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

[alsa-devel] [PATCH] ASoC don't clobber platform DMA driver ops
by Alan Tull 10 Nov '11

10 Nov '11

Previously, only one static struct for ops existed for all platform DMA drivers to share. Half of the ops are shared functions which don't have stubs in the ALSA core. The other half are initialized to point directly to ops in the platform driver. This creates problems where each time soc_new_pcm is called, the new platform driver's ops would overwrite a subset of the ops. This patch creates and populates a ops struct for each call to soc_pcm_new. Also creates a common soc_pcm_ioctl function that calls all the driver components for ioctl calls. Signed-off-by: Alan Tull <alan.tull(a)freescale.com> --- include/sound/soc-dai.h | 2 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 80 +++++++++++++++++++++++++++++++++++----------- 4 files changed, 65 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5ad5f3a..b3f880e 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -172,6 +172,8 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); + int (*ioctl)(struct snd_pcm_substream *, unsigned int, void *, + struct snd_soc_dai *); /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..1574325 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -517,6 +517,7 @@ struct snd_soc_ops { int (*hw_free)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); int (*trigger)(struct snd_pcm_substream *, int); + int (*ioctl)(struct snd_pcm_substream *, unsigned int, void *); }; /* SoC cache ops */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b085d8e..1b48fc6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1547,6 +1547,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_free(&card->dapm); + kfree(card->rtd->ops); kfree(card->rtd); snd_card_free(card->snd_card); return 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2879c88..2bd7504 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -567,16 +567,39 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; +static int soc_pcm_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (rtd->dai_link->ops && rtd->dai_link->ops->ioctl) { + ret = rtd->dai_link->ops->ioctl(substream, cmd, arg); + if (ret < 0) + return ret; + } + + if (codec_dai->driver->ops->ioctl) { + ret = codec_dai->driver->ops->ioctl(substream, cmd, arg, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->ops->ioctl) { + ret = platform->driver->ops->ioctl(substream, cmd, arg); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->ioctl) { + ret = cpu_dai->driver->ops->ioctl(substream, cmd, arg, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) @@ -586,9 +609,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; + struct snd_pcm_ops *soc_pcm_ops; char new_name[64]; int ret = 0, playback = 0, capture = 0; + /* ASoC PCM operations */ + soc_pcm_ops = kzalloc(sizeof(struct snd_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + printk(KERN_ERR "asoc: can't create pcm ops for codec %s\n", codec->name); + return -ENOMEM; + } + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -603,34 +634,45 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) num, playback, capture, &pcm); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + kfree(soc_pcm_ops); return ret; } /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_codec_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->pointer = soc_pcm_pointer; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->ioctl = soc_pcm_ioctl; + rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } + rtd->ops = soc_pcm_ops; + if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { pr_err("asoc: platform pcm constructor failed\n"); + kfree(rtd->ops); return ret; } } -- 1.6.0.4

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[alsa-devel] Playback and capture synchronization
by anish kumar 10 Nov '11

10 Nov '11

> > Patrick Lai wrote: > >> I have a use case requiring playback and capture starting at the same > >> time. Latency between starting of playback and capture streams need to > >> be less than few PCM samples. Base on my understanding on ALSA which > >> does not have the concept of grouping multiple streams, Is this use case for loopback?Right now i am doing loopback with two terminals with one terminal running aplay and other arecord and before running aplay and arecord we use couple of mixer controls to set the paths but we want to do all this from user space i.e. ALSA UCM(use case manager) & Android Audio Flinger.We are not using snd-aloop for loopback. After reading this thread looks like I need to implement support in driver to trigger playback and capture at the same time using the api's mentioned in this thread. Is my understanding correct?Or am I completely off track here? Kernel used is:3.0 > > snd_pcm_link() > Awesome! I will look into this API > > > If the driver doesn't actually implement starting of linked streams > > (which would be pointless if the hardware doesn't support it), ALSA > > calls the second trigger callback after the first one. > > > > It might be possible to implement simultaneous starting of two streams. > > What hardware (driver) are you using? > It's Qualcomm hardware. As far as I know, HW has this synchronization > support but I would have to build driver to make use of this functionality > To see examples of drivers that support HW linking, search for > snd_pcm_group_for_each_entry

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[alsa-devel] [PATCH] ASoC: Remove LZO cache type
by Mark Brown 10 Nov '11

10 Nov '11

There are no current users and new drivers ought to be using the regmap API and its cache implementation directly so just delete the ASoC copy. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 - sound/soc/Kconfig | 15 -- sound/soc/soc-cache.c | 384 ------------------------------------------------- 3 files changed, 0 insertions(+), 400 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c39ac19..c12cee6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -273,7 +273,6 @@ enum snd_soc_control_type { enum snd_soc_compress_type { SND_SOC_FLAT_COMPRESSION = 1, - SND_SOC_LZO_COMPRESSION, SND_SOC_RBTREE_COMPRESSION }; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 1381db8..35e662d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,21 +22,6 @@ menuconfig SND_SOC if SND_SOC -config SND_SOC_CACHE_LZO - bool "Support LZO compression for register caches" - select LZO_COMPRESS - select LZO_DECOMPRESS - ---help--- - Select this to enable LZO compression for register caches. - This will allow machine or CODEC drivers to compress register - caches in memory, reducing the memory consumption at the - expense of performance. If this is not present and is used - the system will fall back to uncompressed caches. - - Usually it is safe to disable this option, where cache - compression in used the rbtree option will typically perform - better. - config SND_SOC_AC97_BUS bool diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9077aa4..18bb6b3 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -14,7 +14,6 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> #include <sound/soc.h> -#include <linux/lzo.h> #include <linux/bitmap.h> #include <linux/rbtree.h> #include <linux/export.h> @@ -439,378 +438,6 @@ err: return ret; } -#ifdef CONFIG_SND_SOC_CACHE_LZO -struct snd_soc_lzo_ctx { - void *wmem; - void *dst; - const void *src; - size_t src_len; - size_t dst_len; - size_t decompressed_size; - unsigned long *sync_bmp; - int sync_bmp_nbits; -}; - -#define LZO_BLOCK_NUM 8 -static int snd_soc_lzo_block_count(void) -{ - return LZO_BLOCK_NUM; -} - -static int snd_soc_lzo_prepare(struct snd_soc_lzo_ctx *lzo_ctx) -{ - lzo_ctx->wmem = kmalloc(LZO1X_MEM_COMPRESS, GFP_KERNEL); - if (!lzo_ctx->wmem) - return -ENOMEM; - return 0; -} - -static int snd_soc_lzo_compress(struct snd_soc_lzo_ctx *lzo_ctx) -{ - size_t compress_size; - int ret; - - ret = lzo1x_1_compress(lzo_ctx->src, lzo_ctx->src_len, - lzo_ctx->dst, &compress_size, lzo_ctx->wmem); - if (ret != LZO_E_OK || compress_size > lzo_ctx->dst_len) - return -EINVAL; - lzo_ctx->dst_len = compress_size; - return 0; -} - -static int snd_soc_lzo_decompress(struct snd_soc_lzo_ctx *lzo_ctx) -{ - size_t dst_len; - int ret; - - dst_len = lzo_ctx->dst_len; - ret = lzo1x_decompress_safe(lzo_ctx->src, lzo_ctx->src_len, - lzo_ctx->dst, &dst_len); - if (ret != LZO_E_OK || dst_len != lzo_ctx->dst_len) - return -EINVAL; - return 0; -} - -static int snd_soc_lzo_compress_cache_block(struct snd_soc_codec *codec, - struct snd_soc_lzo_ctx *lzo_ctx) -{ - int ret; - - lzo_ctx->dst_len = lzo1x_worst_compress(PAGE_SIZE); - lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); - if (!lzo_ctx->dst) { - lzo_ctx->dst_len = 0; - return -ENOMEM; - } - - ret = snd_soc_lzo_compress(lzo_ctx); - if (ret < 0) - return ret; - return 0; -} - -static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec, - struct snd_soc_lzo_ctx *lzo_ctx) -{ - int ret; - - lzo_ctx->dst_len = lzo_ctx->decompressed_size; - lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); - if (!lzo_ctx->dst) { - lzo_ctx->dst_len = 0; - return -ENOMEM; - } - - ret = snd_soc_lzo_decompress(lzo_ctx); - if (ret < 0) - return ret; - return 0; -} - -static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - return (reg * codec_drv->reg_word_size) / - DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); -} - -static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) / - codec_drv->reg_word_size); -} - -static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) -{ - return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); -} - -static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - unsigned int val; - int i; - int ret; - - lzo_blocks = codec->reg_cache; - for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { - WARN_ON(!snd_soc_codec_writable_register(codec, i)); - ret = snd_soc_cache_read(codec, i, &val); - if (ret) - return ret; - codec->cache_bypass = 1; - ret = snd_soc_write(codec, i, val); - codec->cache_bypass = 0; - if (ret) - return ret; - dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - i, val); - } - - return 0; -} - -static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; - int ret, blkindex, blkpos; - size_t blksize, tmp_dst_len; - void *tmp_dst; - - /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); - /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); - /* size of the compressed block */ - blksize = snd_soc_lzo_get_blksize(codec); - lzo_blocks = codec->reg_cache; - lzo_block = lzo_blocks[blkindex]; - - /* save the pointer and length of the compressed block */ - tmp_dst = lzo_block->dst; - tmp_dst_len = lzo_block->dst_len; - - /* prepare the source to be the compressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* decompress the block */ - ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret < 0) { - kfree(lzo_block->dst); - goto out; - } - - /* write the new value to the cache */ - if (snd_soc_set_cache_val(lzo_block->dst, blkpos, value, - codec->driver->reg_word_size)) { - kfree(lzo_block->dst); - goto out; - } - - /* prepare the source to be the decompressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* compress the block */ - ret = snd_soc_lzo_compress_cache_block(codec, lzo_block); - if (ret < 0) { - kfree(lzo_block->dst); - kfree(lzo_block->src); - goto out; - } - - /* set the bit so we know we have to sync this register */ - set_bit(reg, lzo_block->sync_bmp); - kfree(tmp_dst); - kfree(lzo_block->src); - return 0; -out: - lzo_block->dst = tmp_dst; - lzo_block->dst_len = tmp_dst_len; - return ret; -} - -static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value) -{ - struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; - int ret, blkindex, blkpos; - size_t blksize, tmp_dst_len; - void *tmp_dst; - - *value = 0; - /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); - /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); - /* size of the compressed block */ - blksize = snd_soc_lzo_get_blksize(codec); - lzo_blocks = codec->reg_cache; - lzo_block = lzo_blocks[blkindex]; - - /* save the pointer and length of the compressed block */ - tmp_dst = lzo_block->dst; - tmp_dst_len = lzo_block->dst_len; - - /* prepare the source to be the compressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* decompress the block */ - ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret >= 0) - /* fetch the value from the cache */ - *value = snd_soc_get_cache_val(lzo_block->dst, blkpos, - codec->driver->reg_word_size); - - kfree(lzo_block->dst); - /* restore the pointer and length of the compressed block */ - lzo_block->dst = tmp_dst; - lzo_block->dst_len = tmp_dst_len; - return 0; -} - -static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - int i, blkcount; - - lzo_blocks = codec->reg_cache; - if (!lzo_blocks) - return 0; - - blkcount = snd_soc_lzo_block_count(); - /* - * the pointer to the bitmap used for syncing the cache - * is shared amongst all lzo_blocks. Ensure it is freed - * only once. - */ - if (lzo_blocks[0]) - kfree(lzo_blocks[0]->sync_bmp); - for (i = 0; i < blkcount; ++i) { - if (lzo_blocks[i]) { - kfree(lzo_blocks[i]->wmem); - kfree(lzo_blocks[i]->dst); - } - /* each lzo_block is a pointer returned by kmalloc or NULL */ - kfree(lzo_blocks[i]); - } - kfree(lzo_blocks); - codec->reg_cache = NULL; - return 0; -} - -static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - size_t bmp_size; - const struct snd_soc_codec_driver *codec_drv; - int ret, tofree, i, blksize, blkcount; - const char *p, *end; - unsigned long *sync_bmp; - - ret = 0; - codec_drv = codec->driver; - - /* - * If we have not been given a default register cache - * then allocate a dummy zero-ed out region, compress it - * and remember to free it afterwards. - */ - tofree = 0; - if (!codec->reg_def_copy) - tofree = 1; - - if (!codec->reg_def_copy) { - codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL); - if (!codec->reg_def_copy) - return -ENOMEM; - } - - blkcount = snd_soc_lzo_block_count(); - codec->reg_cache = kzalloc(blkcount * sizeof *lzo_blocks, - GFP_KERNEL); - if (!codec->reg_cache) { - ret = -ENOMEM; - goto err_tofree; - } - lzo_blocks = codec->reg_cache; - - /* - * allocate a bitmap to be used when syncing the cache with - * the hardware. Each time a register is modified, the corresponding - * bit is set in the bitmap, so we know that we have to sync - * that register. - */ - bmp_size = codec_drv->reg_cache_size; - sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof(long), - GFP_KERNEL); - if (!sync_bmp) { - ret = -ENOMEM; - goto err; - } - bitmap_zero(sync_bmp, bmp_size); - - /* allocate the lzo blocks and initialize them */ - for (i = 0; i < blkcount; ++i) { - lzo_blocks[i] = kzalloc(sizeof **lzo_blocks, - GFP_KERNEL); - if (!lzo_blocks[i]) { - kfree(sync_bmp); - ret = -ENOMEM; - goto err; - } - lzo_blocks[i]->sync_bmp = sync_bmp; - lzo_blocks[i]->sync_bmp_nbits = bmp_size; - /* alloc the working space for the compressed block */ - ret = snd_soc_lzo_prepare(lzo_blocks[i]); - if (ret < 0) - goto err; - } - - blksize = snd_soc_lzo_get_blksize(codec); - p = codec->reg_def_copy; - end = codec->reg_def_copy + codec->reg_size; - /* compress the register map and fill the lzo blocks */ - for (i = 0; i < blkcount; ++i, p += blksize) { - lzo_blocks[i]->src = p; - if (p + blksize > end) - lzo_blocks[i]->src_len = end - p; - else - lzo_blocks[i]->src_len = blksize; - ret = snd_soc_lzo_compress_cache_block(codec, - lzo_blocks[i]); - if (ret < 0) - goto err; - lzo_blocks[i]->decompressed_size = - lzo_blocks[i]->src_len; - } - - if (tofree) { - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - } - return 0; -err: - snd_soc_cache_exit(codec); -err_tofree: - if (tofree) { - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - } - return ret; -} -#endif - static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) { int i; @@ -889,17 +516,6 @@ static const struct snd_soc_cache_ops cache_types[] = { .write = snd_soc_flat_cache_write, .sync = snd_soc_flat_cache_sync }, -#ifdef CONFIG_SND_SOC_CACHE_LZO - { - .id = SND_SOC_LZO_COMPRESSION, - .name = "LZO", - .init = snd_soc_lzo_cache_init, - .exit = snd_soc_lzo_cache_exit, - .read = snd_soc_lzo_cache_read, - .write = snd_soc_lzo_cache_write, - .sync = snd_soc_lzo_cache_sync - }, -#endif { .id = SND_SOC_RBTREE_COMPRESSION, .name = "rbtree", -- 1.7.7.2

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[alsa-devel] [PATCH v4] ASoC: Add new Realtek ALC5632 CODEC driver.
by leon＠leon.nu 10 Nov '11

10 Nov '11

From: Leon Romanovsky <leon(a)leon.nu> This driver implements basic functionality, using I²C for the control channel. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> --- v4: * Use dev_info instead dev_err for information messages. * Use EINVAL for incorrect pll_id. * Mark cache as needed to sync on resume. * Use mask for power registers to protect reserved bits. * Move power management logic from probe to set_bias function. * Remove references to sound platform device. v3: * DAC is now handled by DAPM * Power down control correct handling. * Fixed speaker dapm routes. * Remove unused variables. * Sync stream names in DAPM and codec dai. * Use update bits instead write to reg in power depop function. * Rename Route to AB-D Amp Mux * Add snd controls to codec driver. Don't register it in probe. * Add dapm widgets and routes to codec driver. Don't register it. * Don't force power for spk apm, main i2s. It is described in DAPM. * Set symmetric_rates flag to prevent different rates for playback and capture. * Removed commented DEBUG definition. * Use SOC infrastructure sync cache mechanism. * Remove non possible IF switch. * Fix indentation issues * Convert alc5632 MICBIAS to a supply widget * Remove codec input parameter * Remove VERSION define * Remove unnecessary dev_kfree() and update copyright notice * Control speaker amplifier with DAPM. * Let machine driver choose master/slave mode for the codec. * Clean up. removed dev_dbg. * Use devm_kzalloc/devm_kzfree for memory allocation/deallocation. * Fixed stereo headphones playback. * Set up correct name for Stereo DAC volume item. v2: Free from checkpatch warnings v1: Initial code drop --- --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/alc5632.c | 1153 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/alc5632.h | 249 ++++++++++ 4 files changed, 1407 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/alc5632.c create mode 100644 sound/soc/codecs/alc5632.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..684cc15 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C + select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -168,6 +169,8 @@ config SND_SOC_AK4671 config SND_SOC_ALC5623 tristate +config SND_SOC_ALC5632 + tristate config SND_SOC_CQ0093VC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..2ed5790 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o +snd-soc-alc5632-objs := alc5632.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -113,6 +114,7 @@ obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c new file mode 100644 index 0000000..68465d8 --- /dev/null +++ b/sound/soc/codecs/alc5632.c @@ -0,0 +1,1153 @@ +/* +* alc5632.c -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.c by Arnaud Patard +* +* 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/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/initval.h> + +#include "alc5632.h" + +/* + * ALC5632 register cache + */ +static const u16 alc5632_reg_defaults[] = { + 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ + 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ + 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ + 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ + 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ + 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ + 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ + 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ + 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ + 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ + 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ + 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ + 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ + 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ + 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ + 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ + 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +}; + +/* codec private data */ +struct alc5632_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; +}; + +static int alc5632_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ALC5632_RESET: + case ALC5632_PWR_DOWN_CTRL_STATUS: + case ALC5632_GPIO_PIN_STATUS: + case ALC5632_OVER_CURR_STATUS: + case ALC5632_HID_CTRL_DATA: + case ALC5632_EQ_CTRL: + return 1; + + default: + break; + } + + return 0; +} + +static inline int alc5632_reset(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ALC5632_RESET, 0); + return snd_soc_read(codec, ALC5632_RESET); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5632 Controls + */ + +/* -34.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +/* -46.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +/* -16.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +/* 0db min scale, 6 db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); +/* 0db min scalem 0.75db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); + +static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { + /* left starts at bit 8, right at bit 0 */ + /* 31 steps (5 bit), -46.5db scale */ + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + /* bit 15 mutes left, bit 7 right */ + SOC_DOUBLE("Line Playback Switch", + ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5632_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Voice DAC Playback Volume", + ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE_TLV("Phone Capture Volume", + ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Stereo DAC Playback Volume", + ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), + SOC_DOUBLE("Stereo DAC Playback Switch", + ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1), +SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1), +SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1), +SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 9, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1), +SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 10, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1), +SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5632_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Mute"}; +static const char *alc5632_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5632_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5632_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5632_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5632_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static const struct snd_kcontrol_new alc5632_auxout_mux_controls = +SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5632_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static const struct snd_kcontrol_new alc5632_spkout_mux_controls = +SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5632_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = +SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5632_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = +SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5632_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = +SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); + +/* speaker amplifier */ +static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5632_amp_enum = + SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static const struct snd_kcontrol_new alc5632_amp_mux_controls = + SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); + + +static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5632_hp_mixer_controls[0], + ARRAY_SIZE(alc5632_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0, + &alc5632_hpr_mixer_controls[0], + ARRAY_SIZE(alc5632_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0, + &alc5632_hpl_mixer_controls[0], + ARRAY_SIZE(alc5632_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, + &alc5632_mono_mixer_controls[0], + ARRAY_SIZE(alc5632_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, + &alc5632_speaker_mixer_controls[0], + ARRAY_SIZE(alc5632_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, + &alc5632_captureL_mixer_controls[0], + ARRAY_SIZE(alc5632_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, + &alc5632_captureR_mixer_controls[0], + ARRAY_SIZE(alc5632_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0), +SND_SOC_DAPM_MIXER("DAC Right Channel", + ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", ALC5632_PWR_MANAG_ADD1, 10, 0, + &alc5632_amp_mux_controls), + +SND_SOC_DAPM_OUTPUT("AUXOUT"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("PHONEP"), +SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + + +static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"Phone Mix", NULL, "Phone"}, + {"Phone Mix", NULL, "Phone ADMix"}, + {"AUXOUT", NULL, "Aux Out"}, + + /* DAC */ + {"DAC Right Channel", NULL, "I2S Mix"}, + {"DAC Left Channel", NULL, "I2S Mix"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + + {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, + {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, + + + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Aux Out", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Phone", NULL, "PHONEP"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"SpeakerOut N Mux", "RN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Right Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"Left Speaker", NULL, "AB-D Amp Mux"}, + {"Right Speaker", NULL, "AB-D Amp Mux"}, + + {"SPKOUT", NULL, "Left Speaker"}, + {"SPKOUT", NULL, "Right Speaker"}, + + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, + +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5632 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +/* FOUT = MCLK*(N+2)/((M+2)*(K+2)) + N: bit 15:8 (div 2 .. div 257) + K: bit 6:4 typical 2 + M: bit 3:0 (div 2 .. div 17) + + same as for 5623 - thanks! +*/ + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5632_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK) + return -EINVAL; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5632_DAI_CONTROL); + if (reg & ALC5632_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5632_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_BCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5632_PLL_FR_BCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_VBCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from voice clock */ + gbl_clk = ALC5632_PLL_FR_VBCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + + /* choose MCLK/BCLK/VBCLK */ + snd_soc_write(codec, ALC5632_GPCR2, gbl_clk); + /* choose PLL1 clock rate */ + snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div); + /* enable PLL1 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + ALC5632_PWR_ADD2_PLL1); + /* enable PLL2 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + ALC5632_PWR_ADD2_PLL2); + /* use PLL1 as main SYSCLK */ + snd_soc_update_bits(codec, ALC5632_GPCR1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {512*1, 0x3075}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5632->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5632->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5632_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5632_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5632_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5632_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5632_DAI_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5632_DAI_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); +} + +static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5632_DAI_CONTROL); + iface &= ~ALC5632_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5632_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5632_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5632_DAI_I2S_DL_24; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff); + + return 0; +} + +static int alc5632_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L \ + |ALC5632_MISC_HP_DEPOP_MUTE_R; + u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg); +} + +#define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF) + +#define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5632_ADD1_POWER_EN \ + (ALC5632_PWR_ADD1_DAC_REF \ + | ALC5632_PWR_ADD1_SOFTGEN_EN \ + | ALC5632_PWR_ADD1_HP_OUT_AMP \ + | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \ + | ALC5632_PWR_ADD1_MAIN_BIAS) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_SOFTGEN_EN, + ALC5632_PWR_ADD1_SOFTGEN_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, + ALC5632_ADD3_POWER_EN, + ALC5632_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + ALC5632_MISC_HP_DEPOP_MODE2_EN); + + /* "normal" mode: 0 @ 26 */ + /* set all PR0-7 mixers to 0 */ + snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, + ALC5632_PWR_DOWN_CTRL_STATUS_MASK, + 0); + + msleep(500); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_ADD2_POWER_EN, + ALC5632_ADD2_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_ADD1_POWER_EN, + ALC5632_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5632_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_MANAG_ADD1_MASK, + ALC5632_PWR_ADD1_MAIN_BIAS); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_MANAG_ADD2_MASK, + ALC5632_PWR_ADD2_VREF); + /* "normal" mode: 0 @ 26 */ + snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, + ALC5632_PWR_DOWN_CTRL_STATUS_MASK, + 0xffff ^ (ALC5632_PWR_VREF_PR3 + | ALC5632_PWR_VREF_PR2)); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_MANAG_ADD2_MASK, 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, + ALC5632_PWR_MANAG_ADD3_MASK, 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_MANAG_ADD1_MASK, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5632_dai_ops = { + .hw_params = alc5632_pcm_hw_params, + .digital_mute = alc5632_mute, + .set_fmt = alc5632_set_dai_fmt, + .set_sysclk = alc5632_set_dai_sysclk, + .set_pll = alc5632_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5632_dai = { + .name = "alc5632-hifi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + + .ops = &alc5632_dai_ops, + .symmetric_rates = 1, +}; + +static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5632_resume(struct snd_soc_codec *codec) +{ + int ret; + + /* mark cache as needed to sync */ + codec->cache_sync = 1; + + ret = snd_soc_cache_sync(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ + | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ + | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ + | ALC5632_ADC_REC_MONOMIX) + +#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ + | ALC5632_MIC_ROUTE_SPK \ + | ALC5632_MIC_ROUTE_MONOMIX) + +#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ + | ALC5632_PWR_DAC_STATUS \ + | ALC5632_PWR_AMIX_STATUS \ + | ALC5632_PWR_VREF_STATUS) + +#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ + / ALC5632_ADC_REC_GAIN_STEP) + +#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) + +#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) + +static int alc5632_probe(struct snd_soc_codec *codec) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5632_reset(codec); + + /* power on device */ + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + switch (alc5632->id) { + case 0x5c: + snd_soc_add_controls(codec, alc5632_vol_snd_controls, + ARRAY_SIZE(alc5632_vol_snd_controls)); + break; + default: + return -EINVAL; + } + + return ret; +} + +/* power down chip */ +static int alc5632_remove(struct snd_soc_codec *codec) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5632 = { + .probe = alc5632_probe, + .remove = alc5632_remove, + .suspend = alc5632_suspend, + .resume = alc5632_resume, + .set_bias_level = alc5632_set_bias_level, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = alc5632_reg_defaults, + .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_register = alc5632_volatile_register, + .controls = alc5632_snd_controls, + .num_controls = ARRAY_SIZE(alc5632_snd_controls), + .dapm_widgets = alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(alc5632_dapm_widgets), + .dapm_routes = alc5632_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), +}; + +/* + * alc5632 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5632_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5632_priv *alc5632; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid1: %x\n", vid1); + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid2: %x\n", vid2); + } + vid2 = (vid2 & 0xff); + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + alc5632 = devm_kzalloc(&client->dev, + sizeof(struct alc5632_priv), GFP_KERNEL); + if (alc5632 == NULL) + return -ENOMEM; + + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + return -EINVAL; + } + + i2c_set_clientdata(client, alc5632); + alc5632->control_data = client; + alc5632->control_type = SND_SOC_I2C; + mutex_init(&alc5632->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5632, &alc5632_dai, 1); + if (ret != 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int alc5632_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id alc5632_i2c_table[] = { + {"alc5632", 0x5c}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5632_i2c_driver = { + .driver = { + .name = "alc5632", + .owner = THIS_MODULE, + }, + .probe = alc5632_i2c_probe, + .remove = __devexit_p(alc5632_i2c_remove), + .id_table = alc5632_i2c_table, +}; + +static int __init alc5632_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5632_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5632_modinit); + +static void __exit alc5632_modexit(void) +{ + i2c_del_driver(&alc5632_i2c_driver); +} +module_exit(alc5632_modexit); + +MODULE_DESCRIPTION("ASoC ALC5632 driver"); +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h new file mode 100644 index 0000000..ff4c0fd --- /dev/null +++ b/sound/soc/codecs/alc5632.h @@ -0,0 +1,249 @@ +/* +* alc5632.h -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.h by Arnaud Patard +* +* 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. +*/ + +#ifndef _ALC5632_H +#define _ALC5632_H + +#define ALC5632_RESET 0x00 +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5632_SPK_OUT_VOL 0x02 /* spe out vol */ +#define ALC5632_SPK_OUT_VOL_STEP 1.5 +#define ALC5632_HP_OUT_VOL 0x04 /* hp out vol */ +#define ALC5632_AUX_OUT_VOL 0x06 /* aux out vol */ +#define ALC5632_PHONE_IN_VOL 0x08 /* phone in vol */ +#define ALC5632_LINE_IN_VOL 0x0A /* line in vol */ +#define ALC5632_STEREO_DAC_IN_VOL 0x0C /* stereo dac in vol */ +#define ALC5632_MIC_VOL 0x0E /* mic in vol */ +/* stero dac/mic routing */ +#define ALC5632_MIC_ROUTING_CTRL 0x10 +#define ALC5632_MIC_ROUTE_MONOMIX (1 << 0) +#define ALC5632_MIC_ROUTE_SPK (1 << 1) +#define ALC5632_MIC_ROUTE_HP (1 << 2) + +#define ALC5632_ADC_REC_GAIN 0x12 /* rec gain */ +#define ALC5632_ADC_REC_GAIN_RANGE 0x1F1F +#define ALC5632_ADC_REC_GAIN_BASE (-16.5) +#define ALC5632_ADC_REC_GAIN_STEP 1.5 + +#define ALC5632_ADC_REC_MIXER 0x14 /* mixer control */ +#define ALC5632_ADC_REC_MIC1 (1 << 6) +#define ALC5632_ADC_REC_MIC2 (1 << 5) +#define ALC5632_ADC_REC_LINE_IN (1 << 4) +#define ALC5632_ADC_REC_AUX (1 << 3) +#define ALC5632_ADC_REC_HP (1 << 2) +#define ALC5632_ADC_REC_SPK (1 << 1) +#define ALC5632_ADC_REC_MONOMIX (1 << 0) + +#define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */ +/* ALC5632_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */ +#define ALC5632_OUTPUT_MIXER_RP (1 << 14) +#define ALC5632_OUTPUT_MIXER_WEEK (1 << 12) +#define ALC5632_OUTPUT_MIXER_HP (1 << 10) +#define ALC5632_OUTPUT_MIXER_AUX_SPK (2 << 6) +#define ALC5632_OUTPUT_MIXER_AUX_HP_LR (1 << 6) +#define ALC5632_OUTPUT_MIXER_HP_R (1 << 8) +#define ALC5632_OUTPUT_MIXER_HP_L (1 << 9) + +#define ALC5632_MIC_CTRL 0x22 /* mic phone ctrl */ +#define ALC5632_MIC_BOOST_BYPASS 0 +#define ALC5632_MIC_BOOST_20DB 1 +#define ALC5632_MIC_BOOST_30DB 2 +#define ALC5632_MIC_BOOST_40DB 3 + +#define ALC5632_DIGI_BOOST_CTRL 0x24 /* digi mic / bost ctl */ +#define ALC5632_MIC_BOOST_RANGE 7 +#define ALC5632_MIC_BOOST_STEP 6 +#define ALC5632_PWR_DOWN_CTRL_STATUS 0x26 +#define ALC5632_PWR_DOWN_CTRL_STATUS_MASK 0xEF00 +#define ALC5632_PWR_VREF_PR3 (1 << 11) +#define ALC5632_PWR_VREF_PR2 (1 << 10) +#define ALC5632_PWR_VREF_STATUS (1 << 3) +#define ALC5632_PWR_AMIX_STATUS (1 << 2) +#define ALC5632_PWR_DAC_STATUS (1 << 1) +#define ALC5632_PWR_ADC_STATUS (1 << 0) +/* stereo/voice DAC / stereo adc func ctrl */ +#define ALC5632_DAC_FUNC_SELECT 0x2E + +/* Main serial data port ctrl (i2s) */ +#define ALC5632_DAI_CONTROL 0x34 + +#define ALC5632_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5632_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5632_DAI_SADLRCK_MODE (1 << 14) +/* 0:voice, 1:main */ +#define ALC5632_DAI_MAIN_I2S_SYSCLK_SEL (1 << 8) +#define ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_MAIN_I2S_LRCK_INV (1 << 6) +#define ALC5632_DAI_I2S_DL_MASK (3 << 2) +#define ALC5632_DAI_I2S_DL_8 (3 << 2) +#define ALC5632_DAI_I2S_DL_24 (2 << 2) +#define ALC5632_DAI_I2S_DL_20 (1 << 2) +#define ALC5632_DAI_I2S_DL_16 (0 << 2) +#define ALC5632_DAI_I2S_DF_MASK (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_B (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_A (2 << 0) +#define ALC5632_DAI_I2S_DF_LEFT (1 << 0) +#define ALC5632_DAI_I2S_DF_I2S (0 << 0) +/* extend serial data port control (VoDAC_i2c/pcm) */ +#define ALC5632_DAI_CONTROL2 0x36 +/* 0:gpio func, 1:voice pcm */ +#define ALC5632_DAI_VOICE_PCM_ENABLE (1 << 15) +/* 0:master, 1:slave */ +#define ALC5632_DAI_VOICE_MODE_SEL (1 << 14) +/* 0:disable, 1:enable */ +#define ALC5632_DAI_HPF_CLK_CTRL (1 << 13) +/* 0:main, 1:voice */ +#define ALC5632_DAI_VOICE_I2S_SYSCLK_SEL (1 << 8) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_VBCLK_SYSCLK_SEL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_I2S_LR_INV (1 << 6) +#define ALC5632_DAI_VOICE_DL_MASK (3 << 2) +#define ALC5632_DAI_VOICE_DL_16 (0 << 2) +#define ALC5632_DAI_VOICE_DL_20 (1 << 2) +#define ALC5632_DAI_VOICE_DL_24 (2 << 2) +#define ALC5632_DAI_VOICE_DL_8 (3 << 2) +#define ALC5632_DAI_VOICE_DF_MASK (3 << 0) +#define ALC5632_DAI_VOICE_DF_I2S (0 << 0) +#define ALC5632_DAI_VOICE_DF_LEFT (1 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_A (2 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_B (3 << 0) + +#define ALC5632_PWR_MANAG_ADD1 0x3A +#define ALC5632_PWR_MANAG_ADD1_MASK 0xEFFF +#define ALC5632_PWR_ADD1_DAC_L_EN (1 << 15) +#define ALC5632_PWR_ADD1_DAC_R_EN (1 << 14) +#define ALC5632_PWR_ADD1_ZERO_CROSS (1 << 13) +#define ALC5632_PWR_ADD1_MAIN_I2S_EN (1 << 11) +#define ALC5632_PWR_ADD1_SPK_AMP_EN (1 << 10) +#define ALC5632_PWR_ADD1_HP_OUT_AMP (1 << 9) +#define ALC5632_PWR_ADD1_HP_OUT_ENH_AMP (1 << 8) +#define ALC5632_PWR_ADD1_VOICE_DAC_MIX (1 << 7) +#define ALC5632_PWR_ADD1_SOFTGEN_EN (1 << 6) +#define ALC5632_PWR_ADD1_MIC1_SHORT_CURR (1 << 5) +#define ALC5632_PWR_ADD1_MIC2_SHORT_CURR (1 << 4) +#define ALC5632_PWR_ADD1_MIC1_EN (1 << 3) +#define ALC5632_PWR_ADD1_MIC2_EN (1 << 2) +#define ALC5632_PWR_ADD1_MAIN_BIAS (1 << 1) +#define ALC5632_PWR_ADD1_DAC_REF (1 << 0) + +#define ALC5632_PWR_MANAG_ADD2 0x3C +#define ALC5632_PWR_MANAG_ADD2_MASK 0x7FFF +#define ALC5632_PWR_ADD2_PLL1 (1 << 15) +#define ALC5632_PWR_ADD2_PLL2 (1 << 14) +#define ALC5632_PWR_ADD2_VREF (1 << 13) +#define ALC5632_PWR_ADD2_OVT_DET (1 << 12) +#define ALC5632_PWR_ADD2_VOICE_DAC (1 << 10) +#define ALC5632_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5632_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5632_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5632_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5632_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5632_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5632_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5632_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5632_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5632_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5632_PWR_MANAG_ADD3 0x3E +#define ALC5632_PWR_MANAG_ADD3_MASK 0x7CFF +#define ALC5632_PWR_ADD3_AUXOUT_VOL (1 << 14) +#define ALC5632_PWR_ADD3_SPK_L_OUT (1 << 13) +#define ALC5632_PWR_ADD3_SPK_R_OUT (1 << 12) +#define ALC5632_PWR_ADD3_HP_L_OUT_VOL (1 << 11) +#define ALC5632_PWR_ADD3_HP_R_OUT_VOL (1 << 10) +#define ALC5632_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5632_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5632_PWR_ADD3_AUXIN_VOL (1 << 5) +#define ALC5632_PWR_ADD3_AUXIN_MIX (1 << 4) +#define ALC5632_PWR_ADD3_MIC1_VOL (1 << 3) +#define ALC5632_PWR_ADD3_MIC2_VOL (1 << 2) +#define ALC5632_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5632_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5632_GPCR1 0x40 +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1 (1 << 15) +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_MCLK (0 << 15) +#define ALC5632_GPCR1_DAC_HI_FLT_EN (1 << 10) +#define ALC5632_GPCR1_SPK_AMP_CTRL (7 << 1) +#define ALC5632_GPCR1_VDD_100 (5 << 1) +#define ALC5632_GPCR1_VDD_125 (4 << 1) +#define ALC5632_GPCR1_VDD_150 (3 << 1) +#define ALC5632_GPCR1_VDD_175 (2 << 1) +#define ALC5632_GPCR1_VDD_200 (1 << 1) +#define ALC5632_GPCR1_VDD_225 (0 << 1) + +#define ALC5632_GPCR2 0x42 +#define ALC5632_GPCR2_PLL1_SOUR_SEL (3 << 12) +#define ALC5632_PLL_FR_MCLK (0 << 12) +#define ALC5632_PLL_FR_BCLK (2 << 12) +#define ALC5632_PLL_FR_VBCLK (3 << 12) +#define ALC5632_GPCR2_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5632_PLL1_CTRL 0x44 +#define ALC5632_PLL1_CTRL_N_VAL(n) (((n) & 0x0f) << 8) +#define ALC5632_PLL1_M_BYPASS (1 << 7) +#define ALC5632_PLL1_CTRL_K_VAL(k) (((k) & 0x07) << 4) +#define ALC5632_PLL1_CTRL_M_VAL(m) (((m) & 0x0f) << 0) + +#define ALC5632_PLL2_CTRL 0x46 +#define ALC5632_PLL2_EN (1 << 15) +#define ALC5632_PLL2_RATIO (0 << 15) + +#define ALC5632_GPIO_PIN_CONFIG 0x4C +#define ALC5632_GPIO_PIN_POLARITY 0x4E +#define ALC5632_GPIO_PIN_STICKY 0x50 +#define ALC5632_GPIO_PIN_WAKEUP 0x52 +#define ALC5632_GPIO_PIN_STATUS 0x54 +#define ALC5632_GPIO_PIN_SHARING 0x56 +#define ALC5632_OVER_CURR_STATUS 0x58 +#define ALC5632_SOFTVOL_CTRL 0x5A +#define ALC5632_GPIO_OUPUT_PIN_CTRL 0x5C + +#define ALC5632_MISC_CTRL 0x5E +#define ALC5632_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5632_MISC_AVC_TRGT_SEL (3 << 12) +#define ALC5632_MISC_AVC_TRGT_RIGHT (1 << 12) +#define ALC5632_MISC_AVC_TRGT_LEFT (2 << 12) +#define ALC5632_MISC_AVC_TRGT_BOTH (3 << 12) +#define ALC5632_MISC_HP_DEPOP_MODE1_EN (1 << 9) +#define ALC5632_MISC_HP_DEPOP_MODE2_EN (1 << 8) +#define ALC5632_MISC_HP_DEPOP_MUTE_L (1 << 7) +#define ALC5632_MISC_HP_DEPOP_MUTE_R (1 << 6) +#define ALC5632_MISC_HP_DEPOP_MUTE (1 << 5) +#define ALC5632_MISC_GPIO_WAKEUP_CTRL (1 << 1) +#define ALC5632_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5632_DAC_CLK_CTRL1 0x60 +#define ALC5632_DAC_CLK_CTRL2 0x62 +#define ALC5632_DAC_CLK_CTRL2_DIV1_2 (1 << 0) +#define ALC5632_VOICE_DAC_PCM_CLK_CTRL1 0x64 +#define ALC5632_PSEUDO_SPATIAL_CTRL 0x68 +#define ALC5632_HID_CTRL_INDEX 0x6A +#define ALC5632_HID_CTRL_DATA 0x6C +#define ALC5632_EQ_CTRL 0x6E + +/* undocumented */ +#define ALC5632_VENDOR_ID1 0x7C +#define ALC5632_VENDOR_ID2 0x7E + +#endif -- 1.7.3.4

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[alsa-devel] [PATCH] MIPS/sound: DB1200 audio support depends on MIPS arch.
by Manuel Lauss 10 Nov '11

10 Nov '11

DB1200 audio support should only be visible if MIPS/Alchemy arch is selected. Fixes an x86-allmodconfig build failure in linux-next. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index c620cbf9..a561040 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -52,7 +52,7 @@ config SND_SOC_DB1000 config SND_SOC_DB1200 tristate "DB1200/DB1300/DB1550 Audio support" - select SND_SOC_AU1XPSC + depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC select SND_SOC_WM9712 -- 1.7.7.2 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] MAINTAINERS: Drop inactive Samsung ASoC maintainer
by Jassi Brar 10 Nov '11

10 Nov '11

Signed-off-by: Jassi Brar <jassisinghbrar(a)gmail.com> --- MAINTAINERS | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 2014c1f..c1a96f9 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5581,7 +5581,6 @@ F: drivers/media/video/*7146* F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS -M: Jassi Brar <jassisinghbrar(a)gmail.com> M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported -- 1.7.4.1

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Re: [alsa-devel] Linux 3.2-rc1
by Linus Torvalds 10 Nov '11

10 Nov '11

Hmm, I don't know what caused this to trigger, but I'm adding both the i915 people and the HDA people to the cc, and they can fight to the death about this in the HDMI Thunderdome. Guys: One.. Two.. Three.. FIGHT! Linus On Tue, Nov 8, 2011 at 6:55 AM, Nick Bowler <nbowler(a)elliptictech.com> wrote: > > Mode switches are very noisy on an Intel G45 in 3.2-rc1: > > HDMI hot plug event: Codec=3 Pin=3 Presence_Detect=1 ELD_Valid=1 > HDMI status: Codec=3 Pin=3 Presence_Detect=1 ELD_Valid=1 > HDMI: detected monitor W2253 at connection type HDMI > HDMI: available speakers: FL/FR > HDMI: supports coding type LPCM: channels = 2, rates = 32000 44100 48000, bits = 16 20 24 > > These lines get printed every single switch; previously only a single > line was printed once at boot (the "HDMI status" line). > > Cheers, > -- > Nick Bowler, Elliptic Technologies (http://www.elliptictech.com/) >

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[alsa-devel] [PATCH] ASoC: Remove extra space in runtime struct definition
by Mark Brown 10 Nov '11

10 Nov '11

My usual technique for finding definitions is to search for "name {" which breaks with the extra space. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 877fcc1..02a5c55 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -843,7 +843,7 @@ struct snd_soc_card { }; /* SoC machine DAI configuration, glues a codec and cpu DAI together */ -struct snd_soc_pcm_runtime { +struct snd_soc_pcm_runtime { struct device dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; -- 1.7.7.2

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Mark Brown 10 Nov '11

10 Nov '11

On Thu, Nov 10, 2011 at 02:00:22PM +0900, Tomoya MORINAGA wrote: > (2011/11/08 23:38), Mark Brown wrote: > >On Tue, Nov 08, 2011 at 06:03:53PM +0900, Tomoya MORINAGA wrote: > >>struct snd_ml7213i2s_pcm { > >> enum snd_soc_control_type control_type; > >> struct snd_ml7213i2s *ml7213i2s; > >> spinlock_t lock; > >> unsigned int irq_pos; > >> unsigned int buf_pos; > >> struct snd_pcm_substream *substream; > >> struct cbdata cbd; /* i2s callback info */ > >> unsigned int channels; > >> unsigned int rw; > >> unsigned int rate; > >> unsigned int ch; > >> unsigned int setup_flag; > >> unsigned int format; > >> unsigned int bclkfs; > >> struct mutex i2c_mutex; > >>}; > >...this looks *really* confused, there's things in here which are a mix > >of DMA controller and CODEC driver things. The CODEC and DMA drivers > >shouldn't know anything about each other, let alone be referencing the > >same data structure. > Let me clarify your saying. > I couldn't understand what your "DMA controller" mean. > Which "snd_ml7213i2s_pcm" structure member do you mean ? Things like irq_pos and buf_pos for example.

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[alsa-devel] [PATCH 0/3] ALSA: intel8x0: Improve virtual environment detection
by Konstantin Ozerkov 10 Nov '11

10 Nov '11

Improve detection logic as discussed early: add secondary checks by PCI SSID. Note: patch [3/3] is made for alsa-driver tree.

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