + return 0; + + if (value) + omap_st_enable(mc->reg); + else + omap_st_disable(mc->reg); + + return 1; +} + +static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + ucontrol->value.integer.value[0] = omap_st_is_enabled(mc->reg); + return 0; +} + +static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = { + SOC_SINGLE_EXT("McBSP2 Sidetone Switch", 1, 0, 1, 0, + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume", + -32768, 32767, + omap_mcbsp2_get_st_ch0_volume, + omap_mcbsp2_set_st_ch0_volume), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume", + -32768, 32767, + omap_mcbsp2_get_st_ch1_volume, + omap_mcbsp2_set_st_ch1_volume), +}; + +static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { + SOC_SINGLE_EXT("McBSP3 Sidetone Switch", 2, 0, 1, 0, + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume", + -32768, 32767, + omap_mcbsp3_get_st_ch0_volume, + omap_mcbsp3_set_st_ch0_volume), + OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume", + -32768, 32767, + omap_mcbsp3_get_st_ch1_volume, + omap_mcbsp3_set_st_ch1_volume), +}; + +int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id) +{ + if (!cpu_is_omap34xx()) + return -ENODEV; + + switch (mcbsp_id) { + case 1: /* McBSP 2 */ + return snd_soc_add_controls(codec, omap_mcbsp2_st_controls, + ARRAY_SIZE(omap_mcbsp2_st_controls)); + case 2: /* McBSP 3 */ + return snd_soc_add_controls(codec, omap_mcbsp3_st_controls, + ARRAY_SIZE(omap_mcbsp3_st_controls)); + default: + break; + } + + return -EINVAL; +} +EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); + static int __init snd_omap_mcbsp_init(void) { return snd_soc_register_dais(omap_mcbsp_dai, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 647d2f981ab..6c363e5f438 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -50,11 +50,13 @@ enum omap_mcbsp_div { #undef NUM_LINKS #define NUM_LINKS 3 #endif -#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) #undef NUM_LINKS #define NUM_LINKS 5 #endif extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; +int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id); + #endif diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h index 9cc04ab2bce..c0bfab8fed3 100644 --- a/sound/soc/sh/siu.h +++ b/sound/soc/sh/siu.h @@ -72,7 +72,7 @@ struct siu_firmware { #include <linux/interrupt.h> #include <linux/io.h> -#include <asm/dma-sh.h> +#include <asm/dmaengine.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index c5efc30f013..ba7f8d05d97 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -32,7 +32,7 @@ #include <sound/pcm_params.h> #include <sound/soc-dai.h> -#include <asm/dma-sh.h> +#include <asm/dmaengine.h> #include <asm/siu.h> #include "siu.h" |