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[alsa-devel] [PATCH TINYCOMPRESS 14/14] cplay: support auto-configuration of fragment size and count
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Default the fragment size and count to zero to take advantage of the auto-configuration support that was added to tinycompress. If user doesn't override these by specifying -f and -b values on the command line, they will be passed to tinycompress as zero and tinycompress will query the driver for size and count. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> diff --git a/cplay.c b/cplay.c index 3af6d13..60995a4 100644 --- a/cplay.c +++ b/cplay.c @@ -152,9 +152,9 @@ static int print_time(struct compress *compress) int main(int argc, char **argv) { char *file; - unsigned long buffer_size = 100*1024; + unsigned long buffer_size = 0; int c; - unsigned int card = 0, device = 0, frag = 4; + unsigned int card = 0, device = 0, frag = 0; if (argc < 2) @@ -276,8 +276,14 @@ void play_samples(char *name, unsigned int card, unsigned int device, codec.level = 0; codec.ch_mode = 0; codec.format = 0; - config.fragment_size = buffer_size/frag; - config.fragments = frag; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } config.codec = &codec; compress = compress_open(card, device, COMPRESS_IN, &config); -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 13/14] compress: Prevent divide-by-zero faults in get_hpointer()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Do not assume that the returned snd_compr_avail contains a snd_compr_tstamp with a valid sampling_rate. Check for zero before attempting to use it. diff --git a/compress.c b/compress.c index cfd0219..c0e81a3 100644 --- a/compress.c +++ b/compress.c @@ -297,10 +297,15 @@ int compress_get_hpointer(struct compress *compress, if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &kavail)) return oops(compress, errno, "cannot get avail"); *avail = (unsigned int)kavail.avail; - time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; - tstamp->tv_sec = time; - time = kavail.tstamp.pcm_io_frames % kavail.tstamp.sampling_rate; - tstamp->tv_nsec = time * 1000000000 / kavail.tstamp.sampling_rate; + if (kavail.tstamp.sampling_rate != 0) { + time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; + tstamp->tv_sec = time; + time = kavail.tstamp.pcm_io_frames % kavail.tstamp.sampling_rate; + tstamp->tv_nsec = time * 1000000000 / kavail.tstamp.sampling_rate; + } else { + tstamp->tv_sec = 0; + tstamp->tv_nsec = 0; + } return 0; } -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 12/14] compress: Add makefile for building under standard linux
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> diff --git a/makefile.linux b/makefile.linux new file mode 100644 index 0000000..60d6c04 --- /dev/null +++ b/makefile.linux @@ -0,0 +1,35 @@ +LIB = libtinycompress +BIN = cplay +VER = 0.1 +LIBSRC = compress.c +SRC = cplay.c +LIBOBJ = ${LIBSRC:.c=.o} +OBJ = ${SRC:.c=.o} + +CC = gcc +CROSS_COMPILE = + +CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include +LDFLAGS += -L. -ltinycompress +LIBLDFLAGS = -lasound + +all: libtinycompress cplay + +$(LIB): ${LIBOBJ} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} + +$(BIN): ${OBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + + +%.o: %.c + @echo " CC "$< + @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< + +clean: + @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + + +.PHONY: all clean -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 11/14] compress: Add missing header includes
by Richard Fitzgerald 22 Feb '13

22 Feb '13

This fixes build problems caused by dependencies that were not explicitly included. diff --git a/compress.c b/compress.c index 7ee287d..34936bd 100644 --- a/compress.c +++ b/compress.c @@ -66,6 +66,7 @@ #include <sys/time.h> #include <limits.h> +#include <linux/types.h> #include <linux/ioctl.h> #define __force #define __bitwise diff --git a/cplay.c b/cplay.c index d2a968c..3af6d13 100644 --- a/cplay.c +++ b/cplay.c @@ -51,6 +51,8 @@ * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. */ +#include <stdint.h> +#include <linux/types.h> #include <fcntl.h> #include <errno.h> #include <unistd.h> @@ -60,6 +62,7 @@ #include <signal.h> #include <stdbool.h> #include <getopt.h> +#include <sys/time.h> #define __force #define __bitwise #define __user -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 10/14] compress: Block if unable to write all data in single write()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Previously compress_write() would only block on poll() if the available space < fragment_size. If the device has a small fragment size this could lead to it never blocking and instead looping around doing many small writes. This is bad for power saving. If we were unable to write all the remaining data in a single write we want to block until the device reaches a buffer high water mark, to allow the CPU to sleep. This change will always attempt to issue the first write providing avail >= fragment_size. All subsequent loops will block on poll() before attempting another write() unless there is enough buffer space to write all remaining data. diff --git a/compress.c b/compress.c index 7c8ddc8..7ee287d 100644 --- a/compress.c +++ b/compress.c @@ -317,7 +317,8 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size { struct snd_compr_avail avail; struct pollfd fds; - int to_write, written, total = 0, ret; + int to_write = 0; /* zero indicates we haven't written yet */ + int written, total = 0, ret; const char* cbuf = buf; if (!(compress->flags & COMPRESS_IN)) @@ -333,24 +334,21 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) return oops(compress, errno, "cannot get avail"); - /* we will write only when avail > fragment size */ - if (avail.avail < compress->config.fragment_size) { - /* nothing to write so wait */ - ret = poll(&fds, 1, compress->max_poll_wait_ms); - /* A pause will cause -EBADFD or zero return from driver - * This is not an error, just stop writing + if ( (avail.avail < compress->config.fragment_size) + || ((to_write != 0) && (avail.avail < size)) ) { + /* not enough space for one fragment, or we have done + * a short write and there isn't enough space for all + * the remaining data */ + ret = poll(&fds, 1, compress->max_poll_wait_ms); + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop writing */ if ((ret == 0) || (ret == -EBADFD)) break; if (ret < 0) return oops(compress, errno, "poll error"); if (fds.revents & POLLOUT) { - if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) - return oops(compress, errno, "cannot get avail"); - if (avail.avail == 0) { - oops(compress, -EIO, "woken up even when avail is 0!!!"); - continue; - } + continue; } if (fds.revents & POLLERR) { return oops(compress, -EIO, "poll returned error!"); -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 9/14] compress: Allow client to pass "don't care" for fragment size/number
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Client cannot know what is a sensible fragment size and count for the device unless it has prior knowledge of the precise hardware. To allow for generic clients not hardcoded for particular hardware, this change allows fragment_size and fragments to be passed as 0, meaning "don't care" and tinycompress will choose default values read from the driver. diff --git a/compress.c b/compress.c index 7457550..7c8ddc8 100644 --- a/compress.c +++ b/compress.c @@ -123,19 +123,14 @@ int is_compress_ready(struct compress *compress) return (compress->fd > 0) ? 1 : 0; } -static bool _is_codec_supported(struct compress *compress, struct compr_config *config) +static bool _is_codec_supported(struct compress *compress, struct compr_config *config, + const struct snd_compr_caps *caps) { - struct snd_compr_caps caps; bool codec = false; unsigned int i; - if (ioctl(compress->fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { - oops(compress, errno, "cannot get device caps"); - return false; - } - - for (i = 0; i < caps.num_codecs; i++) { - if (caps.codecs[i] == config->codec->id) { + for (i = 0; i < caps->num_codecs; i++) { + if (caps->codecs[i] == config->codec->id) { /* found the codec */ codec = true; break; @@ -146,24 +141,24 @@ static bool _is_codec_supported(struct compress *compress, struct compr_config * return false; } - if (config->fragment_size < caps.min_fragment_size) { + if (config->fragment_size < caps->min_fragment_size) { oops(compress, -EINVAL, "requested fragment size %d is below min supported %d\n", - config->fragment_size, caps.min_fragment_size); + config->fragment_size, caps->min_fragment_size); return false; } - if (config->fragment_size > caps.max_fragment_size) { + if (config->fragment_size > caps->max_fragment_size) { oops(compress, -EINVAL, "requested fragment size %d is above max supported %d\n", - config->fragment_size, caps.max_fragment_size); + config->fragment_size, caps->max_fragment_size); return false; } - if (config->fragments < caps.min_fragments) { + if (config->fragments < caps->min_fragments) { oops(compress, -EINVAL, "requested fragments %d are below min supported %d\n", - config->fragments, caps.min_fragments); + config->fragments, caps->min_fragments); return false; } - if (config->fragments > caps.max_fragments) { + if (config->fragments > caps->max_fragments) { oops(compress, -EINVAL, "requested fragments %d are above max supported %d\n", - config->fragments, caps.max_fragments); + config->fragments, caps->max_fragments); return false; } @@ -213,9 +208,6 @@ struct compress *compress_open(unsigned int card, unsigned int device, if (!compress || !config) return &bad_compress; - /* take a local copy of config */ - compress->config = *config; - snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); compress->max_poll_wait_ms = DEFAULT_MAX_POLL_WAIT_MS; @@ -236,11 +228,27 @@ struct compress *compress_open(unsigned int card, unsigned int device, oops(compress, errno, "cannot open device '%s'", fn); goto input_fail; } + + struct snd_compr_caps caps; + if (ioctl(compress->fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { + oops(compress, errno, "cannot get device caps"); + goto codec_fail; + } + + /* If caller passed "don't care" fill in default values */ + if ((config->fragment_size == 0) || (config->fragments == 0)) { + config->fragment_size = caps.min_fragment_size; + config->fragments = caps.max_fragments; + } + + /* take a local copy of config */ + compress->config = *config; + #if 0 /* FIXME need to turn this On when DSP supports * and treat in no support case */ - if (_is_codec_supported(compress, config) == false) { + if (_is_codec_supported(compress, config, &caps) == false) { oops(compress, errno, "codec not supported\n"); goto codec_fail; } diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 1ab53c6..1309d65 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -58,6 +58,8 @@ extern "C" { #endif /* * struct compr_config: config structure, needs to be filled by app + * If fragment_size or fragments are zero, this means "don't care" + * and tinycompress will choose values that the driver supports * * @fragment_size: size of fragment requested, in bytes * @fragments: number of fragments @@ -78,11 +80,14 @@ struct snd_compr_tstamp; /* * compress_open: open a new compress stream * returns the valid struct compress on success, NULL on failure + * If config does not specify a requested fragment size, on return + * it will be updated with the size and number of fragments that + * were configured * * @card: sound card number * @device: device number * @flags: device flags can be COMPRESS_OUT or COMPRESS_IN - * @config: stream config requested + * @config: stream config requested. Returns actual fragment config */ struct compress *compress_open(unsigned int card, unsigned int device, unsigned int flags, struct compr_config *config); -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 8/14] compress: Handle case of pause() during compress_write()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

If stream is paused() during a compress_write() the poll() will return 0 or -EBADFD, or the write() will return -EBADFD. This is not an error so compress_write() should not pass an error code to the caller. It should abort the write and then return the number of bytes it had written up to that point. diff --git a/compress.c b/compress.c index b706f8b..7457550 100644 --- a/compress.c +++ b/compress.c @@ -329,10 +329,13 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size if (avail.avail < compress->config.fragment_size) { /* nothing to write so wait */ ret = poll(&fds, 1, compress->max_poll_wait_ms); + /* A pause will cause -EBADFD or zero return from driver + * This is not an error, just stop writing + */ + if ((ret == 0) || (ret == -EBADFD)) + break; if (ret < 0) return oops(compress, errno, "poll error"); - if (ret == 0) - return oops(compress, -EPIPE, "Poll timeout, Broken Pipe"); if (fds.revents & POLLOUT) { if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &avail)) return oops(compress, errno, "cannot get avail"); @@ -351,6 +354,9 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size else to_write = size; written = write(compress->fd, cbuf, to_write); + /* If play was paused the write returns -EBADFD */ + if (written == -EBADFD) + break; if (written < 0) return oops(compress, errno, "write failed!\n"); -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 7/14] compress: Remove hardcoded limit on length of poll() wait
by Richard Fitzgerald 22 Feb '13

22 Feb '13

For best power-saving we want to sleep on poll() for as long as possible, we don't want to wake unnecessarily for arbitrary time limits. Adds function compress_set_max_poll_wait() so that client can configure the poll() timeout. diff --git a/compress.c b/compress.c index 5235ea4..b706f8b 100644 --- a/compress.c +++ b/compress.c @@ -77,12 +77,16 @@ #define COMPR_ERR_MAX 32 +/* Default maximum time we will wait in a poll() - 20 seconds */ +#define DEFAULT_MAX_POLL_WAIT_MS 20000 + struct compress { int fd; unsigned int flags; char error[COMPR_ERR_MAX]; struct compr_config config; int running; + int max_poll_wait_ms; }; static int oops(struct compress *compress, int e, const char *fmt, ...) @@ -214,6 +218,8 @@ struct compress *compress_open(unsigned int card, unsigned int device, snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); + compress->max_poll_wait_ms = DEFAULT_MAX_POLL_WAIT_MS; + compress->flags = flags; if (!((flags & COMPRESS_OUT) || (flags & COMPRESS_IN))) { oops(compress, -EINVAL, "can't deduce device direction from given flags\n"); @@ -321,8 +327,8 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size /* we will write only when avail > fragment size */ if (avail.avail < compress->config.fragment_size) { - /* nothing to write so wait for 10secs */ - ret = poll(&fds, 1, 1000000); + /* nothing to write so wait */ + ret = poll(&fds, 1, compress->max_poll_wait_ms); if (ret < 0) return oops(compress, errno, "poll error"); if (ret == 0) @@ -432,3 +438,8 @@ bool is_codec_supported(unsigned int card, unsigned int device, return ret; } +void compress_set_max_poll_wait(struct compress *compress, int milliseconds) +{ + compress->max_poll_wait_ms = milliseconds; +} + diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index e285bd4..1ab53c6 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -189,6 +189,16 @@ int compress_drain(struct compress *compress); bool is_codec_supported(unsigned int card, unsigned int device, unsigned int flags, struct snd_codec *codec); +/* + * compress_set_max_poll_wait: set the maximum time tinycompress + * will wait for driver to signal a poll(). Interval is in + * milliseconds. + * Pass interval of -1 to disable timeout and make poll() wait + * until driver signals. + * If this function is not used the timeout defaults to 20 seconds. + */ +void compress_set_max_poll_wait(struct compress *compress, int milliseconds); + int is_compress_running(struct compress *compress); int is_compress_ready(struct compress *compress); -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 6/14] compress: Fix incorrect error message on compress_resume()
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Was reporting compress_resume() errors as "cannot pause" diff --git a/compress.c b/compress.c index 871c19e..5235ea4 100644 --- a/compress.c +++ b/compress.c @@ -392,7 +392,7 @@ int compress_pause(struct compress *compress) int compress_resume(struct compress *compress) { if (ioctl(compress->fd, SNDRV_COMPRESS_RESUME)) - return oops(compress, errno, "cannot pause the stream\n"); + return oops(compress, errno, "cannot resume the stream\n"); return 0; } -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 5/14] compress: Take local copy of compr_config structure
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Take a copy of the compr_config structure passed by the client. This allows the client to use a structure in temporary memory and avoids the possibility that tinycompress can be broken by the client changing the content of its compr_config. diff --git a/compress.c b/compress.c index c4f215c..871c19e 100644 --- a/compress.c +++ b/compress.c @@ -81,7 +81,7 @@ struct compress { int fd; unsigned int flags; char error[COMPR_ERR_MAX]; - struct compr_config *config; + struct compr_config config; int running; }; @@ -209,7 +209,8 @@ struct compress *compress_open(unsigned int card, unsigned int device, if (!compress || !config) return &bad_compress; - compress->config = config; + /* take a local copy of config */ + compress->config = *config; snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); @@ -319,7 +320,7 @@ int compress_write(struct compress *compress, const void *buf, unsigned int size return oops(compress, errno, "cannot get avail"); /* we will write only when avail > fragment size */ - if (avail.avail < compress->config->fragment_size) { + if (avail.avail < compress->config.fragment_size) { /* nothing to write so wait for 10secs */ ret = poll(&fds, 1, 1000000); if (ret < 0) -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 4/14] compress: Change running from bitfield to plain int
by Richard Fitzgerald 22 Feb '13

22 Feb '13

Declaring running as a single-bit bitfield does not save any space or give any coding advantages, but does leave potential opportunities for the compiler to be inefficient in how it handles bitfields compared to how it would handle a machine int. diff --git a/compress.c b/compress.c index fa2d43f..c4f215c 100644 --- a/compress.c +++ b/compress.c @@ -82,7 +82,7 @@ struct compress { unsigned int flags; char error[COMPR_ERR_MAX]; struct compr_config *config; - int running:1; + int running; }; static int oops(struct compress *compress, int e, const char *fmt, ...) -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 3/14] compress: Remove unused buffer_size member from struct compress
by Richard Fitzgerald 22 Feb '13

22 Feb '13

diff --git a/compress.c b/compress.c index ee11988..fa2d43f 100644 --- a/compress.c +++ b/compress.c @@ -80,7 +80,6 @@ struct compress { int fd; unsigned int flags; - __u64 buffer_size; char error[COMPR_ERR_MAX]; struct compr_config *config; int running:1; -- 1.7.2.5

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[alsa-devel] [PATCH TINYCOMPRESS 2/14] compress: Change compress_write() to take buffer as const void *
by Richard Fitzgerald 22 Feb '13

22 Feb '13

The input buffer to compress_write() should be const, and is better defined as void * so the caller doesn't have to cast. diff --git a/compress.c b/compress.c index 19b9366..ee11988 100644 --- a/compress.c +++ b/compress.c @@ -299,11 +299,12 @@ int compress_get_tstamp(struct compress *compress, return 0; } -int compress_write(struct compress *compress, char *buf, unsigned int size) +int compress_write(struct compress *compress, const void *buf, unsigned int size) { struct snd_compr_avail avail; struct pollfd fds; int to_write, written, total = 0, ret; + const char* cbuf = buf; if (!(compress->flags & COMPRESS_IN)) return oops(compress, -EINVAL, "Invalid flag set"); @@ -343,12 +344,12 @@ int compress_write(struct compress *compress, char *buf, unsigned int size) to_write = avail.avail; else to_write = size; - written = write(compress->fd, buf, to_write); + written = write(compress->fd, cbuf, to_write); if (written < 0) return oops(compress, errno, "write failed!\n"); size -= written; - buf += written; + cbuf += written; total += written; } return total; diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 8a98527..e285bd4 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -125,7 +125,7 @@ int compress_get_tstamp(struct compress *compress, * @buf: pointer to data * @size: number of bytes to be written */ -int compress_write(struct compress *compress, char *buf, unsigned int size); +int compress_write(struct compress *compress, const void *buf, unsigned int size); /* * compress_read: read data from the compress stream -- 1.7.2.5

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[alsa-devel] [PATCH] Fix regression in emu1010 firmware loading
by Mihail Zenkov 22 Feb '13

22 Feb '13

This patch fix regression in emu1010 firmware loading after http://git.alsa-project.org/?p=alsa-kprivate.git;a=commitdiff;h=b209c4dfcd9… I just revert small part of this commit. Tested on emu1212m pci.

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[alsa-devel] Axago ADA-X5
by Pavol Lajciak 22 Feb '13

22 Feb '13

Hello. I have USB connected Axago ADA-X5 http://www.axago.eu/cz/produkty/usb-adaptery/ADA-X5.html Chipset CMedia CM6206. Chipset is incorrectly detected as CM106. I have connected this device to speakers by optical toslik (spdif) connection. I hear in equal interval (period) reapeated jitter, when sound is playing. I use Advanced Linux Sound Architecture Driver Version 1.0.24. and Ubuntu 12.04.2 Here is ID of device - incorrectly detected chipset. Bus 002 Device 004: ID 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device S úctou Pavol Lajčiak

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[alsa-devel] [PATCH] ALSA: hda - hdmi: Make jacks phantom, if they're not detectable
by David Henningsson 22 Feb '13

22 Feb '13

Just as for analog codecs, a jack that isn't suitable for detection (in this case, NO_PRESENCE was set) should be a phantom Jack instead of a normal one. Thanks to Raymond Yau for spotting. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/961286 BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=903869 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 21425fb..78e1827 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1640,6 +1640,9 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) if (pcmdev > 0) sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); + if (!is_jack_detectable(codec, per_pin->pin_nid)) + strncat(hdmi_str, " Phantom", + sizeof(hdmi_str) - strlen(hdmi_str) - 1); return snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str, 0); } -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ALSA: usb: Fix Processing Unit Descriptor parsers
by Takashi Iwai 21 Feb '13

21 Feb '13

At Thu, 21 Feb 2013 01:55:50 +0000, Pawel Moll wrote: > > Commit 99fc86450c439039d2ef88d06b222fd51a779176 "ALSA: usb-mixer: > parse descriptors with structs" introduced a set of useful parsers > for descriptors. Unfortunately the parses for the Processing Unit > Descriptor came with a very subtle bug... > > Functions uac_processing_unit_iProcessing() and > uac_processing_unit_specific() were indexing the baSourceID array > forgetting the fields before the iProcessing and process-specific > descriptors. > > The problem was observed with Sound Blaster Extigy mixer, > where nNrModes in Up/Down-mix Processing Unit Descriptor > was accessed at offset 10 of the descriptor (value 0) > instead of offset 15 (value 7). In result the resulting > control had interesting limit values: > > Simple mixer control 'Channel Routing Mode Select',0 > Capabilities: volume volume-joined penum > Playback channels: Mono > Capture channels: Mono > Limits: 0 - -1 > Mono: -1 [100%] > > Fixed by starting from the bmControls, which was calculated > correctly, instead of baSourceID. > > Now the mentioned control is fine: > > Simple mixer control 'Channel Routing Mode Select',0 > Capabilities: volume volume-joined penum > Playback channels: Mono > Capture channels: Mono > Limits: 0 - 6 > Mono: 0 [0%] > > Signed-off-by: Pawel Moll <mail(a)pawelmoll.com> > --- > include/uapi/linux/usb/audio.h | 6 ++++-- > 1 file changed, 4 insertions(+), 2 deletions(-) > > diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h > index ac90037..d2314be 100644 > --- a/include/uapi/linux/usb/audio.h > +++ b/include/uapi/linux/usb/audio.h > @@ -384,14 +384,16 @@ static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_de > int protocol) > { > __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); > - return desc->baSourceID[desc->bNrInPins + control_size]; > + return *(uac_processing_unit_bmControls(desc, protocol) > + + control_size); > } > > static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc, > int protocol) > { > __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); > - return &desc->baSourceID[desc->bNrInPins + control_size + 1]; > + return uac_processing_unit_bmControls(desc, protocol) > + + control_size + 1; > } Looks good to me. An alternative would be to change uac_processing_unit_bControlSize() to correct the header size, but I don't think it's good, as bControlSize fields in other units show the value including the header size. Daniel, what do you think? thanks, Takashi

2 2

[alsa-devel] [PATCH 1/4] compress: make variable 'running' as unsigned
by Vinod Koul 21 Feb '13

21 Feb '13

as we never expect this to be negative Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index 19d618a..a8964f4 100644 --- a/compress.c +++ b/compress.c @@ -83,7 +83,7 @@ struct compress { __u64 buffer_size; char error[COMPR_ERR_MAX]; struct compr_config *config; - int running:1; + unsigned int running:1; }; static int oops(struct compress *compress, int e, const char *fmt, ...) -- 1.7.0.4

1 4

[alsa-devel] [PATCH RFC 000/112] HD-audio generic parser improvements
by Takashi Iwai 21 Feb '13

21 Feb '13

Hi, this is a large series of patches to implement the really generic HD-audio codec parser usable for various codecs. It's based on the Realtek codec parser, so I started hacking patch_realtek.c for more generalization and improvements. Then it's merged to the existing generic parser code, hda_generic.c, and let Realtek parser just uses this new parser code. In the end, patch_realtek.c contains only the device specific fixups and the call of generic parser functions. The series contains a few improvements in the HD-audio core code, some of which aren't really dependent on the generic parser code. They are just there as I've developed this for a couple of weeks. After this patchset, most of other codecs can be migrated as well. Currently analog, ca0110, cirrus, cmedia, conexant and via codecs have been already rewritten (i.e. the own parser codes dropped). patch_sigmatel.c is still not converted yet, but can be done soon later. I'm going to send the conversion patches for these codecs right after this series. I don't expect to merge all conversions of codec parsers in 3.9 kernel, but at least, the conversion of Realtek and this generic parser code addition can be merged to 3.9. Maybe minor and easy ones like ca0110 and cmedia can be merged as well. Takashi

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[alsa-devel] Discovery of device characteristics
by Tim Hewett 21 Feb '13

21 Feb '13

I'm writing software looking to discover the available sample rates, channel formats etc. for a device. I have been taking the approach of testing a sample rate from a list of potential rates, then setting the device to that rate to then test the channel formats etc.. The structure is initialised using snd_pcm_hw_params_any() then snd_pcm_hw_params_test_rate() is called for a set of rates followed by snd_pcm_hw_params_set_rate() if the previous call is successful, then snd_pcm_hw_params_test_format() for a set of formats followed by snd_pcm_hw_params_set_format() if that is successful, then snd_pcm_hw_params_test_channels(), the point being to narrow down the characteristics available for given sample rates and channel formats. For example I have a device which supports up to 8 channels at sample rates of 44.1kHz and 48kHz but only 2 channels at 96kHz. It works ok usually but keeps failing to detect that the device supports 24-bit channel formats, both for SND_PCM_FORMAT_S24_LE/BE and SND_PCM_FORMAT_S24_3LE/3BE. It detects the 16-bit formats without any problem. Online source taken from http://download.atmark-techno.com/misc/demos/usb-audio-a500-howto/hw_params… does detect the 24-bit format but doesn't set the sample rate first - each characteristic is tested in isolation - snd_pcm_hw_params_any() is called before each one. I am wondering if my way is not the correct way to determine device characteristics.

2 3

Re: [alsa-devel] HD-audio generic parser merged to linux-next (desktop with internal speaker)
by Raymond Yau 21 Feb '13

21 Feb '13

> > > > > > > > hda-emu codecs/alc662-intel-d945gclf2 > > > > > > But it's irrelevant from whether you enable independent HP mode or > > > not. It's just the policy of badness evaluation. It's possible to > > > have a different result by adjusting the badness parameters. > > > > IOW, the patch like below will make surrounds back for the device, > > for the cost of losing individual headphone volume control. > > > > It seems that it would make more sense for having surrounds for such > > desktop setups, and the patch doesn't appear to affect too severely. > > Only ALC662/663 and one CMI9880 configs have been changed by this, and > > all look reasonable. So I'm going to apply it. > > BTW, I should mention beforehand that the previous patch won't suffice > for the device like ThinkCenter A58 where the headphone can't be > shared with the primary DAC. For allowing the surrounds on such > machines, we need to implement more complex workaround (i.e. make > surrounds exclusive, and turn off others while using surrounds). http://www.intel.com/content/www/us/en/motherboards/desktop-motherboards/de… Realtek ALC662* codec for 2+2ch (multi-streaming) Intel® High Definition Audio (Intel® HD Audio): Front panel microphone/headphone header S/PDIF digital audio header Internal mono speaker header https://launchpadlibrarian.net/92117241/Card0.Codecs.codec.0.txt The result of hda-emu seem OK for ThinkCenter M55 (AD1988) which have volume controls for the internal speaker https://bugzilla.redhat.com/attachment.cgi?id=335481 But have same problem with those 6-channel HDA codec M57/M57P(AD1882) https://launchpadlibrarian.net/63326460/Card0.Codecs.codec.2.txt

1 0

[alsa-devel] [PATCH v2 0/8] Add support for directly setting the FLL REFCLK
by Charles Keepax 20 Feb '13

20 Feb '13

This patch series adds support for specifying the source of REFCLK directly for Wolfson Arizona based devices. This should allow for improved performance by using the highest available REFCLK source. Should the REFCLK source not be directly specified then the patches will default to the older behaviour of using the 32kHz clock for REFCLK. REFCLK is configured using a call to snd_soc_codec_set_pll specifying one of the new WM51xx_FLLx_REFCLK defines as the pll_id. It is valid to clear the SYNCCLK source by setting the source to ARIZONA_FLL_SRC_NONE, however once a REFCLK source it can only be replaced with another valid REFCLK source. Charles Keepax (8): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK sound/soc/codecs/arizona.c | 191 ++++++++++++++++++++++++++++++-------------- sound/soc/codecs/arizona.h | 9 ++- sound/soc/codecs/wm5102.c | 6 ++ sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 6 ++ sound/soc/codecs/wm5110.h | 6 +- 6 files changed, 158 insertions(+), 66 deletions(-) -- 1.7.2.5

2 9

[alsa-devel] [PATCH] ASoC: max98088: Add TLV data for volume controls.
by Dylan Reid 20 Feb '13

20 Feb '13

Specify volumes as defined in the MAX98088/9 data sheet. Allows ALSA lib snd_mixer_selem_get_playback_dB_range and related functions to work. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index a4c16fd..3a7b7fd 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = { 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), }; +static const unsigned int max98088_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98088_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + static const struct snd_kcontrol_new max98088_snd_controls[] = { - SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L, - M98088_REG_3A_LVL_HP_R, 0, 31, 0), - SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L, - M98088_REG_3E_LVL_SPK_R, 0, 31, 0), - SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L, - M98088_REG_3C_LVL_REC_R, 0, 31, 0), + SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L, + M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L, + M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv), + SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L, + M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv), SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L, M98088_REG_3A_LVL_HP_R, 7, 1, 1), -- 1.8.0

2 1

[alsa-devel] [PATCH 1/6] ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init
by Stephen Warren 20 Feb '13

20 Feb '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra only supports, and always enables, device tree. Remove all runtime checks for DT support from the driver. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index ba419f8..49861c6 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -176,11 +176,7 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else if (of_machine_is_compatible("nvidia,tegra30")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; - else if (!dev->of_node) - /* non-DT is always Tegra20 */ - data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20; else - /* DT boot, but unknown SoC */ return -EINVAL; data->clk_pll_a = clk_get_sys(NULL, "pll_a"); -- 1.7.10.4

3 7

[alsa-devel] [PATCH 0/8] ASoC: arizona: Add support for directly setting the FLL REFCLK
by Charles Keepax 20 Feb '13

20 Feb '13

This patch series adds support for specifying the source of REFCLK directly for Wolfson Arizona based devices. This should allow for improved performance by using the highest available REFCLK source. Should the REFCLK source not be directly specified then the patches will default to the older behaviour of using the 32kHz clock for REFCLK. REFCLK is configured using a call to snd_soc_codec_set_pll specifying one of the new WM51xx_FLLx_REFCLK defines as the pll_id. It is valid to clear the SYNCCLK source by setting the source to ARIZONA_FLL_SRC_NONE, however once a REFCLK source it can only be replaced with another valid REFCLK source. Charles Keepax (8): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK sound/soc/codecs/arizona.c | 191 ++++++++++++++++++++++++++++++-------------- sound/soc/codecs/arizona.h | 9 ++- sound/soc/codecs/wm5102.c | 6 ++ sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 6 ++ sound/soc/codecs/wm5110.h | 6 +- 6 files changed, 158 insertions(+), 66 deletions(-) -- 1.7.2.5

2 12

[alsa-devel] [PATCH v5 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Just more fixups according to Takashi's comments. David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 52 +++++++++++----------- sound/pci/hda/hda_local.h | 28 +++++++----- sound/pci/hda/patch_hdmi.c | 105 ++++++++++++++++++++++++++++++++++---------- 3 files changed, 127 insertions(+), 58 deletions(-) -- 1.7.9.5

1 5

[alsa-devel] [PATCH v4 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Changes since v3: * snd_hdmi_get_eld now does not take a full hdmi_eld struct David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 52 +++++++++++----------- sound/pci/hda/hda_local.h | 28 +++++++----- sound/pci/hda/patch_hdmi.c | 102 ++++++++++++++++++++++++++++++++++---------- 3 files changed, 124 insertions(+), 58 deletions(-) -- 1.7.9.5

2 8

[alsa-devel] [PATCH v3 0/5] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

Changes since v2: Added refactoring patch to split hdmi_eld into hdmi_eld and parsed_hdmi_eld. David Henningsson (5): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld ALSA: hda - hdmi: Protect ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 47 ++++++++++++++---------- sound/pci/hda/hda_local.h | 23 +++++++----- sound/pci/hda/patch_hdmi.c | 86 +++++++++++++++++++++++++++++++++----------- 3 files changed, 107 insertions(+), 49 deletions(-) -- 1.7.9.5

2 8

[alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by wendelin klimann 19 Feb '13

19 Feb '13

hello I try to implement an PCM3168 audio codec from TI to the BeagleBoard but actually i am stuck and it would be rally nice if one could help me a little bit. My ALSA driver version is 1.0.24 and i use an 3.2.18 kernel. As a startingpoint i used the files */linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3008.c* */linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3008.h* The files (see end of e-mail) */linux-mainline-3.2.18-r121b/git/sound/soc/omap/pcm3168_soundcard.c /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.c /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.h* could be compiled successfully and also the insmod loads the modules *$: insmod /lib/modules/3.2.18/kernel/sound/soc/omap/snd-soc-pcm3168_soundcard.ko* *$: lsmod* Module Size Used by snd_soc_pcm3168 1886 0 snd_soc_pcm3168_soundcard 2094 0 twl4 030_madc_hwmon 2569 0 snd_soc_omap3beagle 2239 0 rtc_twl 4382 0 snd_soc_twl4030 37616 1 rtc_ds1307 6642 0 snd_soc_omap 3521 1 snd_soc_omap_mcbsp 8374 1 snd_soc_core 107346 6 snd_soc_pcm3168,snd_soc_pcm3168_soundcard,snd_soc_omap3beagle,snd_soc_twl4030,snd_soc_omap,snd_soc_omap_mcbsp regmap_spi 1263 1 snd_soc_core snd_pcm 78039 4 snd_soc_twl4030,snd_soc_omap,snd_soc_omap_mcbsp,snd_soc_core snd_timer 18848 1 snd_pcm snd 57930 3 snd_soc_core,snd_pcm,snd_timer soundcore 6954 1 snd iptable_nat 3819 0 snd_page_alloc 4979 1 snd_pcm nf_nat 16515 1 iptable_nat nf_conntrack_ipv4 12018 3 iptable_nat,nf_nat nf_conntrack 71219 3 iptable_nat,nf_nat,nf_conntrack_ipv4 nf_defrag_ipv4 1379 1 nf_conntrack_ipv4 ip_tables 10537 1 iptable_nat x_tables 16505 2 iptable_nat,ip_tables gpio_keys 7117 0 autofs4 20492 2 but the new ALSA device is not recognized *$:aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: omap3beagle [omap3beagle], device 0: TWL4030 twl4030-hifi-0 [] Subdevices : 1/1 Subdevice #0: subdevice #0 So i tried to get some information in *$: cd /sys/kernel/debug/asoc* */sys/kernel/debug/asoc$: ls* codecs dais omap3beagle pcm3168 platforms */sys/kernel/debug/asoc$: cat codecs* pcm3168-codec.0 twl4030-codec */sys/kernel/debug/asoc$: cat dais* pcm3168-hifi twl4030-voice twl4030-hifi omap-mcbsp-dai.4 omap-mcbsp-dai.3 omap-mcbsp-dai.2 omap-mcbsp-dai.1 omap-mcbsp-dai.0 */sys/kernel/debug/asoc$: cat platforms* omap-pcm-audio snd-soc-dummy */sys/kernel/debug/asoc$:cd omap3beagle* */sys/kernel/debug/asoc/omap3beagle$:ls* dapm dapm_pop_time twl4030-codec */sys/kernel/debug/asoc/omap3beagle$:cd /sys/kernel/debug/asoc/pcm3168* */sys/kernel/debug/asoc/pcm3168$: ls* dapm_pop_time At this point i have no idea how to solve the problem or how to debug further and i would be very grateful for any hint. Thanks a lot Wendelin * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/omap/pcm3168_soundcard.c * ** *********************************************************************************************** */* * PCM3168 ASoC driver for BeagleBoard. * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> //#include <asm/dma.h> #include <mach/hardware.h> #include <mach/gpio.h> //#include <mach/edma.h> #include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/pcm3168.h" /* * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ /* * SND_SOC_DAIFMT_CBS_CFS * for codec in slave mode */ #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | \ SND_SOC_DAIFMT_IB_NF) static int pcm3168_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int fs; int ret = 0; printk("in hw_params -> by kli"); /* Fsref can be 8000 to 96000. */ fs = params_rate(params); /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); if (ret < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); return ret; } pr_debug("pcm3168_hw_params: rate = %d Hz\n", fs); return 0; } static struct snd_soc_ops pcm3168_ops = { .hw_params = pcm3168_hw_params, }; /* omap-pcm3168 digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link pcm3168_dai = { .name = "PCM3168", /* Codec name */ .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "pcm3168-hifi", .codec_name = "pcm3168-codec", .platform_name = "omap-pcm-audio", .ops = &pcm3168_ops, }; /* omap-pcm3168 audio machine driver */ static struct snd_soc_card snd_soc_pcm3168 = { .name = "pcm3168", // -> "OMAP pcm3168" .owner = THIS_MODULE, .dai_link = &pcm3168_dai, .num_links = 1, }; /* pcm3168 audio private data */ static struct pcm3168_setup_data pcm3168_pcm3168_setup = { // .dem0_pin = GPIO(45), // .dem1_pin = GPIO(46), // .pdad_pin = GPIO(47), // .pdda_pin = GPIO(38), }; struct platform_device pcm3168_codec = { .name = "pcm3168-codec", .id = 0, .dev = { .platform_data = &pcm3168_pcm3168_setup, }, }; static struct platform_device *pcm3168_snd_device; static int __init pcm3168_init(void) { int ret; printk(KERN_ALERT "in init soundcard \n"); if (!(machine_is_omap3_beagle())) return -ENODEV; platform_device_register(&pcm3168_codec); pcm3168_snd_device = platform_device_alloc("soc-audio", 0); if (!pcm3168_snd_device) { printk(KERN_ERR "platform device allocation failed\n"); return -ENOMEM; } platform_set_drvdata(pcm3168_snd_device, &snd_soc_pcm3168); // platform_device_add_data(pcm3168_snd_device, &pcm3168_snd_data, // sizeof(pcm3168_snd_data)); /* ret = platform_device_add_resources(pcm3168_snd_device, pcm3168_snd_resources, ARRAY_SIZE(pcm3168_snd_resources)); if (ret) { printk(KERN_ERR "platform device add resources failed\n"); goto error; } */ ret = platform_device_add(pcm3168_snd_device); if (ret) goto error; return ret; error: platform_device_put(pcm3168_snd_device); return ret; } static void __exit pcm3168_exit(void) { platform_device_unregister(pcm3168_snd_device); platform_device_unregister(&pcm3168_codec); } module_init(pcm3168_init); module_exit(pcm3168_exit); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("PCM3168 Beagle ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:pcm3168-soc-audio"); * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.c * ** *********************************************************************************************** */* * ALSA Soc PCM3168 codec support * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin * * * Based on AC97 Soc codec, original copyright follow: * Copyright 2005 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * * Generic PCM3168 support. */ #include <linux/init.h> #include <linux/kernel.h> #include <linux/device.h> #include <linux/gpio.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> //#include <sound/soc-dapm.h> #include <sound/soc.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include <plat/mcbsp.h> #include "../omap/omap-mcbsp.h" #include "../omap/omap-pcm.h" #include "pcm3168.h" #define PCM3168_VERSION "0.1" #define PCM3168_RATES SNDRV_PCM_RATE_8000_96000 /*(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000)*/ static struct snd_soc_dai_driver pcm3168_dai = { .name = "pcm3168-hifi", .playback = { .stream_name = "PCM3168 Playback", .channels_min = 2, .channels_max = 8, .rates = PCM3168_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE }, .capture = { .stream_name = "PCM3168 Capture", .channels_min = 2, .channels_max = 6, .rates = PCM3168_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE }, }; static void pcm3168_gpio_free(struct pcm3168_setup_data *setup) { // gpio_free(setup->dem0_pin); // gpio_free(setup->dem1_pin); // gpio_free(setup->pdad_pin); // gpio_free(setup->pdda_pin); } static int pcm3168_soc_probe(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; int ret = 0; printk(KERN_ALERT "in SOC Probe -> kli \n"); printk(KERN_INFO "PCM3168 SoC Audio Codec %s\n", PCM3168_VERSION); return ret; gpio_err: // pcm3168_gpio_free(setup); return ret; } static int pcm3168_soc_remove(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; // pcm3168_gpio_free(setup); return 0; } #define pcm3168_soc_suspend NULL #define pcm3168_soc_resume NULL static struct snd_soc_codec_driver soc_codec_dev_pcm3168 = { .probe = pcm3168_soc_probe, .remove = pcm3168_soc_remove, .suspend = pcm3168_soc_suspend, .resume = pcm3168_soc_resume, }; static int __devinit pcm3168_codec_probe(struct platform_device *pdev) { int ret; printk(KERN_ALERT "probe pcm3168 -> kli \n"); ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3168, &pcm3168_dai, 1); printk(KERN_ALERT "probe_after pcm3168 -> kli = %d \n", ret); return ret; } static int __devexit pcm3168_codec_remove(struct platform_device *pdev) { printk(KERN_ALERT "remove pcm3168 -> kli \n"); snd_soc_unregister_codec(&pdev->dev); return 0; } MODULE_ALIAS("platform:pcm3168-codec"); static struct platform_driver pcm3168_codec_driver = { .probe = pcm3168_codec_probe, .remove = __devexit_p(pcm3168_codec_remove), .driver = { .name = "pcm3168-codec", .owner = THIS_MODULE, }, }; static int __init pcm3168_modinit(void) { printk(KERN_ALERT "in init of 3168 -> kli \n"); return platform_driver_register(&pcm3168_codec_driver); } module_init(pcm3168_modinit); static void __exit pcm3168_exit(void) { printk(KERN_ALERT "in exit of 3168 -> kli \n"); platform_driver_unregister(&pcm3168_codec_driver); } module_exit(pcm3168_exit); MODULE_DESCRIPTION("Soc PCM3168 driver"); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_LICENSE("GPL"); * *********************************************************************************************** * /linux-mainline-3.2.18-r121b/git/sound/soc/codecs/pcm3168.h * *********************************************************************************************** * /* * PCM3168 ALSA SoC Layer * * Author: Hugo Villeneuve * Copyright (C) 2008 Lyrtech inc * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #ifndef __LINUX_SND_SOC_PCM3168_2_H #define __LINUX_SND_SOC_PCM3168_2_H struct pcm3168_setup_data { // unsigned dem0_pin; // unsigned dem1_pin; // unsigned pdad_pin; // unsigned pdda_pin; }; #endif

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[alsa-devel] [PATCH v2 0/4] Make HDMI ELD usable for hotplug
by David Henningsson 19 Feb '13

19 Feb '13

New day, new patches. This time there is a proper lock around access to the ELD buffer. I hope this satisfies the previous comments. I've also tested the patches together with some PulseAudio patches I'm writing and the notification works just fine. The ELD data shows up 300 ms after hotplug, so it's likely the first repoll giving me new ELD data). David Henningsson (4): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: protect access to ELD buffer ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/hda_eld.c | 12 ++++++-- sound/pci/hda/hda_local.h | 3 ++ sound/pci/hda/patch_hdmi.c | 67 +++++++++++++++++++++++++++++++++++--------- 3 files changed, 66 insertions(+), 16 deletions(-) -- 1.7.9.5

2 6

[alsa-devel] [PATCH 0/3] Make HDMI ELD usable with hotplug
by David Henningsson 19 Feb '13

19 Feb '13

I'm experimenting with using ELD information in PulseAudio and found that it didn't work very well with hotplugging. Neither the proc file nor the ELD kcontrol was cleared correctly after unplug, and there was no way (except polling) to find out when ELD has become available. Could use advice on which ones of these that should or should not be sent to stable. David Henningsson (3): ALSA: hda - hdmi: ELD shouldn't be valid after unplug ALSA: hda - hdmi: Do not expose eld data when eld is invalid ALSA: hda - hdmi: Notify userspace when ELD control changes sound/pci/hda/patch_hdmi.c | 35 ++++++++++++++++++++++++++++------- 1 file changed, 28 insertions(+), 7 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Jack: add configurable option for irq_flag
by Omair Mohammed Abdullah 18 Feb '13

18 Feb '13

From: Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> For a jack using GPIO based detection, the framework sets it to trigger on both edges (rising and falling). Some codecs may provide only a falling or rising edge interrupt but not both. This patch provides a configurable option for irq_flag during snd_soc_jack_add_gpios(). If not configured, it sets the interrupt line to trigger on both edges. Signed-off-by: Vaibhav Agarwal <vaibhav.agarwal(a)intel.com> Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)linux.intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-jack.c | 7 +++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..d0286ad 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -585,6 +585,7 @@ struct snd_soc_jack_zone { * @report: value to report when jack detected * @invert: report presence in low state * @debouce_time: debouce time in ms + * @irq_flag: Interrupt flags for GPIO-Irq line * @wake: enable as wake source * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an @@ -597,6 +598,7 @@ struct snd_soc_jack_gpio { int report; int invert; int debounce_time; + unsigned long irq_flags; bool wake; struct snd_soc_jack *jack; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..c9f7953 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -318,10 +318,13 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; + if (!gpios[i].irq_flags) + gpios[i].irq_flags = + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING; + ret = request_any_context_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, - IRQF_TRIGGER_RISING | - IRQF_TRIGGER_FALLING, + gpios[i].irq_flags, gpios[i].name, &gpios[i]); if (ret < 0) -- 1.7.4.1

4 9

[alsa-devel] HDMI regression on close (was: ALSA: hda - Always turn on pins for HDMI/DP)
by Anssi Hannula 18 Feb '13

18 Feb '13

Hi! > From: Takashi Iwai <tiwai(a)suse.de> > Date: Fri, 14 Dec 2012 09:22:35 +0000 (+0100) > Subject: ALSA: hda - Always turn on pins for HDMI/DP > X-Git-Tag: v3.8-rc1~29^2~7 > X-Git-Url: http://git.kernel.org/?p=linux%2Fkernel%2Fgit%2Fstable%2Flinux-stable.git;a… > > ALSA: hda - Always turn on pins for HDMI/DP > > We've seen the broken HDMI *video* output on some machines with GM965, > and the debugging session pointed that the culprit is the disabled > audio output pins. Toggling these pins dynamically on demand caused > flickering of HDMI TV. > > This patch changes the behavior to keep the pin ON constantly. Unfortunately this seems to cause a regression on at least some "NVIDIA ION2" systems. When running snd_pcm_open() and snd_pcm_close() on an hdmi device on an affected system, the HDMI port seemingly continues to transmit an empty audio stream. However, when hw_params have been set and snd_pcm_prepare() called before closing the device (and optionally audio has been output), the HDMI port stops transmitting at snd_pcm_close() time. (I use "transmitting" here to mean that the HDMI sink/receiver receives/shows a an active audio stream) On systems _not_ affected by the issue, the HDMI port never starts transmitting if just snd_pcm_open()+snd_pcm_close() have been called (but note [1]). HDMI port continuing transmitting has the unfortunate effect that when the user has connected the system to an amplifier via HDMI (for video playback purposes), the amplifier may continue to play back empty audio from HDMI even when there is actual audio data on its S/PDIF input. An snd_pcm_open()+snd_pcm_close() sequence can be caused by e.g. device probing by an application. Several OpenELEC users have reported this issue on stable 3.7.x, here's some alsainfo: http://onse.fi/files/ion-regression-alsainfo-1.txt http://onse.fi/files/ion-regression-alsainfo-2.txt Both of these have the Nvidia GPU 0b HDMI/DP codecs: Vendor Id: 0x10de000b Subsystem Id: 0x10de0101 Revision Id: 0x100200 However, e.g. my non-ION2 laptop with 0b codecs (but different rev) is _not_ affected: Vendor Id: 0x10de000b Subsystem Id: 0x10de0101 Revision Id: 0x100100 http://www.alsa-project.org/db/?f=bc940d23d19ae7e2ee1e83895d85f711c2d0e5b1 I'm still working with the OpenELEC devs and users to pinpoint the issue further, but it is progressing rather slowly due to not many people having affected systems (especially in the affected dual HDMI + S/PDIF cabled configuration). Feel free to suggest anything we should try out if you have some ideas :) In the meantime, I'll continue investigating exactly why snd_pcm_close() doesn't stop the HDMI output on the affected systems (or why does a simple snd_pcm_open() even start outputting the audio, since that does not seem to happen on other systems). I'll keep you posted. [1] It seems that on my Intel HDMI system ( http://www.alsa-project.org/db/?f=bb3fc8680b372c0475719d42d7fc8b2bb7bfb4eb ) the HDMI port keeps transmitting in approximately reversed situation than in the above ION2 case. That is, if audio has been output in the port e.g. with a speaker-test command, closing the device will not cause the stream to stop (and therefore the above issue happens again, receives ignores S/PDIF stream as HDMI stream is given higher priority by default). I found out that the attached patch solves that one. What are your thoughts on that? (obviously the patch will not solve the ION2 regression, since cleanup() is not called if prepare() has not been called; or at least I assume so, way too tired to check that now...) Note that closing the device after audio has been output stops HDMI port transmission even without the patch on both the NVIDIA ION2 systems and on my NVIDIA laptop, it is just the Intel HDMI codec where that does not happen (well, of the tested ones, at least). > Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51421 > > Cc: <stable(a)vger.kernel.org> > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > index 0fcfa6f..37dd06d 100644 > --- a/sound/pci/hda/patch_hdmi.c > +++ b/sound/pci/hda/patch_hdmi.c > @@ -431,9 +431,11 @@ static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid) > if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) > snd_hda_codec_write(codec, pin_nid, 0, > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); > - /* Disable pin out until stream is active*/ > + /* Enable pin out: some machines with GM965 gets broken output when > + * the pin is disabled or changed while using with HDMI > + */ > snd_hda_codec_write(codec, pin_nid, 0, > - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); > + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); > } > > static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid) > @@ -1341,7 +1343,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, > struct hdmi_spec *spec = codec->spec; > int pin_idx = hinfo_to_pin_index(spec, hinfo); > hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; > - int pinctl; > bool non_pcm; > > non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); > @@ -1350,11 +1351,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, > > hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); > > - pinctl = snd_hda_codec_read(codec, pin_nid, 0, > - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); > - snd_hda_codec_write(codec, pin_nid, 0, > - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); > - > return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); > } > > @@ -1374,7 +1370,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, > int cvt_idx, pin_idx; > struct hdmi_spec_per_cvt *per_cvt; > struct hdmi_spec_per_pin *per_pin; > - int pinctl; > > if (hinfo->nid) { > cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); > @@ -1391,11 +1386,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, > return -EINVAL; > per_pin = &spec->pins[pin_idx]; > > - pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, > - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); > - snd_hda_codec_write(codec, per_pin->pin_nid, 0, > - AC_VERB_SET_PIN_WIDGET_CONTROL, > - pinctl & ~PIN_OUT); > snd_hda_spdif_ctls_unassign(codec, pin_idx); > per_pin->chmap_set = false; > memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); -- Anssi Hannula

2 8

Re: [alsa-devel] [PATCH V2] ARM: dts: omap3-devkit8000: Enable audio support
by Peter Ujfalusi 18 Feb '13

18 Feb '13

On 02/16/2013 08:33 AM, Anil Kumar wrote: > Add the needed sections to enable audio support on > Devkit8000 when booted with DT blob. > > Signed-off-by: Anil Kumar <anilk4.v(a)gmail.com> Looks good: Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > --- > This patch is based on top of kernel 3.8-rc5 and > the following patches. > > Peter Ujfalusi:- > ASoC: twl4030: Correct the support for Voice port > ASoC: twl4030: Convert MICBIAS to SUPPLY widget > ASoC: omap-twl4030: Add support for routing, voice port and jack detect > > Anil Kumar:- > ARM: dts: add minimal DT support for DevKit8000 > https://patchwork.kernel.org/patch/2122461/ > > -Tested for playback and capture on Devkit8000. > > Test process:- > > #amixer set 'PredriveR Mixer AudioR2' on > #amixer set 'PredriveL Mixer AudioL2' on > #amixer set PreDriv 100 unmute > #amixer set 'DAC2 Digital Fine' 100 > #amixer cset numid=27 1 > #arecord | aplay > > For V2: > - Remove pin mux as done at device boot time by default. > > :100644 100644 6338993... c70116f... M arch/arm/boot/dts/omap3-devkit8000.dts > arch/arm/boot/dts/omap3-devkit8000.dts | 23 +++++++++++++++++++---- > 1 files changed, 19 insertions(+), 4 deletions(-) > > diff --git a/arch/arm/boot/dts/omap3-devkit8000.dts b/arch/arm/boot/dts/omap3-devkit8000.dts > index 6338993..c70116f 100644 > --- a/arch/arm/boot/dts/omap3-devkit8000.dts > +++ b/arch/arm/boot/dts/omap3-devkit8000.dts > @@ -42,6 +42,19 @@ > }; > > }; > + > + sound { > + compatible = "ti,omap-twl4030"; > + ti,model = "devkit8000"; > + > + ti,mcbsp = <&mcbsp2>; > + ti,codec = <&twl_audio>; > + ti,audio-routing = > + "Ext Spk", "PREDRIVEL", > + "Ext Spk", "PREDRIVER", > + "MAINMIC", "Main Mic", > + "Main Mic", "Mic Bias 1"; > + }; > }; > > &i2c1 { > @@ -51,6 +64,12 @@ > reg = <0x48>; > interrupts = <7>; /* SYS_NIRQ cascaded to intc */ > interrupt-parent = <&intc>; > + > + twl_audio: audio { > + compatible = "ti,twl4030-audio"; > + codec { > + }; > + }; > }; > }; > > @@ -86,10 +105,6 @@ > status = "disabled"; > }; > > -&mcbsp2 { > - status = "disabled"; > -}; > - > &mcbsp3 { > status = "disabled"; > }; > -- Péter

1 0

[alsa-devel] [PATCH] ALSA: hda - add quirks for mute LED on two HP machines
by David Henningsson 18 Feb '13

18 Feb '13

These two machines have no mute LED string in BIOS. BugLink: https://bugs.launchpad.net/bugs/1128934 Tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Should be sent to stable too, but I guess the patch have to be rewritten to fit. sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 48c9d10..e575685 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2726,6 +2726,18 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, } } +static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_nid = 0x18; + spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; + spec->gen.vmaster_mute_enum = 1; + } +} + static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2771,6 +2783,7 @@ enum { ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, ALC269_FIXUP_HP_MUTE_LED, + ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -2903,6 +2916,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led, }, + [ALC269_FIXUP_HP_MUTE_LED_MIC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_mute_led_mic1, + }, [ALC269_FIXUP_HP_MUTE_LED_MIC2] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, @@ -2947,6 +2964,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), + SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Remove speaker clicks on CX20549
by David Henningsson 18 Feb '13

18 Feb '13

This chip needs the speaker pin to go to D3 to avoid clicks, so default_power_filter does not work here. This was found on Thinkpad R61i/T61i but I guess it applies to the entire chip. If not, quirks should be set for at least PCI SSID 17aa:20ac. Thanks to c4pp4 for testing. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7d941ef..941bf6c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3350,6 +3350,7 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; + codec->power_filter = NULL; /* Needs speaker amp to D3 to avoid click */ break; case 0x14f15047: codec->pin_amp_workaround = 1; -- 1.7.9.5

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Re: [alsa-devel] 3.7 HDMI channel map regression
by Takashi Iwai 17 Feb '13

17 Feb '13

At Sat, 16 Feb 2013 18:22:25 -0600, Shawn Bohrer wrote: > > On Mon, Jan 28, 2013 at 08:52:05PM -0600, Shawn Bohrer wrote: > > On Mon, Jan 28, 2013 at 09:56:33AM +0100, Takashi Iwai wrote: > > > At Sun, 27 Jan 2013 19:18:27 -0600, > > > Shawn Bohrer wrote: > > > > > > > > Hi Takashi, > > > > > > > > I recently updated my HTPC from 3.6.11 to 3.7.2 and this caused my RL > > > > and FC channels to swap, and my RR and LFE channels to swap for PCM > > > > audio. Doing a git bisect identified > > > > d45e6889ee69456a4d5b1bbb32252f460cd48fa9 "ALSA: hda - Provide the > > > > proper channel mapping for generic HDMI driver" as the commit that > > > > caused my channels to swap. The commit doesn't revert cleanly on > > > > 3.7.4, and I haven't really looked to see what the correct fix might > > > > be. > > > > > > > > Some info that may be relevant, the sound card is a: > > > > > > > > 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset > > > > Family High Definition Audio Controller (rev 04) > > > > > > > > The machine is running Fedora 18 and audio goes over HDMI to a 5.1 > > > > receiver. I'm not really sure what other info you might need, but > > > > let me know if you need something else or have any patches you would > > > > like me to test. > > > > > > OK, it's the first time to get a bug report about this. > > > Could you tell me how did you test it (i.e. which application, which > > > sound backend)? Can you confirm that it's reproduced via speaker-test > > > program in alsa-utils package? > > > > I originally noticed the problem when all of the dialog started coming > > out of my rear left speaker in MythTV after the kernel update. Then I > > started using the Gnome 3 sound configuration gui in the system > > settings which has a speaker test and I assume is using pulseaudio. > > Running 'speaker-test -c6 -l1 -twav' also reproduces the problem. > > > > For reference here are the versions of the various packages that I'm > > running: > > > > alsa-utils-1.0.26-1.fc18.x86_64 > > alsa-firmware-1.0.25-2.fc18.noarch > > alsa-plugins-pulseaudio-1.0.26-2.fc18.x86_64 > > alsa-lib-devel-1.0.26-2.fc18.x86_64 > > alsa-lib-1.0.26-2.fc18.x86_64 > > alsa-tools-firmware-1.0.26.1-1.fc18.x86_64 > > pulseaudio-gdm-hooks-2.1-5.fc18.x86_64 > > pulseaudio-libs-2.1-5.fc18.x86_64 > > pulseaudio-libs-glib2-2.1-5.fc18.x86_64 > > pulseaudio-module-x11-2.1-5.fc18.x86_64 > > pulseaudio-module-bluetooth-2.1-5.fc18.x86_64 > > pulseaudio-2.1-5.fc18.x86_64 > > pulseaudio-utils-2.1-5.fc18.x86_64 > > Hi Takashi, > > Any updates on this issue? I'd really like to see this issue fixed > and am happy to help in any way I can. Until this gets fixed I'm > stuck on a 3.6.* kernel. There is one fix in sound git tree regarding the HDMI channel map, but it's queued for 3.9 kernel (then backported to stable tree). Try sound.git tree or wait for a while until the upstream merge process above is done. Takashi

1 0

[alsa-devel] ASoC updates for v3.9
by Mark Brown 17 Feb '13

17 Feb '13

The following changes since commit 699ba4546cfe3f4eee73aff4f307bf16362f9232: Merge remote-tracking branch 'asoc/topic/wm8993' into asoc-next (2013-02-11 11:06:55 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.9-updates for you to fetch changes up to 54759313ececcbd07e43fad75e0683e39a1f2a25: Merge remote-tracking branch 'asoc/topic/da7213' into asoc-next (2013-02-14 17:28:00 +0000) ---------------------------------------------------------------- ASoC: Final updates for v3.9 A few more updates from the past week - a new driver from Dialog and some small fixes and tweaks. ---------------------------------------------------------------- Adam Thomson (1): ASoC: codecs: Add da7213 codec Fabio Estevam (1): ASoC: fsl: imx-audmux: Fix sparse warning Mark Brown (6): ASoC: wm8994: Fix variable double use ASoC: arizona: Automatically manage input mutes Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branch 'asoc/topic/da7213' into asoc-next Sebastien Guiriec (1): ASoC: omap-mcpdm: Remove useless ressource get. include/sound/da7213.h | 52 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 17 + sound/soc/codecs/da7213.c | 1599 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7213.h | 523 ++++++++++++++ sound/soc/codecs/wm5102.c | 13 - sound/soc/codecs/wm5110.c | 17 - sound/soc/codecs/wm8994.c | 10 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/omap/omap-mcpdm.c | 4 - 11 files changed, 2203 insertions(+), 40 deletions(-) create mode 100644 include/sound/da7213.h create mode 100644 sound/soc/codecs/da7213.c create mode 100644 sound/soc/codecs/da7213.h

2 1

Re: [alsa-devel] [PATCH V6 02/10] ASoC: SAMSUNG: Add DT support for i2s
by Mark Brown 17 Feb '13

17 Feb '13

On Thu, Feb 14, 2013 at 09:33:00PM +0100, Sylwester Nawrocki wrote: > My apologies for the late review. It's already sixth version of this patch > series... But I noticed it just now when applying it for Exynos4. The GPIO > part of the bindings is now incompatible with Exynos4 and the bindings will > need to be changed. Leaving the compatibility code in the I2S driver, when > we could have it done well right from the beginning. If I'm not mistaken > a file similar to arch/arm/boot/dts/exynos4x12-pinctrl.dtsi with at least > entries for the I2S devices is needed and an addition of related pinctrl > nodes in exynos5250.dtsi. But I guess this is not something that could be > fixed in the -rc period ? Yes, and incremental fix would be fine.

2 1

[alsa-devel] Takashi: please help (compiling alsa-driver)
by Jonas Petersen 15 Feb '13

15 Feb '13

Hi Takashi, I'm talking to you directly because this ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-unstable-snapshot.… is the only alsa-driver source that I'm able to compile so far. I can unpack this and simply run ./configure && make && make install. It works great. (I compile against the kernel-headers from ubuntu). I spent hours in trying to compile from several other sources. * The plain1.0.25 release (complains about missing symbols). * The snapshots fromhttp://www.alsa-project.org/snapshot/ (will complain about missing 'uapi' stuff) * Btw. where did the snapshots fromhttp://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/ go (Advertised here:http://www.alsa-project.org/main/index.php/Download)? * Checked out alsa-driver from git. But no luck, tried many things. e.g.: Branch release, then autoconf, but not even ./configure succeeds. Id like to submit some patches and be able to work with the current state from git. How is that supposed to work? Btw. I often read that the alsa-kernel is needed to compile. But it is not neede with this above mentioned 'unstable-snapshot'. It will just compile against the ubuntu kernel-headers (Lubuntu 12.10). How do you generate these snapshots? Thanks and sorry if that's stupid questions.. Jonas

3 6

[alsa-devel] Allocating DMA buffer for non-PCM
by Adrian Knoth 15 Feb '13

15 Feb '13

Hi! Would you say that the following is the proper way to allocate a DMA buffer used to hold level data? struct hdspm { struct pci_dev *pci; /* and an pci info */ struct snd_dma_buffer dmaLevelBuffer; u32 *level_buffer; /* suitably aligned address */ [..] } /* allocate level buffer */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(hdspm->pci), MADIFX_LEVEL_BUFFER_SIZE, &hdspm->dmaLevelBuffer); if (err < 0) { /* error */ [..] } hdspm->level_buffer = snd_sgbuf_get_ptr(&(hdspm->dmaLevelBuffer), 0); memset(hdspm->level_buffer, 0, MADIFX_LEVEL_BUFFER_SIZE); This used to work on my development machine (kernel 3.6.x), but now a kernel 3.2.0 user reports a NULL pointer dereference of hdspm->level_buffer, so apparently, snd_sgbuf_get_ptr() returned NULL for him. How could this possibly happen? Am I missing something? Better use SNDRV_DMA_TYPE_DEV instead? TIA

3 8

Re: [alsa-devel] [PATCH] ASoC: codecs: Add da7213 codec
by Mark Brown 14 Feb '13

14 Feb '13

On Thu, Feb 14, 2013 at 12:02:51PM +0000, Adam Thomson wrote: > This patch adds support for the Dialog DA7213 audio codec. Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA - au88x0 - Define channel map for au88x0
by Raymond Yau 14 Feb '13

14 Feb '13

Define channel map for playback, capture devices of au88x0 Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com>

2 1

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 14 Feb '13

14 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> ---- v6: - update ioctls GET_METADATA to IOWR - add documenetation of enum v5: - update metadata value with 8 words - add reset of new states in set_params - use statically allocated structs v4: - update metadata struct with single key/value v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation ----- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 31 ++++++++- sound/core/compress_offload.c | 96 +++++++++++++++++++++++++ 4 files changed, 180 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..d630163 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,27 @@ struct snd_compr_codec_caps { }; /** + * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the + * end of the track + * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the + * beginning of the track + */ +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +166,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +177,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..cbb592f 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -483,6 +483,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->metadata_set = false; + stream->next_track = false; } else { return -EPERM; } @@ -514,6 +516,49 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->get_metadata(stream, &metadata); + if (retval != 0) + return retval; + + if (copy_to_user((void __user *)arg, &metadata, sizeof(metadata))) + return -EFAULT; + + return 0; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->set_metadata(stream, &metadata); + stream->metadata_set = true; + + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +639,44 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +706,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +733,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 2

Re: [alsa-devel] [PATCH V6 02/10] ASoC: SAMSUNG: Add DT support for i2s
by Sylwester Nawrocki 14 Feb '13

14 Feb '13

On 01/18/2013 12:47 PM, Padmavathi Venna wrote: > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- ... > /* Lock for cross i/f checks */ > @@ -997,19 +1006,76 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) > return i2s; > } > > +#ifdef CONFIG_OF > +static int samsung_i2s_parse_dt_gpio(struct i2s_dai *i2s) > +{ > + struct device *dev = &i2s->pdev->dev; > + int index, gpio, ret; > + > + for (index = 0; index < 7; index++) { > + gpio = of_get_gpio(dev->of_node, index); > + if (!gpio_is_valid(gpio)) { > + dev_err(dev, "invalid gpio[%d]: %d\n", index, gpio); > + goto free_gpio; > + } > + > + ret = gpio_request(gpio, dev_name(dev)); > + if (ret) { > + dev_err(dev, "gpio [%d] request failed\n", gpio); > + goto free_gpio; > + } > + i2s->gpios[index] = gpio; > + } > + return 0; > + > +free_gpio: > + while (--index >= 0) > + gpio_free(i2s->gpios[index]); > + return -EINVAL; > +} > + > +static void samsung_i2s_dt_gpio_free(struct i2s_dai *i2s) > +{ > + unsigned int index; > + for (index = 0; index < 7; index++) > + gpio_free(i2s->gpios[index]); > +} > +#else > +static int samsung_i2s_parse_dt_gpio(struct i2s_dai *dai) > +{ > + return -EINVAL; > +} > + > +static void samsung_i2s_dt_gpio_free(struct i2s_dai *dai) > +{ > +} > + > +#endif NAK. Why we should leave with this temporary code when there is already pinctrl support for Exynos SoCs ? If the pinctrl driver for Exynos5 was updated you could instead just do devm_pinctrl_get_select_default(&i2s->pdev->dev);

2 1

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 14 Feb '13

14 Feb '13

this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v5: - update metadata value with 8 words - add reset of new states in set_params - use statically allocated structs v4: - update metadata struct with single key/value v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation --- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 25 ++++++- sound/core/compress_offload.c | 96 +++++++++++++++++++++++++ 4 files changed, 174 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..5081a66 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -121,6 +121,21 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +}; + /** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP @@ -145,6 +160,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +171,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..cbb592f 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -483,6 +483,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; + stream->metadata_set = false; + stream->next_track = false; } else { return -EPERM; } @@ -514,6 +516,49 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->get_metadata(stream, &metadata); + if (retval != 0) + return retval; + + if (copy_to_user((void __user *)arg, &metadata, sizeof(metadata))) + return -EFAULT; + + return 0; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + if (copy_from_user(&metadata, (void __user *)arg, sizeof(metadata))) + return -EFAULT; + + retval = stream->ops->set_metadata(stream, &metadata); + stream->metadata_set = true; + + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +639,44 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +706,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +733,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 4

Re: [alsa-devel] Use of of_parse_phandle()/of_node_put()
by Peter Ujfalusi 14 Feb '13

14 Feb '13

On 02/13/2013 06:02 PM, Anil Kumar wrote: > ------------8--------------- > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index b7e84a7..9000f4a 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -1044,6 +1044,13 @@ static void soc_remove_dai_links(struct > snd_soc_card *card) > soc_remove_link_dais(card, dai, order); > } > > + /* release cpu_of_node */ > + if(card->dai_link) { > + int i; > + for(i = 0; i < card->num_links; i++) > + of_node_put(card->dai_link[i].cpu_of_node); and the same for codec_of_node and platform_of_node for that matter. Mark: what do you think? Does it make sense to do this in the core or should we let the machine drivers to take care of this? > + } > + > -- Péter

1 0

Re: [alsa-devel] Use of of_parse_phandle()/of_node_put()
by Peter Ujfalusi 14 Feb '13

14 Feb '13

On 02/13/2013 05:40 PM, Anil Kumar wrote: > It looks issue is different here. > I have done some testing here with patch[1]. In this patch i tried to > get same "of_device" node pointer > again and found refcount for this dt node is 1 [Result]. > > As of_parse_phandle() says it "returns the device_node pointer with > refcount incremented". > But why refcount value is 1 [Result] again ? Documentation/kref.txt > > -----------------------------8--------------------- > Patch[1]:- > diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c > index fd98509..0828a5c 100644 > --- a/sound/soc/omap/omap-twl4030.c > +++ b/sound/soc/omap/omap-twl4030.c > @@ -297,6 +297,16 @@ static int omap_twl4030_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "McBSP node is not provided\n"); > return -EINVAL; > } > + printk(KERN_ERR"refcount 0x%x", > atomic_read(&dai_node->kref.refcount)); > + > + dai_node = of_parse_phandle(node, "ti,mcbsp", 0); > + if (!dai_node) { > + dev_err(&pdev->dev, "McBSP node is not provided\n"); > + return -EINVAL; > + } > + printk(KERN_ERR"refcount 0x%x", > atomic_read(&dai_node->kref.refcount)); > + > + > omap_twl4030_dai_links[0].cpu_dai_name = NULL; > omap_twl4030_dai_links[0].cpu_of_node = dai_node; > > [Result]:- > root@DevKit8000:/# insmod snd-soc-omap-twl4030.ko > [ 95.718109] refcount 0x1 > [ 95.720611] refcount 0x1 [ 95.818054] omap-twl4030 sound.20: > twl4030-hifi <-> 49022000.mcbsp mapping ok > > > Thanks, > Anil > [...] > -- Péter

1 0

[alsa-devel] Channel map of alsa-plugin (pulse, jack)
by Raymond Yau 14 Feb '13

14 Feb '13

Does channel map need to be implemented in alsa-plugin (e.g. alsa-pulse and alsa-jack) ?

1 0

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 13 Feb '13

13 Feb '13

this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> ---- v4: - update metadata struct witj single key/value v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation -- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 25 +++++- sound/core/compress_offload.c | 113 +++++++++++++++++++++++++ 4 files changed, 191 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..d7544b5 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -121,6 +121,21 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value; +}; + /** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP @@ -145,6 +160,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +171,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..8d6812e 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,66 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + retval = stream->ops->get_metadata(stream, metadata); + if (retval != 0) + goto out; + + if (copy_to_user((void __user *)arg, metadata, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + +out: + kfree(metadata); + return retval; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + retval = stream->ops->set_metadata(stream, metadata); + stream->metadata_set = true; + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +654,46 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +723,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +750,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 4

Re: [alsa-devel] Use of of_parse_phandle()/of_node_put()
by Peter Ujfalusi 13 Feb '13

13 Feb '13

On 02/13/2013 07:04 AM, Kumar, Anil wrote: > Hi Peter, > > Just trying to understand. > > In omap-twl4030.c file probe function :- > > dai_node = of_parse_phandle(node, "ti,mcbsp", 0); > if (!dai_node) { > dev_err(&pdev->dev, "McBSP node is not provided\n"); > return -EINVAL; > } > > Here “of_parse_phandle()” is used to get “of_device” node pointer. > of_parse_phandle() suggest to use of_node_put() on it when done. > > It looks when code request for an “of_device” node, kernel maintains > “refcount” for this. > It check “refcount” before giving pointer of of_device node and WARN_ON() > in case of refcount > 0 and increase it on success. > > Should this code need to use of_node_put() on the requested “of_device” when done > so that this can be get again ? Hrm, one thing or sure we should not call it of_node_put() while we have the card loaded since the node is used runtime by the core. However when we unload the machine driver it might be needed, but not sure about this. None of the existing machine drivers doing it (tegra, samsung, omap, etc). But if it is needed it might be better to be done by the core? > > I am taking reference of Linux-next kernel. > > I'm sorry if this question is very vague or missing something. > > Thanks, > Anil > > > > > > -- Péter

1 0

[alsa-devel] HD-audio generic parser merged to linux-next
by Takashi Iwai 13 Feb '13

13 Feb '13

Hi, as most of feedbacks about the new generic parser migration were positive, I decided to take a risk: now all commit that have been in test/hda-migrate branch in sound-unstable git tree are merged to for-next branch of sound git tree for better test coverage. It means that all changes are merged to linux-next, i.e. will be included in 3.9 kernel. For the early adapter people, I'm going to keep test/hda-migrate branch of sound-unstable git tree updated for a while, but the branch will be abandoned some time later. If you encounter any regression by the recent update, please let me know. thanks, Takashi

3 14

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 13 Feb '13

13 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v3: - Change back to ioctl struct with padding - adding states for next_track and metadata for proper transistion v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation --- Documentation/sound/alsa/compress_offload.txt | 46 ++++++++++ include/sound/compress_driver.h | 8 ++ include/uapi/sound/compress_offload.h | 23 +++++- sound/core/compress_offload.c | 116 +++++++++++++++++++++++++ 4 files changed, 192 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..ff6c741 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -71,6 +71,8 @@ struct snd_compr_runtime { * @runtime: pointer to runtime structure * @device: device pointer * @direction: stream direction, playback/recording + * @metadata_set: metadata set flag, true when set + * @next_track: has userspace signall next track transistion, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -79,6 +81,8 @@ struct snd_compr_stream { struct snd_compr_runtime *runtime; struct snd_compr *device; enum snd_compr_direction direction; + bool metadata_set; + bool next_track; void *private_data; }; @@ -110,6 +114,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..8903fe4 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,19 @@ struct snd_compr_codec_caps { }; /** + * struct snd_compr_metadata: compressed stream metadata + * @encoder_delay: no of samples inserted by the encoder at the beginning + * of the track + * @encoder_padding: no of samples appended by the encoder at the end + * of the track + */ +struct snd_compr_metadata { + __u32 encoder_delay; + __u32 encoder_padding; + __u32 reserved[12]; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +158,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +169,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..ba2f0ad 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,69 @@ out: return retval; } +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->get_metadata) + return -ENXIO; + + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + retval = stream->ops->get_metadata(stream, metadata); + if (retval != 0) + goto out; + + if (copy_to_user((void __user *)arg, metadata, sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + +out: + kfree(metadata); + return retval; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + pr_debug("metadata encoder delay=%x encoder padding=%x\n", + metadata->encoder_delay, metadata->encoder_padding); + + retval = stream->ops->set_metadata(stream, metadata); + stream->metadata_set = true; + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +657,46 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* you can signal next track isf this is intended to be a gapless stream + * and current track metadata is set + */ + if (stream->metadata_set == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->metadata_set = false; + stream->next_track = true; + return 0; +} + +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + /* stream can be drained only when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + + stream->next_track = false; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +726,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +753,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

3 5

[alsa-devel] [PATCH] ASoC: omap-dmic: Clean up with devm_request_and_ioremap
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up dmic code with devm_request_and_ioremap function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-dmic.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index ba49ccd..77e9e7e 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -493,16 +493,9 @@ static int asoc_dmic_probe(struct platform_device *pdev) goto err_put_clk; } - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(dmic->dev, "memory region already claimed\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); + dmic->io_base = devm_request_and_ioremap(&pdev->dev, res); if (!dmic->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); ret = -ENOMEM; goto err_put_clk; } -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Clean up with devm_* function
by Sebastien Guiriec 13 Feb '13

13 Feb '13

Clean up McPDM driver with devm_ function. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 5ca11bd..079f277 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -369,7 +369,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -389,7 +389,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; @@ -465,14 +464,11 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) if (res == NULL) return -ENOMEM; - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), "McPDM")) - return -EBUSY; - - mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!mcpdm->io_base) + mcpdm->io_base = devm_request_and_ioremap(&pdev->dev, res); + if (!mcpdm->io_base) { + dev_err(&pdev->dev, "cannot remap\n"); return -ENOMEM; + } mcpdm->irq = platform_get_irq(pdev, 0); if (mcpdm->irq < 0) -- 1.7.10.4

3 2

Re: [alsa-devel] [RFC]Connection list of hda widget
by Takashi Iwai 13 Feb '13

13 Feb '13

At Thu, 07 Feb 2013 16:09:35 +0100, David Henningsson wrote: > > On 02/07/2013 06:38 AM, Raymond Yau wrote: > > > > Treat vendor defined widget different from audio selector/audio input > > widget since hda_proc.c query the connection list of vendor defined widget > > > > ADI codecs have vendor defined widgets ( BIAS Power-Down and VREF > > Power-Down) which have connection list > > > > static void print_conn_list(struct snd_info_buffer *buffer, > > struct hda_codec *codec, hda_nid_t nid, > > unsigned int wid_type, hda_nid_t *conn, > > int conn_len) > > { > > int c, curr = -1; > > > > if (conn_len > 1 && > > wid_type != AC_WID_AUD_MIX && > > wid_type != AC_WID_VOL_KNB && > > + wide_type != AC_WID_VENDOR && > > wid_type != AC_WID_POWER) > > curr = snd_hda_codec_read(codec, nid, 0, > > AC_VERB_GET_CONNECT_SEL, 0); > > snd_iprintf(buffer, " Connection: %d\n", conn_len); > > > > I think there are some Sigmatel/IDT (or was it Analog Devices?) codecs > that actually have sensible connections for their vendor specific nodes, > so I would prefer not to change this. Right. The widget connection longer than 32 itself is OK. It's just ugly. That being said, it's basically a bug that we restrict the max connection size in the core code. So I applied the patch below. This removes the limit while we still keep the efficiency (except for proc output, which isn't too critical). The patch became bigger than I wanted originally, but it's life. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Remove limit of widget connections Currently we set the max number of connections to be 32, but there seems codec that gives longer connection lists like AD1988, and we see errors in proc output and else. (Though, in the case of AD1988, it's a list of all codecs connected to a single vendor widget, so this must be something fishy, but it's still valid from the h/w design POV.) This patch tries to remove this restriction. For efficiency, we still use the fixed size array in the parser, but takes a dynamic array when the size is reported to be greater than that. Now the fixed array size is found only in patch_hdmi.c, but it should be fine, as the codec itself can't support so many pins. Reported-by: Raymond Yau <superquad.vortex2(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 62 ++++++++++++++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 4 +-- sound/pci/hda/hda_proc.c | 19 +++++++++++--- sound/pci/hda/patch_hdmi.c | 3 +++ 4 files changed, 60 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f82a64d..3c92514 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -382,13 +382,23 @@ static void remove_conn_list(struct hda_codec *codec) /* read the connection and add to the cache */ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) { - hda_nid_t list[HDA_MAX_CONNECTIONS]; + hda_nid_t list[32]; + hda_nid_t *result = list; int len; len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list)); - if (len < 0) - return len; - return snd_hda_override_conn_list(codec, nid, len, list); + if (len == -ENOSPC) { + len = snd_hda_get_num_raw_conns(codec, nid); + result = kmalloc(sizeof(hda_nid_t) * len, GFP_KERNEL); + if (!result) + return -ENOMEM; + len = snd_hda_get_raw_connections(codec, nid, result, len); + } + if (len >= 0) + len = snd_hda_override_conn_list(codec, nid, len, result); + if (result != list) + kfree(result); + return len; } /** @@ -466,6 +476,27 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_get_connections); +/* return CONNLIST_LEN parameter of the given widget */ +static unsigned int get_num_conns(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!(wcaps & AC_WCAP_CONN_LIST) && + get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); + if (parm == -1) + parm = 0; + return parm; +} + +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid) +{ + return get_num_conns(codec, nid) & AC_CLIST_LENGTH; +} + /** * snd_hda_get_raw_connections - copy connection list without cache * @codec: the HDA codec @@ -483,19 +514,16 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; int i, conn_len, conns; unsigned int shift, num_elems, mask; - unsigned int wcaps; hda_nid_t prev_nid; int null_count = 0; if (snd_BUG_ON(!conn_list || max_conns <= 0)) return -EINVAL; - wcaps = get_wcaps(codec, nid); - if (!(wcaps & AC_WCAP_CONN_LIST) && - get_wcaps_type(wcaps) != AC_WID_VOL_KNB) + parm = get_num_conns(codec, nid); + if (!parm) return 0; - parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); if (parm & AC_CLIST_LONG) { /* long form */ shift = 16; @@ -552,21 +580,13 @@ int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, continue; } for (n = prev_nid + 1; n <= val; n++) { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = n; } } else { - if (conns >= max_conns) { - snd_printk(KERN_ERR "hda_codec: " - "Too many connections %d for NID 0x%x\n", - conns, nid); - return -EINVAL; - } + if (conns >= max_conns) + return -ENOSPC; conn_list[conns++] = val; } prev_nid = val; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c59272..0be1826 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -551,9 +551,6 @@ enum { AC_JACK_PORT_BOTH, }; -/* max. connections to a widget */ -#define HDA_MAX_CONNECTIONS 32 - /* max. codec address */ #define HDA_MAX_CODEC_ADDRESS 0x0f @@ -958,6 +955,7 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) { return snd_hda_get_connections(codec, nid, NULL, 0); } +int snd_hda_get_num_raw_conns(struct hda_codec *codec, hda_nid_t nid); int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 5e02f26..97cdbef 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -623,7 +623,7 @@ static void print_codec_info(struct snd_info_entry *entry, snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); unsigned int wid_type = get_wcaps_type(wid_caps); - hda_nid_t conn[HDA_MAX_CONNECTIONS]; + hda_nid_t *conn = NULL; int conn_len = 0; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, @@ -660,9 +660,18 @@ static void print_codec_info(struct snd_info_entry *entry, if (wid_type == AC_WID_VOL_KNB) wid_caps |= AC_WCAP_CONN_LIST; - if (wid_caps & AC_WCAP_CONN_LIST) - conn_len = snd_hda_get_raw_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); + if (wid_caps & AC_WCAP_CONN_LIST) { + conn_len = snd_hda_get_num_raw_conns(codec, nid); + if (conn_len > 0) { + conn = kmalloc(sizeof(hda_nid_t) * conn_len, + GFP_KERNEL); + if (!conn) + return; + if (snd_hda_get_raw_connections(codec, nid, conn, + conn_len) < 0) + conn_len = 0; + } + } if (wid_caps & AC_WCAP_IN_AMP) { snd_iprintf(buffer, " Amp-In caps: "); @@ -735,6 +744,8 @@ static void print_codec_info(struct snd_info_entry *entry, if (codec->proc_widget_hook) codec->proc_widget_hook(buffer, codec, nid); + + kfree(conn); } snd_hda_power_down(codec); } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 85236da..899c4fb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -64,6 +64,9 @@ struct hdmi_spec_per_cvt { unsigned int maxbps; }; +/* max. connections to a widget */ +#define HDA_MAX_CONNECTIONS 32 + struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; -- 1.8.1.2

2 6

[alsa-devel] Channel map for Logitech Web Camera
by Raymond Yau 13 Feb '13

13 Feb '13

It seem that channel map is still missing for Logitech Web Camera since I can find the corresponding entries of hda-intel and the patched au8830 in asound.state except the Logitech Web Camera state.U0x46d0x992 { control.1 { iface MIXER name 'Mic Capture Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.2 { iface MIXER name 'Mic Capture Volume' value 3041 comment { access 'read write' type INTEGER count 1 range '0 - 3072' dbmin 1800 dbmax 3000 dbvalue.0 2987 } } } Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 2 Streaming bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 38 bInCollection 1 baInterfaceNr( 0) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0000 iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 1 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 1 bControlSize 1 bmaControls( 0) 0x03 Mute Control Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 16000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0044 1x 68 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined

1 0

[alsa-devel] [RFC] Fix for conflict of HDMI and SPDIF IEC958 controls
by Takashi Iwai 12 Feb '13

12 Feb '13

Hi, there is a long-standing problem in HD-audio regarding IEC958 controls. When both an SPDIF and an HDMI are created on the same card (e.g. one from analog codec and one from graphics chip), the driver assigns the IEC958 controls just with new indices, 0, 1, 2... The problem is that there is no way to connect between this index and the actual PCM device. Currently, alsa-lib HDA-Intel.conf has a fixed configuration: spdif -> IEC958 xxx index=0, PCM dev=1 hdmi,0 -> IEC958 xxx index=0, PCM dev=3 hdmi,1 -> IEC958 xxx index=1, PCM dev=7 hdmi,2 -> IEC958 xxx index=2, PCM dev=8 hdmi,3 -> IEC958 xxx index=3, PCM dev=9 So obviously spdif and the first hdmi conflict. Basically this can be fixed by reassigning each IEC958 control with the same device number corresponding to the PCM device. That is, for SPDIF, assign a control element with device=1, for HDMI, device=3,7... However, this obviously breaks the old configuration unless user upgrades the alsa-lib configuration. Thus this is no-go. Now here is a compromise: the IEC958 control for SPDIF is reassigned to device=1 *only* when SPDIF and HDMI coexist. Since the configuration is anyway broken as is now in such a case, it's no big deal to fix one side in an incompatible way. (The reason why SPDIF is re-assigned is that I guess majority of user require more HDMI than SPDIF in such a configuration.) In addition, we need a fix in alsa-lib. Also not for breaking the compatibility with older kernel, we need some fallback check. I did a quick hack to alsa-lib conf code and added "skip_rest" option to the hook element, so that only the first matching element is taken. Long story short, I cooked up three patches. One patch is for kernel, to add the workaround above, and the two are for alsa-lib, one clean up and one to introduce the skip_rest option (and application to HDA-Intel.conf). My plan is to merge this to 3.8 tree, so it's no urgent issue. But it's of course always good to fix something. thanks, Takashi

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Re: [alsa-devel] Xonar DG pci sound card driver incomplete
by Clemens Ladisch 12 Feb '13

12 Feb '13

kernel kernel wrote: > Requesting guidance on how to implement the missing mic input support > for this Asus Xonar card. I've downloaded the relevant datasheets but > am unsure how to proceed. Rip out the card, and look (or eletrically trace) how the CMI8788's GPIOs and the CS4245's inputs/outputs are connected. Test how to configure and set the GPIOs so that the mic works. Also check the register configuration of the CS4245. Regards, Clemens

1 0

Re: [alsa-devel] [PATCH 2/3] ALSA: hda - no-primary-hp is a quirk for model ALC889 not ALC882
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:54:16 +0900, Fernando Luis Vázquez Cao wrote: > > Substitute ALC889 for ALC882 in macro and function names. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> This fixup can work generically to all ALC882 variants, including ALC889. Unless you are really sure that this must be used exclusively only for ALC889, no reason to change the prefix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:52:40.301209823 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:58:54.695063675 +0900 > @@ -5030,7 +5030,7 @@ enum { > ALC889_FIXUP_MBP_VREF, > ALC889_FIXUP_IMAC91_VREF, > ALC882_FIXUP_INV_DMIC, > - ALC882_FIXUP_NO_PRIMARY_HP, > + ALC889_FIXUP_NO_PRIMARY_HP, > }; > > static void alc889_fixup_coef(struct hda_codec *codec, > @@ -5156,7 +5156,7 @@ static void alc889_fixup_imac91_vref(str > * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > -static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > +static void alc889_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > { > struct alc_spec *spec = codec->spec; > @@ -5350,9 +5350,9 @@ static const struct alc_fixup alc882_fix > .type = ALC_FIXUP_FUNC, > .v.func = alc_fixup_inv_dmic_0x12, > }, > - [ALC882_FIXUP_NO_PRIMARY_HP] = { > + [ALC889_FIXUP_NO_PRIMARY_HP] = { > .type = ALC_FIXUP_FUNC, > - .v.func = alc882_fixup_no_primary_hp, > + .v.func = alc889_fixup_no_primary_hp, > }, > }; > > @@ -5388,8 +5388,8 @@ static const struct snd_pci_quirk alc882 > SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), > SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), > SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), > - SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), > - SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC889_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC889_FIXUP_NO_PRIMARY_HP), > > /* All Apple entries are in codec SSIDs */ > SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), > @@ -5432,7 +5432,7 @@ static const struct alc_model_fixup alc8 > {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, > {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, > {.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"}, > - {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > + {.id = ALC889_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, > {} > }; > > >

1 0

Re: [alsa-devel] [PATCH 2/3] ALSA: hda - update documentation for no-primary-hp fixup
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:49:46 +0900, Fernando Luis Vázquez Cao wrote: > > The problem addressed by this fixup is not specific to Vaio Z, affecting > some Vaio all-in-one desktop PCs too. Update the code comments accordingly. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> I applied this now to sound git tree but dropped Cc to stable because the patch is no longer cleanly applicable due to the recent code change. And this is just some minor documentation update, thus we can still live peacefully without this fix. thanks, Takashi > --- > > diff -urNp linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt > --- linux-3.7.6-orig/Documentation/sound/alsa/HD-Audio-Models.txt 2012-12-11 12:30:57.000000000 +0900 > +++ linux-3.7.6/Documentation/sound/alsa/HD-Audio-Models.txt 2013-02-09 22:44:28.614772054 +0900 > @@ -53,7 +53,7 @@ ALC882/883/885/888/889 > acer-aspire-8930g Acer Aspire 8330G/6935G > acer-aspire Acer Aspire others > inv-dmic Inverted internal mic workaround > - no-primary-hp VAIO Z workaround (for fixed speaker DAC) > + no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) > > ALC861/660 > ========== > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-09 22:39:11.745202168 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-09 22:50:10.924470402 +0900 > @@ -5153,7 +5153,8 @@ static void alc889_fixup_imac91_vref(str > } > > /* Don't take HP output as primary > - * strangely, the speaker output doesn't work on VAIO Z through DAC 0x05 > + * Strangely, the speaker output doesn't work on Vaio Z and some Vaio > + * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 > */ > static void alc882_fixup_no_primary_hp(struct hda_codec *codec, > const struct alc_fixup *fix, int action) > >

1 0

Re: [alsa-devel] [PATCH 1/3] ALSA: hda - Workaround for silent output on Sony Vaio VGC-LN51JGB with ALC889
by Takashi Iwai 12 Feb '13

12 Feb '13

At Tue, 12 Feb 2013 16:47:44 +0900, Fernando Luis Vázquez Cao wrote: > > Some Vaio all-in-one desktop PCs (for example VGC-LN51JGB) are affected by > the same issue that caused Vaio Z laptops to become silent: the speaker pin > must be connected to the first DAC even though the codec itself advertises > flexible routing through any of the DACs. > > Use the no-primary-hp fixup for choosing the speaker pin as the primary so > that the right DAC is assigned on this device. > > Cc: stable(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Fernando Luis Vazquez Cao <fernando(a)oss.ntt.co.jp> Thanks, applied now. Takashi > --- > > diff -urNp linux-3.7.6-orig/sound/pci/hda/patch_realtek.c linux-3.7.6/sound/pci/hda/patch_realtek.c > --- linux-3.7.6-orig/sound/pci/hda/patch_realtek.c 2013-02-08 23:27:05.949484427 +0900 > +++ linux-3.7.6/sound/pci/hda/patch_realtek.c 2013-02-08 23:41:04.897644545 +0900 > @@ -5388,6 +5388,7 @@ static const struct snd_pci_quirk alc882 > SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), > SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), > SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), > + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), > > /* All Apple entries are in codec SSIDs */ > SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), > >

1 0

[alsa-devel] [patch] ALSA: hda/ca0132 - fix error handling for dsp_chip_to_dsp_addx()
by Dan Carpenter 12 Feb '13

12 Feb '13

The error handling here won't work on 64 bit systems. INVALID_CHIP_ADDRESS is an unsigned long defined like this: #define INVALID_CHIP_ADDRESS (~0UL) But we truncate the high bits away before returning and then we save it in an unsigned 32bit and then we compare it against the unsigned long version. I've changed everything to unsigned long. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 639a282..42114e9 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1581,7 +1581,7 @@ static int dsp_reset(struct hda_codec *codec) /* * Convert chip address to DSP address */ -static unsigned int dsp_chip_to_dsp_addx(unsigned int chip_addx, +static unsigned long dsp_chip_to_dsp_addx(unsigned int chip_addx, bool *code, bool *yram) { *code = *yram = false; @@ -1596,7 +1596,7 @@ static unsigned int dsp_chip_to_dsp_addx(unsigned int chip_addx, return Y_OFF(chip_addx); } - return (unsigned int)INVALID_CHIP_ADDRESS; + return INVALID_CHIP_ADDRESS; } /* @@ -1620,7 +1620,7 @@ static int dsp_dma_setup_common(struct hda_codec *codec, { int status = 0; unsigned int chnl_prop; - unsigned int dsp_addx; + unsigned long dsp_addx; unsigned int active; bool code, yram; @@ -1712,7 +1712,7 @@ static int dsp_dma_setup_common(struct hda_codec *codec, snd_printdd(KERN_INFO " dsp_dma_setup_common() Write IRQCNT"); snd_printdd( - "ChipA=0x%x,DspA=0x%x,dmaCh=%u, " + "ChipA=0x%x,DspA=0x%lx,dmaCh=%u, " "CHSEL=0x%x,CHPROP=0x%x,Active=0x%x\n", chip_addx, dsp_addx, dma_chan, port_map_mask, chnl_prop, active); @@ -1732,7 +1732,7 @@ static int dsp_dma_setup(struct hda_codec *codec, { int status = 0; bool code, yram; - unsigned int dsp_addx; + unsigned long dsp_addx; unsigned int addr_field; unsigned int incr_field; unsigned int base_cnt;

2 1

[alsa-devel] [PATCH] alsa-lib: Add sys/types.h to include list
by David Henningsson 12 Feb '13

12 Feb '13

This is needed by snd_pcm_format_silence* functions which return u_int*_t. It was discovered while trying to compile ALSA programs with eglibc 2.17. Credits to Richard Shaw, Gary Buhrmaster, Matthieu Baerts and Adam Conrad for this fix. BugLink: https://bugs.launchpad.net/bugs/1109298 BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=885306 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/asoundlib-head.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/asoundlib-head.h b/include/asoundlib-head.h index 20c8a68..6edbab0 100644 --- a/include/asoundlib-head.h +++ b/include/asoundlib-head.h @@ -31,6 +31,7 @@ #include <unistd.h> #include <stdio.h> #include <stdlib.h> +#include <sys/types.h> #include <string.h> #include <fcntl.h> #include <assert.h> -- 1.7.9.5

3 4

Re: [alsa-devel] Sound record on Devkit8000 with DT boot issue
by Peter Ujfalusi 11 Feb '13

11 Feb '13

Hi, On 02/11/2013 06:58 PM, Anil Kumar wrote: >> Here you should have something like: >> ti,audio-routing = >> "Ext Spk", "PREDRIVEL", >> "Ext Spk", "PREDRIVER", >> "MAINMIC", "Main Mic", >> "Main Mic", "Mic Bias 1"; >> >> So you connect the MicBias1 to Main Mic. When you start playback MicBias1 will >> be turned on for you. > > I tried with this routing (next-20130211 tag kernel). But it is not > working for me. > So i have gone through other board and found snd_soc_dapm_route for > > 1) zoom2.c > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Ext Mic"}, > > 2) omap3pandora.c > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Mic (internal)"}, > > But for omap-twl4030.c it is > {"MAINMIC", NULL, "Main Mic"}, > {"Main Mic", NULL, "Mic Bias 1"}, > > It Should will be as ? > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Main Mic"}, > > Because with this i am able to get audio on Devkit8000 as > ti,audio-routing = > "Ext Spk", "PREDRIVEL", > "Ext Spk", "PREDRIVER", > "MAINMIC", "Mic Bias 1", > "Mic Bias 1", "Main Mic"; In 3.9 this should not work... linux-next should not have zoom2 at least and I converted the machine drivers as well in regards to micbias handling: e04d6e5 ASoC: twl4030: Convert MICBIAS to SUPPLY widget 57d61b9 ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 -- Péter

1 0

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Remove useless ressource get.
by Sebastien Guiriec 11 Feb '13

11 Feb '13

Remove unused memory ressource get from McPDM driver. Signed-off-by: Sebastien Guiriec <s-guiriec(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 2fe8be2..5ca11bd 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -449,10 +449,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) omap_mcpdm_dai_dma_params[0].port_addr = res->start + MCPDM_REG_DN_DATA; omap_mcpdm_dai_dma_params[1].port_addr = res->start + MCPDM_REG_UP_DATA; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) - return -ENOMEM; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); if (!res) return -ENODEV; -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: fsl: imx-audmux: Fix sparse warning
by Fabio Estevam 11 Feb '13

11 Feb '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Fix the following sparse warning: sound/soc/fsl/imx-audmux.c:182:3: warning: symbol 'audmux_type' was not declared. Should it be static? Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d9..fab912e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -176,7 +176,7 @@ static inline void audmux_debugfs_remove(void) } #endif -enum imx_audmux_type { +static enum imx_audmux_type { IMX21_AUDMUX, IMX31_AUDMUX, } audmux_type; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] compress: add support for gapless playback
by Vinod Koul 11 Feb '13

11 Feb '13

this adds new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v2: - Make it a patch, not RFC - split metadata to key/value pairs, and send multiple keys - add get_metadata api - split partial_drain to next_data & partial_drain - add stream states for transistion - update documentation --- Documentation/sound/alsa/compress_offload.txt | 46 +++++++++ include/sound/compress_driver.h | 4 + include/uapi/sound/asound.h | 4 +- include/uapi/sound/compress_offload.h | 36 +++++++- sound/core/compress_offload.c | 129 ++++++++++++++++++++++++- 5 files changed, 215 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..0bcc551 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,52 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +Also, there might be low-intensity noises due to encoding. Perfect gapless is +difficult to reach with all types of compressed data, but works fine with most +music content. The decoder needs to know the encoder delay and encoder padding. +So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers +and are not present by default in the bitstream, hence the need for a new +interface to pass this information to the DSP. Also DSP and userspace needs to +switch from one track to another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence. This needs to be set before the data in the track +is written. + +- set_next_track +This routine tells DSP that metadata and write operation sent after this would +correspond to subsequent track + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + +Sequence flow for gapless would be: +- Open +- Get caps / codec caps +- Set params +- Set metadata of the first track +- Fill data of the first track +- Trigger start +- User-space finished sending all, +- Indicaite next track data by sending set_next_track +- Set metadata of the next track +- then call partial_drain to flush most of buffer in DSP +- Fill data of the next track +- DSP switches to second track +(note: order for partial_drain and write for next track can be reversed as well) + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..95e955f 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -110,6 +110,10 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); + int (*get_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c..88a6770 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -271,7 +271,9 @@ typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_PAUSED ((__force snd_pcm_state_t) 6) /* stream is paused */ #define SNDRV_PCM_STATE_SUSPENDED ((__force snd_pcm_state_t) 7) /* hardware is suspended */ #define SNDRV_PCM_STATE_DISCONNECTED ((__force snd_pcm_state_t) 8) /* hardware is disconnected */ -#define SNDRV_PCM_STATE_LAST SNDRV_PCM_STATE_DISCONNECTED +#define SNDRV_PCM_STATE_NEXT_TRACK ((__force snd_pcm_state_t) 9) /* stream will move to next track */ +#define SNDRV_PCM_STATE_PARTIAL_DRAIN ((__force snd_pcm_state_t) 10) /* stream is draining partially */ +#define SNDRV_PCM_STATE_LAST SNDRV_PCM_STATE_PARTIAL_DRAIN enum { SNDRV_PCM_MMAP_OFFSET_DATA = 0x00000000, diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..1d63993 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -121,6 +121,32 @@ struct snd_compr_codec_caps { struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; }; +enum { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_keyvalue: compressed stream key/value pairs + * @key: key id + * @value: key value + */ + +struct snd_compr_keyvalue { + __u32 key; + __u32 value; +}; + +/** + * struct snd_compr_metadata: compressed stream metadata + * @count: number of keys + * @keys: pointer to count keys + */ +struct snd_compr_metadata { + __u32 count; + struct snd_compr_keyvalue *keys; +}; + /** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP @@ -145,6 +171,10 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOW('C', 0x15,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +182,14 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..4928209 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -262,9 +262,14 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* write is allowed when stream is running or has been steup */ + /* write is allowed when stream is running or has been steup + * also stream cna be written when next_track info has been setup + * or its has partially drained + */ if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && - stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + stream->runtime->state != SNDRV_PCM_STATE_RUNNING && + stream->runtime->state != SNDRV_PCM_STATE_NEXT_TRACK && + stream->runtime->state != SNDRV_PCM_STATE_PARTIAL_DRAIN) { mutex_unlock(&stream->device->lock); return -EBADFD; } @@ -288,6 +293,9 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream->runtime->state = SNDRV_PCM_STATE_PREPARED; pr_debug("stream prepared, Houston we are good to go\n"); } + if (stream->runtime->state == SNDRV_PCM_STATE_NEXT_TRACK || + stream->runtime->state == SNDRV_PCM_STATE_PARTIAL_DRAIN) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; mutex_unlock(&stream->device->lock); return retval; @@ -514,6 +522,78 @@ out: return retval; } +static int +snd_compr_copy_metadata(struct snd_compr_metadata **arg, unsigned long user) +{ + struct snd_compr_metadata _mdata, *mdata; + int len; + + if (copy_from_user(&_mdata, (void __user *)user, + sizeof(_mdata))) + return -EFAULT; + + len = sizeof(_mdata.count) + _mdata.count * sizeof(*_mdata.keys); + + mdata = kmalloc(len, GFP_KERNEL); + if (!mdata) + return -ENOMEM; + + if (copy_from_user(mdata, (void __user *)arg, len)) { + kfree(mdata); + return -EFAULT; + } + *arg = mdata; + return len; +} + +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *mdata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + retval = snd_compr_copy_metadata(&mdata, arg); + if (retval <= 0) + return retval; + + retval = stream->ops->set_metadata(stream, mdata); + + kfree(mdata); + return retval; +} + +static int +snd_compr_get_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *mdata; + int retval, len; + + if (!stream->ops->get_metadata) + return -ENXIO; + + len = snd_compr_copy_metadata(&mdata, arg); + if (len <= 0) + return len; + retval = stream->ops->get_metadata(stream, mdata); + if (retval != 0) + goto out; + + if (copy_to_user((void __user *)arg, mdata, len)) { + retval = -EFAULT; + goto out; + } + +out: + kfree(mdata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +674,38 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + + /* we can partially drain streams, only when next track info has been + * passed to the dsp + */ + if (stream->runtime->state != SNDRV_PCM_STATE_NEXT_TRACK) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval != 0) + return retval; + stream->runtime->state = SNDRV_PCM_STATE_PARTIAL_DRAIN; + return 0; +} + +static int snd_compr_next_track(struct snd_compr_stream *stream) +{ + int retval; + + /* only a running stream can transition to next track */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_NEXT_TRACK); + if (retval != 0) + return retval; + stream->runtime->state = SNDRV_PCM_STATE_NEXT_TRACK; + return 0; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +735,12 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; + case _IOC_NR(SNDRV_COMPRESS_GET_METADATA): + retval = snd_compr_get_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +762,13 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; + case _IOC_NR(SNDRV_COMPRESS_NEXT_TRACK): + retval = snd_compr_next_track(stream); + break; + } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

2 5

[alsa-devel] [PATCH] ALSA: core: don't return uninitialized snd_compr_tstamp
by Mark Brown 11 Feb '13

11 Feb '13

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> The snd_compr_update_tstamp() can only fill in the snd_compr_tstamp if the codec implements the pointer() function. If that happened the code was previously returning uninitialized garbage in the tstamp because it wasn't initialized anywhere. This change zero-fills the tstamp in the two places it is used before calling snd_compr_update_tstamp(), and also has snd_compr_update_tstamp() return an error indication if it can't provide a tstamp. For the case of snd_compr_calc_avail() it ignores this error because we still need to return info on the available buffer space even if we can't provide tstamp info - when the tstamp is not valid all fields are now guaranteed to be zero. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2d62068 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -144,16 +144,17 @@ static int snd_compr_free(struct inode *inode, struct file *f) return 0; } -static void snd_compr_update_tstamp(struct snd_compr_stream *stream, +static int snd_compr_update_tstamp(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { if (!stream->ops->pointer) - return; + return -ENOTSUPP; stream->ops->pointer(stream, tstamp); pr_debug("dsp consumed till %d total %d bytes\n", tstamp->byte_offset, tstamp->copied_total); stream->runtime->hw_pointer = tstamp->byte_offset; stream->runtime->total_bytes_transferred = tstamp->copied_total; + return 0; } static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, @@ -161,7 +162,9 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, { long avail_calc; /*this needs to be signed variable */ + memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); + /* Still need to return avail even if tstamp can't be filled in */ /* FIXME: This needs to be different for capture stream, available is # of compressed data, for playback it's @@ -517,11 +520,14 @@ out: static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { - struct snd_compr_tstamp tstamp; + struct snd_compr_tstamp tstamp = {0}; + int ret; - snd_compr_update_tstamp(stream, &tstamp); - return copy_to_user((struct snd_compr_tstamp __user *)arg, - &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + ret = snd_compr_update_tstamp(stream, &tstamp); + if (ret == 0) + ret = copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + return ret; } static int snd_compr_pause(struct snd_compr_stream *stream) -- 1.7.10.4

2 1

[alsa-devel] ASoC updates for v3.9
by Mark Brown 11 Feb '13

11 Feb '13

The following changes since commit 836dc9e3fbbab0c30aa6e664417225f5c1fb1c39: Linux 3.8-rc7 (2013-02-09 08:20:39 +1100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.9 for you to fetch changes up to 699ba4546cfe3f4eee73aff4f307bf16362f9232: Merge remote-tracking branch 'asoc/topic/wm8993' into asoc-next (2013-02-11 11:06:55 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.9 A fairly quiet release for ASoC: - Support for a wider range of hardware in the compressed stream code. - The ability to mute capture streams as well as playback streams while inactive. - DT support for AK4642, FSI, Samsung I2S and WM8962. - AC'97 support for Tegra. - New driver for max98090, replacing the stub which was there. Due to dependencies we've also got support for asynchronous I/O in regmap and DTification of DMA support for Samsung platforms (used only by the I2S driver and SPI) merged here as well. ---------------------------------------------------------------- Alexey Galakhov (1): ASoC: samsung: Fix compilation error on S3C2440 Andrey Smirnov (2): regmap: Add provisions to have user-defined read operation regmap: Add provisions to have user-defined write operation Axel Lin (2): ASoC: cs42l52: Convert to devm_input_allocate_device() regmap: Export regmap_async_complete_cb Benoît Thébaudeau (1): ASoC: tlv320aic3x: Remove mono support Bo Shen (4): ASoC: atmel_pcm: make it buildable as module ASoC: atmel_ssc_dai: remove error set private data ASoC: atmel_ssc_dai: correct sequence when unload ASoC: sam9g20_wm8731: disable clock and correct sequence when unload Charles Keepax (5): ASoC: wm_adsp: Release firmware on error ASoC: soc-compress: Add missing brackets around else ASoC: soc-compress: Serialise compressed ops ASoC: soc-compress: Initialise delayed work to power down audio ASoC: soc-compress: Add support for not memory mapped DSPs Chris Rattray (6): ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock ASoC: wm2200: Set system clock control register is adsp structs ASoC: wm2200: Add Rx ANC input select control ASoC: wm2200: correct IN2L and IN3L digital mute ASoC: wm2200: Provide platform data for MICBIAS configuration ASoC: wm_adsp: round to 4-byte boundary for coeff file blocks Chuansheng Liu (1): ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off Dan Carpenter (2): ASoC: pcm: delete some dead code ASoC: dwc: fix support for more than two channels Daniel Mack (1): ALSA: ASoC: cs4271: add optional soft reset workaround Fabio Estevam (6): ASoC: wm8962: Add device tree support ASoC: wm8804: Remove redundant check ASoC: wm8993: Refactor set_pll code to avoid GCC warnings ASoC: wm8985: Refactor set_pll code to avoid gcc warnings ASoC: mxs-saif: Remove platform data ASoC: mxs-saif: Use a signed integer for error value Hebbar Gururaja (1): ASoC: tlv320aic3x: Convert mic bias to a supply widget Jerry Wong (2): ASoC: Replace max98090 Device Driver ASoC: Replace max98090 Device Driver Joachim Eastwood (2): ASoC: atmel-ssc: make it buildable on other architectures ASoC: atmel-soc: make it buildable on other architectures Kuninori Morimoto (12): ASoC: fsi: remove platform depended .set_rate() callback support ASoC: fsi: remove SH_FSI_xxx_INV flags ASoC: fsi: cleanup sh_fsi.h ASoC: simple-card: use struct device pointer for dev_xxx() ASoC: fsi: don't use platform info pointer on probe() ASoC: ak4642: add Device Tree support ASoC: simple-card: add asoc_simple_dai for initializing ASoC: fsi: add device tree support ASoC: SND_SOC_DAIFMT_NB_NF become 0 as default settings ASoC: add snd_soc_of_parse_daifmt() for DeviceTree ASoC: SND_SOC_DAIFMT_GATED become 0 as default settings ASoC: clock gating is decided by bool on snd_soc_of_parse_daifmt() Lars-Peter Clausen (1): ASoC: Constify ops and compr_ops fields of snd_soc_dai_link Lee Jones (1): ASoC: Ux500: Fix build error pertaining to missing include file Lucas Stach (4): ASoC: tegra: add function to set ac97 rate ASoC: tegra: setup DAP3<->DAC3 connection by default ASoC: tegra: add ac97 host driver ASoC: tegra: add tegra machine driver using wm9712 codec Mark Brown (97): ASoC: wm8962: Convert to devm_input_allocate_device() ASoC: wm8962: Unconditionally wait for the FLL to lock ASoC: wm5102: Split input PGA controls ASoC: wm5110: Split input PGA controls ASoC: arizona: Add noise gate hold time enumeration ASoC: wm5102: Add noise gate control ASoC: wm5110: Add noise gate control ASoC: arizona: Implement tristate support ASoC: wm2000: Integrate with clock API ASoC: wm2000: Use clock API integration to configure MCLK divisor Merge branch 'asoc-fix-cs4271' into asoc-cs4271 ASoC: wm5102: Implement routing and power management for ISRCs ASoC: arizona: Allow runtime reconfiguration of the output mode Merge branch 'fix/arizona' of git://git.kernel.org/.../broonie/sound into asoc-arizona ASoC: arizona: Factor out rate selection code Merge remote-tracking branch 'asoc/fix/pxa' into asoc-pxa ASoC: wm_adsp: Add support for parsing algorithms ASoC: wm_adsp: Factor out calculation of memory base addresses ASoC: wm_asdp: Validate sanity of algorithm count ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks ASoC: wm_adsp: Add basic firmware selection support Merge remote-tracking branch 'asoc/topic/adsp' into asoc-arizona Merge remote-tracking branch 'asoc/fix/arizona' into asoc-arizona Merge branch 'topic/fsi' of git://git.kernel.org/.../broonie/sound into asoc-simple-card ASoC: wm_adsp: Add initialisation function for ADSP1 Merge remote-tracking branch 'asoc/topic/adsp' into asoc-wm2200 ASoC: wm2200: Initialise the ADSPs ASoC: arizona: Don't request FLL lock IRQ Merge remote-tracking branch 'asoc/fix/arizona' into asoc-arizona ASoC: wm_adsp: Implement support for coefficeint file format 1 ASoC: wm2200: Add controls for firmware enumeration ASoC: wm5102: Add controls for firmware selection Merge remote-tracking branch 'asoc/topic/adsp' into asoc-wm2200 ASoC: wm5100: Implement DRC, EQ and LHPF coefficient configuration ASoC: wm2200: Implement EQ and LHPF coefficient configuration ASoC: wm_adsp: Make region identification errors more informative ASoC: wm_adsp: Use GFP_DMA for algorithm readback ASoC: arizona: Support clearing clocks Merge remote-tracking branch 'asoc/fix/adsp' into asoc-adsp ASoC: wm_adsp: Provide explicit trace of coefficient writes ASoC: arizona: Allow number of channels clocked to be restricted ASoC: core: Ensure SND_SOC_BYTES writes are from DMA safe memory ASoC: wm_adsp: Correct handling of some coefficeint blocks ASoC: wm2000: Expose some additional registers ASoC: wm_adsp: Add speaker Tx as a firmware option ASoC: wm_adsp: Accept 0 as a parameter block address ASoC: wm_adsp: Ensure ADSP2 DMAs are quiesced when DSP is halted Merge remote-tracking branch 'asoc/fix/adsp' into asoc-adsp regmap: Add asynchronous I/O support regmap: spi: Support asynchronous I/O for SPI ASoC: wm_adsp: Fix support for firmware switching on DSP2 and higher ASoC: wm2000: Lock state machine updates ASoC: wm2000: Expose ANC gain adjustment Merge tag 'async' of git://git.kernel.org/.../broonie/regmap into asoc-adsp ASoC: wm_adsp: Use asynchronous I/O to write firmware and coefficients regmap: Export regmap_async_complete() Merge tag 'async' of git://git.kernel.org/.../broonie/regmap into asoc-adsp ASoC: arizona: Fix debug logging level for FLLs and AIFs ASoC: compress: Only mute playback streams ASoC: core: Allow digital mute for capture Merge remote-tracking branch 'asoc/fix/adsp' into asoc-next Merge remote-tracking branch 'asoc/fix/arizona' into asoc-next Merge remote-tracking branch 'asoc/fix/davinci' into asoc-next Merge remote-tracking branch 'asoc/fix/dwc' into asoc-next Merge remote-tracking branch 'asoc/fix/samsung' into asoc-next Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/fix/ux500' into asoc-next Merge remote-tracking branch 'asoc/fix/wm2000' into asoc-next Merge remote-tracking branch 'asoc/fix/wm2200' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l52' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/of' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/simple-card' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320dac33' into asoc-next Merge remote-tracking branch 'asoc/topic/twl' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2200' into asoc-next Merge remote-tracking branch 'asoc/topic/wm5100' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8350' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8804' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8985' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8993' into asoc-next Michal Bachraty (1): ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback Mike Dunn (3): ASoC: palm27x: fix widgets and routes in dai_link init ASoC: palm27x: register card in platform_driver probe ALSA: pxa27x: rename pxa27x_assert_ac97reset() Misael Lopez Cruz (1): ASoC: twl6040: Convert PLUGINT to no-suspend irq Padmavathi Venna (10): ASoC: samsung: Rename samsung i2s secondary device name ASoC: Samsung: Add I2S S/W RST in startup function ASoC: samsung: Add DT support for i2s ARM: SAMSUNG: Make dma request compatible to generic dma bindings. spi: s3c64xx: Modify SPI driver to use generic DMA DT support ARM: dts: Add nodes for i2s controllers for Samsung Exynos5 platforms ARM: EXYNOS: Enable platform support for I2S controllers ASoC: SMDK: WM8994: Add device tree support for machine file ARM: dts: Modify SPI nodes according generic DMA DT bindings ASoC: Samsung: Update Kconfig for I2S,SPDIF and PCM audio Peter Ujfalusi (15): ASoC: twl4030: Correct the support for Voice port ASoC: zoom2: No need to configure the Voice port anymore ASoC: sdp3430: No need to configure the Voice port anymore ASoC: twl4030: Convert MICBIAS to SUPPLY widget ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use ASoC: sdp3430: No need to configure pin mux for extmute ASoC: omap-twl4030: Add support for routing, voice port and jack detect ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430 ASoC: twl4030: Remove suspend/resume soc driver operations ASoC: twl6040: Only set the bias_level once in twl6040_resume() ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation ASoC: tlv320dac33: Remove suspend/resume soc driver operations ASoC: omap-pcm: No need to set constraint at open time R. Chandrasekar (1): ASoC: Samsung: I2S: Add support for runtime S2R Ricardo Neri (1): ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word Ryo Tsutsui (1): ASoC: arizona: Fixed a bug in FLL fractional calculation Sachin Kamat (4): ASoC: tegra_wm9712: Remove __devinitconst attribute ASoC: tegra20_ac97: Remove __devinitconst attribute ASoC: tegra: Use NULL instead of 0 for pointers ASoC: tegra: Staticize some functions in tegra30_i2s.c Stephen Rothwell (1): ASoC: ak4642: remove __devinitconst annotation Stephen Warren (1): regmap: include linux/sched.h to fix build Tejun Heo (1): ASoC: wm8350: don't use [delayed_]work_pending() Thierry Reding (1): ASoC: fsi: Remove __devinitconst Documentation/devicetree/bindings/sound/ak4642.txt | 17 + Documentation/devicetree/bindings/sound/cs4271.txt | 12 + .../bindings/sound/nvidia,tegra-audio-wm9712.txt | 51 + .../bindings/sound/nvidia,tegra20-ac97.txt | 22 + .../devicetree/bindings/sound/omap-twl4030.txt | 46 + .../devicetree/bindings/sound/renesas,fsi.txt | 26 + .../bindings/sound/samsung,smdk-wm8994.txt | 14 + .../devicetree/bindings/sound/samsung-i2s.txt | 63 + .../devicetree/bindings/sound/tlv320aic3x.txt | 6 + Documentation/devicetree/bindings/sound/wm8962.txt | 16 + arch/arm/boot/dts/exynos5250-smdk5250.dts | 26 + arch/arm/boot/dts/exynos5250.dtsi | 44 +- arch/arm/mach-exynos/mach-exynos5-dt.c | 6 + arch/arm/mach-pxa/pxa27x.c | 20 +- arch/arm/mach-shmobile/board-ap4evb.c | 42 +- arch/arm/mach-shmobile/board-armadillo800eva.c | 34 +- arch/arm/mach-shmobile/board-kzm9g.c | 20 +- arch/arm/mach-shmobile/board-mackerel.c | 42 +- arch/arm/plat-samsung/dma-ops.c | 10 +- arch/arm/plat-samsung/include/plat/dma-ops.h | 3 +- arch/arm/plat-samsung/s3c-dma-ops.c | 3 +- arch/sh/boards/mach-ecovec24/setup.c | 27 +- arch/sh/boards/mach-se/7724/setup.c | 29 +- drivers/base/regmap/internal.h | 18 + drivers/base/regmap/regmap-spi.c | 52 + drivers/base/regmap/regmap.c | 301 ++- drivers/misc/Kconfig | 2 +- drivers/misc/atmel-ssc.c | 2 +- drivers/spi/spi-s3c64xx.c | 77 +- include/linux/mfd/arizona/pdata.h | 9 + include/linux/regmap.h | 28 + include/sound/cs4271.h | 15 + include/sound/max98090.h | 29 + include/sound/saif.h | 16 - include/sound/sh_fsi.h | 70 +- include/sound/simple_card.h | 12 +- include/sound/soc-dai.h | 8 +- include/sound/soc.h | 6 +- include/sound/tlv320aic3x.h | 10 + include/sound/wm2000.h | 3 - include/sound/wm2200.h | 22 +- sound/arm/pxa2xx-ac97-lib.c | 8 +- sound/soc/atmel/Kconfig | 6 +- sound/soc/atmel/atmel-pcm-pdc.c | 4 +- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel-pcm.h | 6 +- sound/soc/atmel/atmel_ssc_dai.c | 14 +- sound/soc/atmel/sam9g20_wm8731.c | 6 +- sound/soc/codecs/ak4642.c | 33 +- sound/soc/codecs/arizona.c | 247 +- sound/soc/codecs/arizona.h | 8 +- sound/soc/codecs/cs4271.c | 34 + sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/max98090.c | 2685 ++++++++++++++++---- sound/soc/codecs/max98090.h | 1549 +++++++++++ sound/soc/codecs/tlv320aic3x.c | 87 +- sound/soc/codecs/tlv320aic3x.h | 4 + sound/soc/codecs/tlv320dac33.c | 16 - sound/soc/codecs/twl4030.c | 85 +- sound/soc/codecs/twl6040.c | 62 +- sound/soc/codecs/wm2000.c | 66 +- sound/soc/codecs/wm2000.h | 3 + sound/soc/codecs/wm2200.c | 62 +- sound/soc/codecs/wm5100.c | 13 + sound/soc/codecs/wm5102.c | 172 +- sound/soc/codecs/wm5110.c | 111 +- sound/soc/codecs/wm8350.c | 10 +- sound/soc/codecs/wm8804.c | 3 - sound/soc/codecs/wm8962.c | 37 +- sound/soc/codecs/wm8983.c | 41 +- sound/soc/codecs/wm8985.c | 43 +- sound/soc/codecs/wm_adsp.c | 525 +++- sound/soc/codecs/wm_adsp.h | 18 + sound/soc/codecs/wmfw.h | 15 +- sound/soc/davinci/davinci-evm.c | 6 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/dwc/designware_i2s.c | 4 + sound/soc/generic/simple-card.c | 63 +- sound/soc/mxs/mxs-saif.c | 49 +- sound/soc/omap/Kconfig | 19 +- sound/soc/omap/Makefile | 4 - sound/soc/omap/n810.c | 4 +- sound/soc/omap/omap-hdmi.c | 2 + sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-twl4030.c | 204 +- sound/soc/omap/omap3pandora.c | 8 +- sound/soc/omap/rx51.c | 8 +- sound/soc/omap/sdp3430.c | 278 -- sound/soc/omap/zoom2.c | 207 -- sound/soc/pxa/palm27x.c | 38 +- sound/soc/samsung/Kconfig | 6 +- sound/soc/samsung/dma.c | 3 +- sound/soc/samsung/dma.h | 1 + sound/soc/samsung/i2s.c | 267 +- sound/soc/samsung/i2s.h | 7 - sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 7 +- sound/soc/samsung/smdk_wm8994.c | 30 +- sound/soc/sh/fsi.c | 242 +- sound/soc/soc-compress.c | 123 +- sound/soc/soc-core.c | 135 +- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-pcm.c | 19 +- sound/soc/tegra/Kconfig | 19 + sound/soc/tegra/Makefile | 4 + sound/soc/tegra/tegra20_ac97.c | 480 ++++ sound/soc/tegra/tegra20_ac97.h | 95 + sound/soc/tegra/tegra20_das.c | 13 + sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/tegra/tegra_asoc_utils.c | 53 + sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_wm9712.c | 176 ++ sound/soc/ux500/mop500.c | 2 +- 114 files changed, 7824 insertions(+), 2023 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,fsi.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt create mode 100755 include/sound/max98090.h delete mode 100644 include/sound/saif.h mode change 100644 => 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h delete mode 100644 sound/soc/omap/sdp3430.c delete mode 100644 sound/soc/omap/zoom2.c create mode 100644 sound/soc/tegra/tegra20_ac97.c create mode 100644 sound/soc/tegra/tegra20_ac97.h create mode 100644 sound/soc/tegra/tegra_wm9712.c

2 1

Re: [alsa-devel] Sound record on Devkit8000 with DT boot issue
by Peter Ujfalusi 11 Feb '13

11 Feb '13

On 02/10/2013 04:21 PM, Anil Kumar wrote: > Hi All, > > I am working on Devkit8000 and adding DT support for that. I am trying > to add sound DT node support for this boards. > I have nearly done with this but facing issue on record. I do not have > much knowledge of twl4030 audio codec but with > small Hack[1] (as MainMic connected to the input jack) i am able to > record sound on Devkit8000. > I do not know that MICBIAS1_EN can be enable by amixer commands. > Could you please help me to get out of this ? You need to pass the routing information via DT for Devkit8000. This is not done for the BeagleBoard since it works without it (no need for MICBIAS for example) > > > 1) Hack [1] > > diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c > index 63b280b..6780681 100644 > --- a/sound/soc/codecs/twl4030.c > +++ b/sound/soc/codecs/twl4030.c > @@ -1473,7 +1473,7 @@ static const struct snd_soc_dapm_widget > twl4030_dapm_widgets[] = { > SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0, > NULL, 0), > > - SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), > + SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 1), Yes, this is HACK and it is wrong at the same time. > SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), > SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0), > > > 2) Sound DT node patch for Devkit8000 > > diff --git a/arch/arm/boot/dts/omap3- > devkit8000.dts b/arch/arm/boot/dts/omap3-devkit8000.dts > index dc59272..5f3eec0 100644 > --- a/arch/arm/boot/dts/omap3-devkit8000.dts > +++ b/arch/arm/boot/dts/omap3-devkit8000.dts > @@ -44,11 +44,22 @@ > }; > > }; > + > + sound { > + compatible = "ti,omap-twl4030"; > + ti,model = "devkit8000"; > + > + ti,mcbsp = <&mcbsp2>; > + ti,codec = <&twl_audio>; Here you should have something like: ti,audio-routing = "Ext Spk", "PREDRIVEL", "Ext Spk", "PREDRIVER", "MAINMIC", "Main Mic", "Main Mic", "Mic Bias 1"; So you connect the MicBias1 to Main Mic. When you start playback MicBias1 will be turned on for you. > + }; > }; > > &omap3_pmx_core { > pinctrl-names = "default"; > - pinctrl-0 = <&i2c1_pins>; > + pinctrl-0 = < > + &i2c1_pins > + &mcbsp2_pins > + >; > > leds_pins: pinmux_led_pins { > pinctrl-single,pins = < > @@ -65,6 +76,21 @@ > 0x18c 0x118 /* I2C1_SDA */ > >; > }; > + > + mcbsp2_pins: pinmux_mcbsp2_pins { > + pinctrl-single,pins = < > + /* > + * MCBSP2_FSX, OMAP_MUX_MODE0 | OMAP_PIN_INPUT, > + * MCBSP2_CLKX, OMAP_MUX_MODE0 | OMAP_PIN_INPUT > + */ > + 0x10c 0x01000100 > + /* > + * MCBSP2_DR, OMAP_MUX_MODE0 | OMAP_PIN_INPUT, > + * MCBSP2_DX, OMAP_MUX_MODE0 | OMAP_PIN_OUTPUT > + */ > + 0x110 0x00000100 > + >; > + }; > }; > > &i2c1 { > @@ -74,6 +100,14 @@ > reg = <0x48>; > interrupts = <7>; /* SYS_NIRQ cascaded to intc */ > interrupt-parent = <&intc>; > + > + twl_audio: audio { > + compatible = "ti,twl4030-audio"; > + codec { > + ti,ramp_delay_value = <3>; > + ti,offset_cncl_path = <0x31>; > + }; > + }; > }; > }; > > 3) Playback amixer commands > amixer set 'PredriveR Mixer AudioR2' on > amixer set 'PredriveL Mixer AudioL2' on > amixer set PreDriv 100 unmute > amixer set 'DAC2 Digital Fine' 100 > > Thanks, > Anil > -- Péter

1 0

[alsa-devel] [PATCH RFC 00/20] More patches for HD-audio generic auto-parser
by Takashi Iwai 11 Feb '13

11 Feb '13

Hi, this is a series of patches for more enhancements of HD-audio generic parser I posted shortly ago. All patches are found in sound-unstable git tree test/hda-gen-parser branch (and merged to master branch, too). git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Takashi

2 22

[alsa-devel] [PATCH] ASoC: wm_adsp: round to 4-byte boundary for coeff file blocks
by Chris Rattray 11 Feb '13

11 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index be45e2b..f3f7e75 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -711,6 +711,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) int ret, pos, blocks, type, offset, reg; char *file; struct wm_adsp_buf *buf; + int tmp; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -842,7 +843,12 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } } - pos += le32_to_cpu(blk->len) + sizeof(*blk); + tmp = le32_to_cpu(blk->len) % 4; + if (tmp) + pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk); + else + pos += le32_to_cpu(blk->len) + sizeof(*blk); + blocks++; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH] Add M2Tech hiFace USB-SPDIF driver
by Antonio Ospite 11 Feb '13

11 Feb '13

Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices. The supported products are: * M2Tech Young * M2Tech hiFace * M2Tech North Star * M2Tech W4S Young * M2Tech Corrson * M2Tech AUDIA * M2Tech SL Audio * M2Tech Empirical * M2Tech Rockna * M2Tech Pathos * M2Tech Metronome * M2Tech CAD * M2Tech Audio Esclusive * M2Tech Rotel * M2Tech Eeaudio * The Chord Company CHORD * AVA Group A/S Vitus Signed-off-by: Antonio Ospite <ao2(a)amarulasolutions.com> --- Hi, this is my first ALSA driver I still have some doubts and I marked them in the comments below with a XXX prefix, could someone please take a look at them? I plan on submitting a version 2 of the driver after I get some feedback, especially on those doubts. I also noticed that many drivers under sound/usb/ have text in the Kconfig entry stating "Say Y here to include support for ..." but I only see "[N/m/?]" as options when I run "make oldconfig" maybe because of some dependency, I am not sure, should such text be removed in order to avoid confusion? Thanks, Antonio sound/usb/Kconfig | 33 +++ sound/usb/Makefile | 2 +- sound/usb/hiface/Makefile | 2 + sound/usb/hiface/chip.c | 331 ++++++++++++++++++++++ sound/usb/hiface/chip.h | 34 +++ sound/usb/hiface/pcm.c | 666 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/hiface/pcm.h | 24 ++ 7 files changed, 1091 insertions(+), 1 deletion(-) create mode 100644 sound/usb/hiface/Makefile create mode 100644 sound/usb/hiface/chip.c create mode 100644 sound/usb/hiface/chip.h create mode 100644 sound/usb/hiface/pcm.c create mode 100644 sound/usb/hiface/pcm.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd7..2b206e8 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -115,5 +115,38 @@ config SND_USB_6FIRE and further help can be found at http://sixfireusb.sourceforge.net +config SND_USB_HIFACE + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help + Say Y here to include support for M2Tech hiFace USB-SPDIF interface. + + M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output + Interface, see http://www.m2tech.biz/hiface.html + + This driver supports the original M2Tech hiFace and some other + compatible devices. The supported products are: + + * M2Tech Young + * M2Tech hiFace + * M2Tech North Star + * M2Tech W4S Young + * M2Tech Corrson + * M2Tech AUDIA + * M2Tech SL Audio + * M2Tech Empirical + * M2Tech Rockna + * M2Tech Pathos + * M2Tech Metronome + * M2Tech CAD + * M2Tech Audio Esclusive + * M2Tech Rotel + * M2Tech Eeaudio + * The Chord Company CHORD + * AVA Group A/S Vitus + + To compile this driver as a module, choose M here: the module + will be called snd-usb-hiface. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index ac256dc..abe668f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile new file mode 100644 index 0000000..463b136 --- /dev/null +++ b/sound/usb/hiface/Makefile @@ -0,0 +1,2 @@ +snd-usb-hiface-objs := chip.o pcm.o +obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c new file mode 100644 index 0000000..dce74e9 --- /dev/null +++ b/sound/usb/hiface/chip.c @@ -0,0 +1,331 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <sound/initval.h> + +#include "chip.h" +#include "pcm.h" + +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_AUTHOR("Antonio Ospite <ao2(a)amarulasolutions.com>"); +MODULE_DESCRIPTION("M2Tech HiFace USB audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{HiFace, Evo}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +#define DRIVER_NAME "snd-usb-hiface" +#define CARD_NAME "HiFace" + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +static struct hiface_chip *chips[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; + +static DEFINE_MUTEX(register_mutex); + +struct hiface_vendor_quirk { + const char *device_name; + u8 extra_freq; +}; + +static int hiface_dev_free(struct snd_device *device) +{ + struct hiface_chip *chip = device->device_data; + kfree(chip); + return 0; +} + +static int hiface_chip_create(struct usb_device *device, int idx, + const struct hiface_vendor_quirk *quirk, + struct hiface_chip **rchip) +{ + struct snd_card *card = NULL; + struct hiface_chip *chip; + int ret; + static struct snd_device_ops ops = { + .dev_free = hiface_dev_free, + }; + + *rchip = NULL; + + /* if we are here, card can be registered in alsa. */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot create alsa card.\n"); + return ret; + } + + strcpy(card->driver, DRIVER_NAME); + + if (quirk && quirk->device_name) + strcpy(card->shortname, quirk->device_name); + else + strcpy(card->shortname, "M2Tech generic audio"); + + sprintf(card->longname, "%s at %d:%d", card->shortname, + device->bus->busnum, device->devnum); + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + snd_card_free(card); + return -ENOMEM; + } + + chip->dev = device; + chip->index = idx; + chip->card = card; + + ret = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (ret < 0) { + kfree(chip); + snd_card_free(card); + return ret; + } + + *rchip = chip; + return 0; +} + +static int hiface_chip_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct hiface_vendor_quirk *quirk = (struct hiface_vendor_quirk *)usb_id->driver_info; + int ret; + int i; + struct hiface_chip *chip; + struct usb_device *device = interface_to_usbdev(intf); + + pr_info("Probe " DRIVER_NAME " driver.\n"); + + ret = usb_set_interface(device, 0, 0); + if (ret != 0) { + snd_printk(KERN_ERR "can't set first interface for " CARD_NAME " device.\n"); + return -EIO; + } + + /* check whether the card is already registered */ + chip = NULL; + mutex_lock(&register_mutex); + for (i = 0; i < SNDRV_CARDS; i++) { + if (chips[i] && chips[i]->dev == device) { + if (chips[i]->shutdown) { + snd_printk(KERN_ERR CARD_NAME " device is in the shutdown state, cannot create a card instance\n"); + ret = -ENODEV; + goto err; + } + chip = chips[i]; + break; + } + } + if (!chip) { + /* it's a fresh one. + * now look for an empty slot and create a new card instance + */ + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i] && !chips[i]) { + ret = hiface_chip_create(device, i, quirk, + &chip); + if (ret < 0) + goto err; + + snd_card_set_dev(chip->card, &intf->dev); + break; + } + if (!chip) { + snd_printk(KERN_ERR "no available " CARD_NAME " audio device\n"); + ret = -ENODEV; + goto err; + } + } + + ret = hiface_pcm_init(chip, quirk ? quirk->extra_freq : 0); + if (ret < 0) + goto err_chip_destroy; + + ret = snd_card_register(chip->card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot register " CARD_NAME " card\n"); + goto err_chip_destroy; + } + + chips[chip->index] = chip; + chip->intf_count++; + + mutex_unlock(&register_mutex); + + usb_set_intfdata(intf, chip); + return 0; + +err_chip_destroy: + snd_card_free(chip->card); +err: + mutex_unlock(&register_mutex); + return ret; +} + +static void hiface_chip_disconnect(struct usb_interface *intf) +{ + struct hiface_chip *chip; + struct snd_card *card; + + pr_debug("%s: called.\n", __func__); + + chip = usb_get_intfdata(intf); + if (!chip) + return; + + card = chip->card; + chip->intf_count--; + if (chip->intf_count <= 0) { + /* Make sure that the userspace cannot create new request */ + snd_card_disconnect(card); + + mutex_lock(&register_mutex); + chips[chip->index] = NULL; + mutex_unlock(&register_mutex); + + chip->shutdown = true; + hiface_pcm_abort(chip); + snd_card_free_when_closed(card); + } +} + +static struct usb_device_id device_table[] = { + { + USB_DEVICE(0x04b4, 0x0384), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Young", + .extra_freq = 1, + } + }, + { + USB_DEVICE(0x04b4, 0x930b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "hiFace", + } + }, + { + USB_DEVICE(0x04b4, 0x931b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "North Star", + } + }, + { + USB_DEVICE(0x04b4, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "W4S Young", + } + }, + { + USB_DEVICE(0x04b4, 0x931d), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Corrson", + } + }, + { + USB_DEVICE(0x04b4, 0x931e), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "AUDIA", + } + }, + { + USB_DEVICE(0x04b4, 0x931f), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "SL Audio", + } + }, + { + USB_DEVICE(0x04b4, 0x9320), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Empirical", + } + }, + { + USB_DEVICE(0x04b4, 0x9321), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rockna", + } + }, + { + USB_DEVICE(0x249c, 0x9001), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Pathos", + } + }, + { + USB_DEVICE(0x249c, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Metronome", + } + }, + { + USB_DEVICE(0x249c, 0x9006), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CAD", + } + }, + { + USB_DEVICE(0x249c, 0x9008), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Audio Esclusive", + } + }, + { + USB_DEVICE(0x249c, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rotel", + } + }, + { + USB_DEVICE(0x249c, 0x932c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Eeaudio", + } + }, + { + USB_DEVICE(0x245f, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CHORD", + } + }, + { + USB_DEVICE(0x25c6, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Vitus", + } + }, + {} +}; + +MODULE_DEVICE_TABLE(usb, device_table); + +static struct usb_driver hiface_usb_driver = { + .name = DRIVER_NAME, + .probe = hiface_chip_probe, + .disconnect = hiface_chip_disconnect, + .id_table = device_table, +}; + +module_usb_driver(hiface_usb_driver); diff --git a/sound/usb/hiface/chip.h b/sound/usb/hiface/chip.h new file mode 100644 index 0000000..c2e9fda --- /dev/null +++ b/sound/usb/hiface/chip.h @@ -0,0 +1,34 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_CHIP_H +#define HIFACE_CHIP_H + +#include <linux/usb.h> +#include <sound/core.h> + +struct pcm_runtime; + +struct hiface_chip { + struct usb_device *dev; + struct snd_card *card; + int intf_count; /* number of registered interfaces */ + int index; /* index in module parameter arrays */ + bool shutdown; + + struct pcm_runtime *pcm; +}; +#endif /* HIFACE_CHIP_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c new file mode 100644 index 0000000..34fe0ed --- /dev/null +++ b/sound/usb/hiface/pcm.c @@ -0,0 +1,666 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/slab.h> +#include <sound/pcm.h> + +#include "pcm.h" +#include "chip.h" + +#define OUT_EP 0x2 +#define PCM_N_URBS 8 +#define PCM_MAX_PACKET_SIZE 4096 +#define MAX_BUFSIZE (2 * PCM_N_URBS * PCM_MAX_PACKET_SIZE) + +struct pcm_urb { + struct hiface_chip *chip; + + struct urb instance; + struct usb_anchor submitted; + u8 *buffer; +}; + +struct pcm_substream { + spinlock_t lock; + struct snd_pcm_substream *instance; + + bool active; + snd_pcm_uframes_t dma_off; /* current position in alsa dma_area */ + snd_pcm_uframes_t period_off; /* current position in current period */ +}; + +enum { /* pcm streaming states */ + STREAM_DISABLED, /* no pcm streaming */ + STREAM_STARTING, /* pcm streaming requested, waiting to become ready */ + STREAM_RUNNING, /* pcm streaming running */ + STREAM_STOPPING +}; + +struct pcm_runtime { + struct hiface_chip *chip; + struct snd_pcm *instance; + + struct pcm_substream playback; + bool panic; /* if set driver won't do anymore pcm on device */ + + struct pcm_urb out_urbs[PCM_N_URBS]; + + struct mutex stream_mutex; + u8 stream_state; /* one of STREAM_XXX */ + u8 rate; /* one of PCM_RATE_XXX */ + u8 extra_freq; + wait_queue_head_t stream_wait_queue; + bool stream_wait_cond; +}; + +static const unsigned int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000 }; +static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates) - 2, /* by default rates up to 192000 are supported */ + .list = rates, + .mask = 0, +}; + +static const int rates_alsaid[] = { + SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_48000, + SNDRV_PCM_RATE_88200, SNDRV_PCM_RATE_96000, + SNDRV_PCM_RATE_176400, SNDRV_PCM_RATE_192000, + SNDRV_PCM_RATE_KNOT, SNDRV_PCM_RATE_KNOT }; + +static const struct snd_pcm_hardware pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, + + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + + .rate_min = 44100, + .rate_max = 192000, /* changes in hiface_pcm_open to support extra rates */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_BUFSIZE, + .period_bytes_min = PCM_MAX_PACKET_SIZE, + .period_bytes_max = MAX_BUFSIZE, + .periods_min = 2, + .periods_max = 1024 +}; + +static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) +{ + u8 rate_value[] = { 0x43, 0x4b, 0x42, 0x4a, 0x40, 0x48, 0x58, 0x68 }; + int ret; + struct usb_device *device = rt->chip->dev; + + /* We are already sure that the rate is supported here thanks to the + * contraints set in hiface_pcm_open(), but we have to retrieve the + * index to access rate_value. + */ + for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + if (rate == rates[rt->rate]) + break; + + if (unlikely(rt->rate == ARRAY_SIZE(rates))) { + pr_err("Unsupported rate %d\n", rate); + return -EINVAL; + } + + /* + * USBIO: Vendor 0xb0(wValue=0x0043, wIndex=0x0000) + * 43 b0 43 00 00 00 00 00 + * USBIO: Vendor 0xb0(wValue=0x004b, wIndex=0x0000) + * 43 b0 4b 00 00 00 00 00 + * This control message doesn't have any ack from the + * other side + */ + ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), + 0xb0, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + rate_value[rt->rate], 0, NULL, 0, 100); + if (ret < 0) { + snd_printk(KERN_ERR "Error setting samplerate %d.\n", + rates[rt->rate]); + return ret; + } + + return 0; +} + +static struct pcm_substream *hiface_pcm_get_substream( + struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rt->playback; + + pr_debug("Error getting pcm substream slot.\n"); + return NULL; +} + +/* call with stream_mutex locked */ +static void hiface_pcm_stream_stop(struct pcm_runtime *rt) +{ + int i, time; + + if (rt->stream_state != STREAM_DISABLED) { + rt->stream_state = STREAM_STOPPING; + + for (i = 0; i < PCM_N_URBS; i++) { + time = usb_wait_anchor_empty_timeout( + &rt->out_urbs[i].submitted, 100); + if (!time) + usb_kill_anchored_urbs( + &rt->out_urbs[i].submitted); + usb_kill_urb(&rt->out_urbs[i].instance); + } + + rt->stream_state = STREAM_DISABLED; + } +} + +/* call with stream_mutex locked */ +static int hiface_pcm_stream_start(struct pcm_runtime *rt) +{ + int ret = 0, i; + + if (rt->stream_state == STREAM_DISABLED) { + /* submit our out urbs zero init */ + rt->stream_state = STREAM_STARTING; + for (i = 0; i < PCM_N_URBS; i++) { + memset(rt->out_urbs[i].buffer, 0, PCM_MAX_PACKET_SIZE); + usb_anchor_urb(&rt->out_urbs[i].instance, + &rt->out_urbs[i].submitted); + ret = usb_submit_urb(&rt->out_urbs[i].instance, + GFP_ATOMIC); + if (ret) { + hiface_pcm_stream_stop(rt); + return ret; + } + } + + /* wait for first out urb to return (sent in in urb handler) */ + wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, + HZ); + if (rt->stream_wait_cond) { + pr_debug("%s: Stream is running wakeup event\n", + __func__); + rt->stream_state = STREAM_RUNNING; + } else { + hiface_pcm_stream_stop(rt); + return -EIO; + } + } + return ret; +} + + +/* The hardware wants word-swapped 32-bit values */ +void memcpy_swahw32(u8 *dest, u8 *src, unsigned int n) +{ + unsigned int i; + + for (i = 0; i < n / 4; i++) + ((u32 *)dest)[i] = swahw32(((u32 *)src)[i]); +} + +/* call with substream locked */ +static int hiface_pcm_playback(struct pcm_substream *sub, + struct pcm_urb *urb) +{ + struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; + u8 *source; + unsigned int pcm_buffer_size; + + /* XXX Can an invalid format make it to here? + * Isn't ALSA checking pcm_hw.formats ? + */ + if (alsa_rt->format != SNDRV_PCM_FORMAT_S32_LE) { + pr_err("Unsupported sample format\n"); + return -EINVAL; + } + + pcm_buffer_size = snd_pcm_lib_buffer_bytes(sub->instance); + + if (sub->dma_off + PCM_MAX_PACKET_SIZE <= pcm_buffer_size) { + pr_debug("%s: (1) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, PCM_MAX_PACKET_SIZE); + } else { + /* wrap around at end of ring buffer */ + unsigned int len; + + pr_debug("%s: (2) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + len = pcm_buffer_size - sub->dma_off; + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, len); + + source = alsa_rt->dma_area; + memcpy_swahw32(urb->buffer + len, source, + PCM_MAX_PACKET_SIZE - len); + } + sub->dma_off += PCM_MAX_PACKET_SIZE; + if (sub->dma_off >= pcm_buffer_size) + sub->dma_off -= pcm_buffer_size; + + sub->period_off += PCM_MAX_PACKET_SIZE; + + return 0; +} + +static void hiface_pcm_out_urb_handler(struct urb *usb_urb) +{ + struct pcm_urb *out_urb = usb_urb->context; + struct pcm_runtime *rt = out_urb->chip->pcm; + struct pcm_substream *sub; + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (usb_urb->status || rt->panic || rt->stream_state == STREAM_STOPPING) + return; + + if (rt->stream_state == STREAM_STARTING) { + rt->stream_wait_cond = true; + wake_up(&rt->stream_wait_queue); + } + + /* now send our playback data (if a free out urb was found) */ + sub = &rt->playback; + spin_lock_irqsave(&sub->lock, flags); + if (sub->active) { + int ret; + + ret = hiface_pcm_playback(sub, out_urb); + if (ret < 0) { + spin_unlock_irqrestore(&sub->lock, flags); + goto out_fail; + } + if (sub->period_off >= sub->instance->runtime->period_size) { + sub->period_off %= sub->instance->runtime->period_size; + spin_unlock_irqrestore(&sub->lock, flags); + snd_pcm_period_elapsed(sub->instance); + } else { + spin_unlock_irqrestore(&sub->lock, flags); + } + } else { + memset(out_urb->buffer, 0, PCM_MAX_PACKET_SIZE); + spin_unlock_irqrestore(&sub->lock, flags); + } +out_fail: + usb_submit_urb(&out_urb->instance, GFP_ATOMIC); +} + +static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = NULL; + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + + mutex_lock(&rt->stream_mutex); + alsa_rt->hw = pcm_hw; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + /* XXX can we avoid setting hw.rates below? + * + * FWIU setting alsa_rt->hw.rates in the open callback can be + * useful in two cases: + * + * - different hardware revisions have different + * capabilities + * + * - different stream types must have the same capabilities + * + * but in our case it should not be needed as we only have one + * playback stream and we define the capabilities via + * constraints already to avoid invalid rates to be used. + * + * Moreover, following the code stolen from 6fire it seems + * impossible to get to setting alsa_rt->hw.rates here: + * + * hiface_pcm_init: + * rt->rate = ARRAY_SIZE(rates); + * + * hiface_pcm_open: + * if (rt->rate < ARRAY_SIZE(rates)) + * alsa_rt->hw.rates = rates_alsaid[rt->rate]; + * // So it does not get here on the first open + * + * hiface_pcm_prepare: + * hiface_pcm_set_rate: + * for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + * if (rate == rates[rt->rate]) + * break; + * + * hiface_pcm_close: + * rt->rate = ARRAY_SIZE(rates); + * // So it won't set alsa_rt->hw.rates at the next open either + * + * So the only way to set alsa_rt->hw.rates is to have two + * calls to the open callback, is that even possible? ALSA + * does not allow sharing PCM streams AFAIK and we use only + * one playback stream. + * + * Maybe the 6fire driver was setting alsa_rt->hw.rates in + * order to have the playback stream and the capture stream + * use the same rate? I.e. the first open,prepare,set_rate() + * sequence on one type of stream will decide the rate and the + * second open on the other type would restrict the supported + * rate to the one already in use, but I am not sure. + * + * If the rationale makes sense then we don't need to set + * alsa_rt->hw.rates and we can remove this and related code. + */ + if (rt->rate < ARRAY_SIZE(rates)) + alsa_rt->hw.rates = rates_alsaid[rt->rate]; + + sub = &rt->playback; + } + + if (!sub) { + mutex_unlock(&rt->stream_mutex); + pr_err("Invalid stream type\n"); + return -EINVAL; + } + + if (rt->extra_freq) { + alsa_rt->hw.rate_max = 384000; + constraints_rates.count = ARRAY_SIZE(rates); + } + + ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + + sub->instance = alsa_sub; + sub->active = false; + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + unsigned long flags; + + if (rt->panic) + return 0; + + pr_debug("%s: called.\n", __func__); + + mutex_lock(&rt->stream_mutex); + if (sub) { + /* deactivate substream */ + spin_lock_irqsave(&sub->lock, flags); + sub->instance = NULL; + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + + /* all substreams closed? if so, stop streaming */ + if (!rt->playback.instance) { + hiface_pcm_stream_stop(rt); + rt->rate = ARRAY_SIZE(rates); + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, + struct snd_pcm_hw_params *hw_params) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_malloc_pages(alsa_sub, + params_buffer_bytes(hw_params)); +} + +static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_free_pages(alsa_sub); +} + +static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + mutex_lock(&rt->stream_mutex); + + sub->dma_off = 0; + sub->period_off = 0; + + if (rt->stream_state == STREAM_DISABLED) { + + ret = hiface_pcm_set_rate(rt, alsa_rt->rate); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + ret = hiface_pcm_stream_start(rt); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_trigger(struct snd_pcm_substream *alsa_sub, int cmd) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&sub->lock, flags); + sub->active = true; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&sub->lock, flags); + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + snd_pcm_uframes_t dma_offset; + + if (rt->panic || !sub) + return SNDRV_PCM_STATE_XRUN; + + spin_lock_irqsave(&sub->lock, flags); + dma_offset = sub->dma_off; + spin_unlock_irqrestore(&sub->lock, flags); + return bytes_to_frames(alsa_sub->runtime, dma_offset); +} + +static struct snd_pcm_ops pcm_ops = { + .open = hiface_pcm_open, + .close = hiface_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hiface_pcm_hw_params, + .hw_free = hiface_pcm_hw_free, + .prepare = hiface_pcm_prepare, + .trigger = hiface_pcm_trigger, + .pointer = hiface_pcm_pointer, +}; + +static int hiface_pcm_init_urb(struct pcm_urb *urb, + struct hiface_chip *chip, + unsigned int ep, + void (*handler)(struct urb *)) +{ + urb->chip = chip; + usb_init_urb(&urb->instance); + + urb->buffer = kzalloc(PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!urb->buffer) + return -ENOMEM; + + usb_fill_bulk_urb(&urb->instance, chip->dev, + usb_sndbulkpipe(chip->dev, ep), (void *)urb->buffer, + PCM_MAX_PACKET_SIZE, handler, urb); + init_usb_anchor(&urb->submitted); + + return 0; +} + +void hiface_pcm_abort(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + + if (rt) { + rt->panic = true; + + if (rt->playback.instance) { + snd_pcm_stop(rt->playback.instance, + SNDRV_PCM_STATE_XRUN); + } + mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + mutex_unlock(&rt->stream_mutex); + } +} + +static void hiface_pcm_destroy(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + int i; + + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + + kfree(chip->pcm); + chip->pcm = NULL; +} + +static void hiface_pcm_free(struct snd_pcm *pcm) +{ + struct pcm_runtime *rt = pcm->private_data; + + pr_debug("%s: called.\n", __func__); + + if (rt) + hiface_pcm_destroy(rt->chip); +} + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) +{ + int i; + int ret; + struct snd_pcm *pcm; + struct pcm_runtime *rt; + + rt = kzalloc(sizeof(*rt), GFP_KERNEL); + if (!rt) + return -ENOMEM; + + rt->chip = chip; + rt->stream_state = STREAM_DISABLED; + rt->rate = ARRAY_SIZE(rates); + if (extra_freq) + rt->extra_freq = 1; + + init_waitqueue_head(&rt->stream_wait_queue); + mutex_init(&rt->stream_mutex); + spin_lock_init(&rt->playback.lock); + + for (i = 0; i < PCM_N_URBS; i++) + hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, + hiface_pcm_out_urb_handler); + + ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); + if (ret < 0) { + kfree(rt); + pr_err("Cannot create pcm instance\n"); + return ret; + } + + pcm->private_data = rt; + pcm->private_free = hiface_pcm_free; + + strcpy(pcm->name, "USB-SPDIF Audio"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MAX_BUFSIZE, MAX_BUFSIZE); + rt->instance = pcm; + + chip->pcm = rt; + return 0; +} diff --git a/sound/usb/hiface/pcm.h b/sound/usb/hiface/pcm.h new file mode 100644 index 0000000..ddc9fe7 --- /dev/null +++ b/sound/usb/hiface/pcm.h @@ -0,0 +1,24 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_PCM_H +#define HIFACE_PCM_H + +struct hiface_chip; + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq); +void hiface_pcm_abort(struct hiface_chip *chip); +#endif /* HIFACE_PCM_H */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH - add support for M-Audio FT C600 1/1] ALSA: usb-audio: add support for M-Audio FT C600
by Matt Gruskin 10 Feb '13

10 Feb '13

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e90daf8..638e7f7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -807,6 +807,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, { switch (cval->mixer->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (strcmp(kctl->id.name, "Effect Duration") == 0) { cval->min = 0x0000; cval->max = 0xffff; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0e2ed3d..cc2dd1f 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -380,6 +380,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .selector_map = c400_selectors, }, { + .id = USB_ID(0x0763, 0x2031), + .selector_map = c400_selectors, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 15520de..497d274 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -637,7 +637,7 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, } /* M-Audio FastTrack Ultra quirks */ -/* FTU Effect switch (also used by C400) */ +/* FTU Effect switch (also used by C400/C600) */ struct snd_ftu_eff_switch_priv_val { struct usb_mixer_interface *mixer; int cached_value; @@ -1029,32 +1029,45 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } -/* M-Audio Fast Track C400 */ -/* C400 volume controls, this control needs a volume quirk, see mixer.c */ +/* M-Audio Fast Track C400/C600 */ +/* C400/C600 volume controls, this control needs a volume quirk, see mixer.c */ static int snd_c400_create_vol_ctls(struct usb_mixer_interface *mixer) { char name[64]; unsigned int cmask, offset; int out, chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - for (out = 0; out < 6; out++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + for (out = 0; out < num_outs; out++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "PCM%d-Out%d Playback Volume", chan + 1, out + 1); } else { snprintf(name, sizeof(name), "In%d-Out%d Playback Volume", - chan - 5, out + 1); + chan - num_outs + 1, out + 1); } cmask = (out == 0) ? 0 : 1 << (out - 1); - offset = chan * 6; + offset = chan * num_outs; err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1110,20 +1123,33 @@ static int snd_c400_create_effect_vol_ctls(struct usb_mixer_interface *mixer) char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x42; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "Effect Send DOut%d", chan + 1); } else { snprintf(name, sizeof(name), "Effect Send AIn%d", - chan - 5); + chan - num_outs + 1); } cmask = (chan == 0) ? 0 : 1 << (chan - 1); @@ -1142,20 +1168,33 @@ static int snd_c400_create_effect_ret_vol_ctls(struct usb_mixer_interface *mixer char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int offset = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - const int chan_id[6] = { 0, 7, 2, 9, 4, 0xb }; - const unsigned int offset = 0x3c; - /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ - for (chan = 0; chan < 6; chan++) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + offset = 0x3c; + /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + offset = 0x70; + /* { 0x70, 0x79, 0x72, 0x7b, 0x74, 0x7d, 0x76, 0x7f } */ + break; + } + + for (chan = 0; chan < num_outs; chan++) { snprintf(name, sizeof(name), "Effect Return %d", chan + 1); - cmask = (chan_id[chan] == 0) ? 0 : 1 << (chan_id[chan] - 1); + cmask = (chan == 0) ? 0 : + 1 << (chan + (chan % 2) * num_outs - 1); err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1299,6 +1338,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C400 */ err = snd_c400_create_mixer(mixer); break; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d82e378..749752a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -363,6 +363,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (is_playback) { implicit_fb = 1; ep = 0x81; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 64d25a7..4901400 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2326,6 +2326,77 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0763, 0x2031), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + /* .vendor_name = "M-Audio", */ + /* .product_name = "Fast Track C600", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* Capture */ + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* MIDI */ + { + .ifnum = -1 /* Interface = 4 */ + } + } + } +}, +{ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2c97185..7d7ad0b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -863,13 +863,14 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 4; /* - * M-Audio Fast Track C400 - when packets are not skipped, real world - * latency varies by approx. +/- 50 frames (at 96KHz) each time the - * stream is (re)started. When skipping packets 16 at endpoint start - * up, the real world latency is stable within +/- 1 frame (also + * M-Audio Fast Track C400/C600 - when packets are not skipped, real + * world latency varies by approx. +/- 50 frames (at 96KHz) each time + * the stream is (re)started. When skipping packets 16 at endpoint + * start up, the real world latency is stable within +/- 1 frame (also * across power cycles). */ - if (ep->chip->usb_id == USB_ID(0x0763, 0x2030) && + if ((ep->chip->usb_id == USB_ID(0x0763, 0x2030) || + ep->chip->usb_id == USB_ID(0x0763, 0x2031)) && ep->type == SND_USB_ENDPOINT_TYPE_DATA) ep->skip_packets = 16; } -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 2/5] hda_intel: Add Device IDs for Intel Wellsburg PCH
by Takashi Iwai 10 Feb '13

10 Feb '13

At Fri, 8 Feb 2013 17:29:40 -0800, James Ralston wrote: > > This patch adds the HD Audio Device IDs for the Intel Wellsburg PCH > > Signed-off-by: James Ralston <james.d.ralston(a)intel.com> Thanks, applied. Takashi > --- > sound/pci/hda/hda_intel.c | 5 +++++ > 1 files changed, 5 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index c78286f..3218f4a 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -3611,6 +3611,11 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* Lynx Point */ > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > + /* Wellsburg */ > + { PCI_DEVICE(0x8086, 0x8d20), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > + { PCI_DEVICE(0x8086, 0x8d21), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > /* Lynx Point-LP */ > { PCI_DEVICE(0x8086, 0x9c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > -- > 1.7.7.6 >

1 0

[alsa-devel] [PATCH - add support for M-Audio FT C600 1/1] ALSA: usb-audio: add support for M-Audio FT C600
by Matt Gruskin 09 Feb '13

09 Feb '13

Adds quirks and mixer support for the M-Audio Fast Track C600 USB audio interface. This device is very similar to the C400 - the C600 simply has some more inputs and outputs, so the existing C400 support is extended to support this device as well. Signed-off-by: Matt Gruskin <matthew.gruskin(a)gmail.com> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e90daf8..638e7f7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -807,6 +807,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, { switch (cval->mixer->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (strcmp(kctl->id.name, "Effect Duration") == 0) { cval->min = 0x0000; cval->max = 0xffff; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 0e2ed3d..cc2dd1f 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -380,6 +380,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .selector_map = c400_selectors, }, { + .id = USB_ID(0x0763, 0x2031), + .selector_map = c400_selectors, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 15520de..497d274 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -637,7 +637,7 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, } /* M-Audio FastTrack Ultra quirks */ -/* FTU Effect switch (also used by C400) */ +/* FTU Effect switch (also used by C400/C600) */ struct snd_ftu_eff_switch_priv_val { struct usb_mixer_interface *mixer; int cached_value; @@ -1029,32 +1029,45 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, } } -/* M-Audio Fast Track C400 */ -/* C400 volume controls, this control needs a volume quirk, see mixer.c */ +/* M-Audio Fast Track C400/C600 */ +/* C400/C600 volume controls, this control needs a volume quirk, see mixer.c */ static int snd_c400_create_vol_ctls(struct usb_mixer_interface *mixer) { char name[64]; unsigned int cmask, offset; int out, chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - for (out = 0; out < 6; out++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + for (out = 0; out < num_outs; out++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "PCM%d-Out%d Playback Volume", chan + 1, out + 1); } else { snprintf(name, sizeof(name), "In%d-Out%d Playback Volume", - chan - 5, out + 1); + chan - num_outs + 1, out + 1); } cmask = (out == 0) ? 0 : 1 << (out - 1); - offset = chan * 6; + offset = chan * num_outs; err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1110,20 +1123,33 @@ static int snd_c400_create_effect_vol_ctls(struct usb_mixer_interface *mixer) char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int num_ins = 0; const unsigned int id = 0x42; const int val_type = USB_MIXER_S16; const int control = 1; - for (chan = 0; chan < 10; chan++) { - if (chan < 6) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + num_ins = 4; + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + num_ins = 6; + break; + } + + for (chan = 0; chan < num_outs + num_ins; chan++) { + if (chan < num_outs) { snprintf(name, sizeof(name), "Effect Send DOut%d", chan + 1); } else { snprintf(name, sizeof(name), "Effect Send AIn%d", - chan - 5); + chan - num_outs + 1); } cmask = (chan == 0) ? 0 : 1 << (chan - 1); @@ -1142,20 +1168,33 @@ static int snd_c400_create_effect_ret_vol_ctls(struct usb_mixer_interface *mixer char name[64]; unsigned int cmask; int chan, err; + int num_outs = 0; + int offset = 0; const unsigned int id = 0x40; const int val_type = USB_MIXER_S16; const int control = 1; - const int chan_id[6] = { 0, 7, 2, 9, 4, 0xb }; - const unsigned int offset = 0x3c; - /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ - for (chan = 0; chan < 6; chan++) { + switch (mixer->chip->usb_id) { + case USB_ID(0x0763, 0x2030): + num_outs = 6; + offset = 0x3c; + /* { 0x3c, 0x43, 0x3e, 0x45, 0x40, 0x47 } */ + break; + case USB_ID(0x0763, 0x2031): + num_outs = 8; + offset = 0x70; + /* { 0x70, 0x79, 0x72, 0x7b, 0x74, 0x7d, 0x76, 0x7f } */ + break; + } + + for (chan = 0; chan < num_outs; chan++) { snprintf(name, sizeof(name), "Effect Return %d", chan + 1); - cmask = (chan_id[chan] == 0) ? 0 : 1 << (chan_id[chan] - 1); + cmask = (chan == 0) ? 0 : + 1 << (chan + (chan % 2) * num_outs - 1); err = snd_create_std_mono_ctl_offset(mixer, id, control, cmask, val_type, offset, name, &snd_usb_mixer_vol_tlv); @@ -1299,6 +1338,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C400 */ err = snd_c400_create_mixer(mixer); break; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d82e378..749752a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -363,6 +363,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ if (is_playback) { implicit_fb = 1; ep = 0x81; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 64d25a7..4901400 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2326,6 +2326,77 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0763, 0x2031), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + /* .vendor_name = "M-Audio", */ + /* .product_name = "Fast Track C600", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* Capture */ + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + }, + .clock = 0x80, + } + }, + /* MIDI */ + { + .ifnum = -1 /* Interface = 4 */ + } + } + } +}, +{ USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2c97185..7d7ad0b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -863,13 +863,14 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 4; /* - * M-Audio Fast Track C400 - when packets are not skipped, real world - * latency varies by approx. +/- 50 frames (at 96KHz) each time the - * stream is (re)started. When skipping packets 16 at endpoint start - * up, the real world latency is stable within +/- 1 frame (also + * M-Audio Fast Track C400/C600 - when packets are not skipped, real + * world latency varies by approx. +/- 50 frames (at 96KHz) each time + * the stream is (re)started. When skipping packets 16 at endpoint + * start up, the real world latency is stable within +/- 1 frame (also * across power cycles). */ - if (ep->chip->usb_id == USB_ID(0x0763, 0x2030) && + if ((ep->chip->usb_id == USB_ID(0x0763, 0x2030) || + ep->chip->usb_id == USB_ID(0x0763, 0x2031)) && ep->type == SND_USB_ENDPOINT_TYPE_DATA) ep->skip_packets = 16; } -- 1.7.9.5

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[alsa-devel] [PATCH 3/3] ALSA: hda - Yet another fix for broken HSW HDMI pin connections
by mengdong.lin＠intel.com 08 Feb '13

08 Feb '13

From: Takashi Iwai <tiwai(a)suse.de> A Haswell test machine showed that the invalid connection list, but this time it has only a single pin on the codec, thus the former fixup code doesn't work as it assumes the three pins blindly. This patch splits the former fixup code to two parts: - Enable eDP 1.2 for Haswell codec - Fix the connection list of pins on Haswell codec; the converter list is recorded dynamically in hdmi_add_cvt(), and applied in hdmi_add_pin() Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 54243c4..b9af281b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -84,6 +84,7 @@ struct hdmi_spec_per_pin { struct hdmi_spec { int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; + hda_nid_t cvt_nids[MAX_HDMI_CVTS]; int num_pins; struct hdmi_spec_per_pin pins[MAX_HDMI_PINS]; @@ -1197,6 +1198,9 @@ static void hdmi_repoll_eld(struct work_struct *work) hdmi_present_sense(per_pin, per_pin->repoll_count); } +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + hda_nid_t nid); + static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) { struct hdmi_spec *spec = codec->spec; @@ -1216,6 +1220,9 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (snd_BUG_ON(spec->num_pins >= MAX_HDMI_PINS)) return -E2BIG; + if (codec->vendor_id == 0x80862807) + intel_haswell_fixup_connect_list(codec, pin_nid); + pin_idx = spec->num_pins; per_pin = &spec->pins[pin_idx]; @@ -1263,7 +1270,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) if (err < 0) return err; - spec->num_cvts++; + spec->cvt_nids[spec->num_cvts++] = cvt_nid; return 0; } @@ -1691,27 +1698,22 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; -static void intel_haswell_fixup_connect_list(struct hda_codec *codec) -{ - unsigned int vendor_param; - hda_nid_t list[3] = {0x2, 0x3, 0x4}; - vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); - if (vendor_param == -1 || vendor_param & 0x02) - return; - - /* enable DP1.2 mode */ - vendor_param |= 0x02; - snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + hda_nid_t nid) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t conns[4]; + int nconns; - vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); - if (vendor_param == -1 || !(vendor_param & 0x02)) + nconns = snd_hda_get_connections(codec, nid, conns, ARRAY_SIZE(conns)); + if (nconns == spec->num_cvts && + !memcmp(conns, spec->cvt_nids, spec->num_cvts * sizeof(hda_nid_t))) return; - /* override 3 pins connection list */ - snd_hda_override_conn_list(codec, 0x05, 3, list); - snd_hda_override_conn_list(codec, 0x06, 3, list); - snd_hda_override_conn_list(codec, 0x07, 3, list); + /* override pins connection list */ + snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); + snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } #define INTEL_VENDOR_NID 0x08 @@ -1742,6 +1744,22 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec, return; } +static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) +{ + unsigned int vendor_param; + + vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + INTEL_GET_VENDOR_VERB, 0); + if (vendor_param == -1 || vendor_param & INTEL_EN_DP12) + return; + + /* enable DP1.2 mode */ + vendor_param |= INTEL_EN_DP12; + snd_hda_codec_write_cache(codec, INTEL_VENDOR_NID, 0, + INTEL_SET_VENDOR_VERB, vendor_param); +} + + /* available models for fixup */ enum { @@ -1780,7 +1798,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); if (codec->vendor_id == 0x80862807) - intel_haswell_fixup_connect_list(codec); + intel_haswell_fixup_enable_dp12(codec); if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; -- 1.7.10.4

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[alsa-devel] [PATCH 2/3] [ALSA] hda - Add fixup for Haswell to enable all pin and convertor widgets
by mengdong.lin＠intel.com 08 Feb '13

08 Feb '13

From: Mengdong Lin <mengdong.lin(a)intel.com> Some Haswell machines support more than one display outputs (HDMI or DP), but its BIOS may not enable the codec's 2nd and 3rd pin and output cvt widgets. This patch implements a board-specific fixup for Intel Haswell Machines: If the hidden pins are not enabled by BIOS, the driver will enable them and call common code to update the codec tree. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 899c4fb..54243c4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1714,6 +1714,57 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec) snd_hda_override_conn_list(codec, 0x07, 3, list); } +#define INTEL_VENDOR_NID 0x08 +#define INTEL_GET_VENDOR_VERB 0xf81 +#define INTEL_SET_VENDOR_VERB 0x781 +#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ +#define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ + +static void intel_haswell_enable_all_pins(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + INTEL_GET_VENDOR_VERB, 0); + if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS) + return; + + vendor_param |= INTEL_EN_ALL_PIN_CVTS; + vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, + INTEL_SET_VENDOR_VERB, vendor_param); + if (vendor_param == -1) + return; + + snd_hda_codec_update_widgets(codec); + return; +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_enable_all_pins, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { @@ -1725,6 +1776,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (codec->vendor_id == 0x80862807) intel_haswell_fixup_connect_list(codec); -- 1.7.10.4

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[alsa-devel] channel names
by Adrian Knoth 08 Feb '13

08 Feb '13

Hi! Am I missing something or doesn't ALSA support channel names? When dealing with multichannel interfaces, it's pretty inconvenient to remember whether channel 193 is on MADI 3, AES-EBU or phones. We can address this on the jackd level, however, since OSX allows audio drivers to name their channels, I wonder how to do the same with ALSA? Cheers

4 5

[alsa-devel] [PATCH 1/3] [ALSA] hda - Support rereading widgets under the function group
by mengdong.lin＠intel.com 08 Feb '13

08 Feb '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A codec may allow software to hide some unused pin/cvt widgets. Sometimes BIOS does not enable the hidden widgets properly although they are needed for the board. Thus the driver need to enable them as a board-specific fixup and the whole tree will change. This patch implements a common code for rereading codec widgets. So the fixup code can call it after enabling the hidden widgets. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3c92514..e80f835 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1441,6 +1441,30 @@ int snd_hda_codec_new(struct hda_bus *bus, } EXPORT_SYMBOL_HDA(snd_hda_codec_new); +int snd_hda_codec_update_widgets(struct hda_codec *codec) +{ + hda_nid_t fg; + int err; + + /* Assume the function group node does not change, + * only the widget nodes may change. + */ + kfree(codec->wcaps); + fg = codec->afg ? codec->afg : codec->mfg; + err = read_widget_caps(codec, fg); + if (err < 0) { + snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); + return err; + } + + snd_array_free(&codec->init_pins); + err = read_pin_defaults(codec); + + return err; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_update_widgets); + + /** * snd_hda_codec_configure - (Re-)configure the HD-audio codec * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0be1826..e8c9442 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -935,6 +935,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); +int snd_hda_codec_update_widgets(struct hda_codec *codec); /* * low level functions -- 1.7.10.4

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Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 07 Feb '13

07 Feb '13

On Thu, Feb 07, 2013 at 11:37:38AM +0000, Richard Fitzgerald wrote: > > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > > > 2) Send SNDRV_COMPRESS_DRAIN > > The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which > > expects the decoder to completely drain its buffers and come to complete halt. > > This would also mean the framework will treat a drained stream as stopped and > > needs a new start. Certainly we dont want that in this case. So we can't use > > SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that > > would overtly complicate this. If we are not doing proper drian lets not > > call it that. > > Ok, so let's keep the partial drain but split out the NEXT_TRACK hint so we keep > the two operations separate. It's clearer to understand. NEXT_TRACK tells DSP to > prepare for data for new track, PARTIAL_DRAIN asks DSP when it has played all > data of current track. I don't think that application _must_ drain when doing > gapless, only if it cares to know when DSP reaches the join between tracks. If we split this then I dont see why next write/meta_data call should wait. Then the PARTIAL_DRAIN should be used for knowing in userpsace that switch happened, right? > Possible problem - application calls NEXT_TRACK and PARTIAL_DRAIN when DSP has > already reached end of current track, so what does PARTIAL_DRAIN mean in this > case? It should just return immediately, and if your output is starved you are in error state, that can happen when you dont give data as well. > Also can we make metadata a key-pair list as Takashi suggested, with unlimited > length. Something like > > struct snd_compr_metadata_pair { > enum snd_compr_metadata_key key; > u32 value; > }; No enums... Thats not how kernel ABI is designed. I am leaning towards key/value pair, but is this fair assumption that value is always 32bit, what if we have larger single value, then should we split? > > struct snd_compr_metadata { > int count; /* number of actual entries in following array */ > struct snd_compr_metadata_pair *pairs; > }; > > Should we union the u32 value with a u8* buf to allow for any case where the > value needs to be a buffer of binary data? Is there any known case like this? Again nope, that aint good from ABI POV. What I think it should be is: enum { SNDRV_COMPRESS_ENCODER_PADDING = 1, SNDRV_COMPRESS_ENCODER_DELAY = 2, }; struct snd_compr_metadata { u32 key; u32 value; }; -- ~Vinod

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Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 07 Feb '13

07 Feb '13

On Thu, Feb 07, 2013 at 01:15:54AM +0000, Richard Fitzgerald wrote: > > +- partial drain > > +This is called when end of file is reached. The userspace can inform DSP that > > +EOF is reached and now DSP can start skipping padding delay. Also next write > > +data would belong to next track > > We're really doing two different tasks inside the "partial drain". The name, > and the reason we are doing two tasks in one, comes from a particular higher- > layer scenario, but there's no reason the driver API need use the same > terminology as one particular application of the functionality. > The two tasks are: > > 1) Tell the DSP that we have sent all data for the current track and following > data will be for the next track. This lets the DSP lay down a marker for where > it should strip padding at the end of a track, and know it should be expecting > more data for another track to follow gaplessly where it must strip the > encoder delay from the start. > > 2) Ask for notification when DSP has reached the changeover point between the > playback of the two tracks > > I think it would be more logical and less confusing not to combine the two into > a single ioctl. Instead, add only one new ioctl specifically to provide the > track changeover hint, something like SNDRV_COMPRESS_NEXT_TRACK (meaning DSP > should expect data for next track to follow). Don't add a new drain, just use > the existing SNDRV_COMPRESS_DRAIN - the driver/DSP can make a decision whether > it needs to do something special with the drain if we have told it that we > will be sending it some more data for a following track. > > So the SNDRV_COMPRESS_PARTIAL_DRAIN in this patch would become > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > 2) Send SNDRV_COMPRESS_DRAIN The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which expects the decoder to completely drain its buffers and come to complete halt. This would also mean the framework will treat a drained stream as stopped and needs a new start. Certainly we dont want that in this case. So we can't use SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that would overtly complicate this. If we are not doing proper drian lets not call it that. But I think I like the idea of splitting the two up to do a cleaner interface. Let me check this... > When we reach the final track we just do SNDRV_COMPRESS_DRAIN to wait for it > to finish. > > If we setup a next track by doing SNDRV_COMPRESS_NEXT_TRACK and then change > our mind and decide that this is going to be the last track, we do a > SNDRV_COMPRESS_DRAIN without sending any next track data, then we do > SNDRV_COMPRESS_STOP Nope that is where we would have issue. You dont call SNDRV_COMPRESS_STOP for compressed streams. You have to wait till DSP has decoded and rendered data for the last track which can be done by using SNDRV_COMPRESS_DRAIN only. SNDRV_COMPRESS_STOP should not be called in this case. -- ~Vinod

2 2

[alsa-devel] [PATCH] hda-analyzer: Make parsing code more robust against future proc file changes
by David Henningsson 07 Feb '13

07 Feb '13

Historically, new additions to the codec proc file has crashed hda-analyzer. With this patch, new additions will instead be ignored. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-analyzer/hda_proc.py | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/hda-analyzer/hda_proc.py b/hda-analyzer/hda_proc.py index 3d544c8..10aa662 100644 --- a/hda-analyzer/hda_proc.py +++ b/hda-analyzer/hda_proc.py @@ -644,9 +644,8 @@ class HDACodecProc(HDACodec, HDABaseProc): if line == 'Invalid AFG subtree': print "Invalid AFG subtree for codec %s?" % self.proc_codec_id return - while line.startswith('Power-Map: ') or \ - line.startswith('Analog Loopback: '): - print 'Sigmatel specific "%s" verb ignored for the moment' % line + while not line.startswith('Node '): + print 'Vendor specific "%s" verb ignored at the moment' % line idx += 1 line = lines[idx].strip() node = None @@ -728,7 +727,7 @@ class HDACodecProc(HDACodec, HDABaseProc): elif line.startswith(' Volume-Knob: '): node.add_volknob(line[15:]) else: - self.wrongfile(line) + print 'Node line "%s" ignored at the moment' % line idx += 1 if idx < len(lines): line = lines[idx] -- 1.7.9.5

1 0

[alsa-devel] [PATCH] hda-emu: Add sanify check of initial pinctl
by David Henningsson 07 Feb '13

07 Feb '13

The pinctl in alsa-info is not the initial pinctl but the current, so sanify the current pinctl before accepting it as the initial pinctl. (The initial pinctl is used during pm test.) Several false positives in the test tool is removed with this improvement. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-int.c | 64 +++++++++++++++++++++++++++++++++++---------------- hda-parse.c | 2 ++ include/hda-types.h | 4 ++++ 3 files changed, 50 insertions(+), 20 deletions(-) diff --git a/hda-int.c b/hda-int.c index c25af56..fbd3cee 100644 --- a/hda-int.c +++ b/hda-int.c @@ -400,29 +400,37 @@ static int get_channel_streamid(struct xhda_codec *codec, return node->streamid; } -static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, - unsigned int cmd) +void hda_verify_pin_ctl(struct xhda_node *node, int log, + unsigned int *sanified_pinctl) { - if (!node) - return 0; - node->pinctl = cmd & 0xff; + int pinctl = node->pinctl; /* sanity checks */ if ((node->pinctl & AC_PINCTL_OUT_EN) && - !(node->pincap & AC_PINCAP_OUT)) - hda_log(HDA_LOG_ERR, - "setting OUT_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_OUT)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting OUT_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_OUT_EN; + } if ((node->pinctl & AC_PINCTL_HP_EN) && - !(node->pincap & AC_PINCAP_HP_DRV)) - hda_log(HDA_LOG_ERR, - "setting HP_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_HP_DRV)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting HP_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_HP_EN; + } if ((node->pinctl & AC_PINCTL_IN_EN) && - !(node->pincap & AC_PINCAP_IN)) - hda_log(HDA_LOG_ERR, - "setting IN_EN to pin 0x%x without caps\n", - node->nid); + !(node->pincap & AC_PINCAP_IN)) { + if (log) + hda_log(HDA_LOG_ERR, + "setting IN_EN to pin 0x%x without caps\n", + node->nid); + pinctl &= ~AC_PINCTL_IN_EN; + } + if (node->pinctl & AC_PINCTL_IN_EN) { unsigned int cap_set, cap_check; const char *vref; @@ -455,11 +463,27 @@ static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, if (!cap_set) cap_set = AC_PINCAP_VREF_HIZ; if (cap_check && !(cap_set & cap_check)) { - hda_log(HDA_LOG_ERR, - "setting VREF %s to pin 0x%x without caps 0x%x\n", - vref, node->nid, node->pincap); + if (log) + hda_log(HDA_LOG_ERR, + "setting VREF %s to pin 0x%x without caps 0x%x\n", + vref, node->nid, node->pincap); + pinctl &= ~AC_PINCTL_VREFEN; } } + + if (sanified_pinctl) + *sanified_pinctl = pinctl; +} + +static int set_pin_ctl(struct xhda_codec *codec, struct xhda_node *node, + unsigned int cmd) +{ + if (!node) + return 0; + node->pinctl = cmd & 0xff; + + hda_verify_pin_ctl(node, 1, NULL); + return 0; } diff --git a/hda-parse.c b/hda-parse.c index c75b03b..d38bc04 100644 --- a/hda-parse.c +++ b/hda-parse.c @@ -240,6 +240,8 @@ static int parse_node_items(const char *buf) &node->orig_amp_out_vals); } else if ((p = strmatch(head, "Pin-ctls: "))) { node->pinctl = strtoul(p, NULL, 0); + if (node->pincap) + hda_verify_pin_ctl(node, 0, &node->pinctl); node->orig_pinctl = node->pinctl; } else if ((p = strmatch(head, "Pincap "))) { const char *end; diff --git a/include/hda-types.h b/include/hda-types.h index 3754bca..68cfaa8 100644 --- a/include/hda-types.h +++ b/include/hda-types.h @@ -146,6 +146,10 @@ void hda_free_route_lists(struct xhda_route_list *list); void hda_show_routes(int nid, unsigned int flags); +void hda_verify_pin_ctl(struct xhda_node *node, int log, + unsigned int *sanified_pinctl); + + void *xalloc(size_t size); #endif /* __HDA_TYPES_H */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: compress: Only mute playback streams
by Mark Brown 07 Feb '13

07 Feb '13

Otherwise capture activity on a compressed DAI would mute any playback on the same DAI. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index c81aeec..35726cb 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,13 +116,12 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; + snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } - snd_soc_dai_digital_mute(codec_dai, 1); - cpu_dai->active--; codec_dai->active--; codec->active--; @@ -179,10 +178,16 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cmd == SNDRV_PCM_TRIGGER_START) - snd_soc_dai_digital_mute(codec_dai, 0); - else if (cmd == SNDRV_PCM_TRIGGER_STOP) - snd_soc_dai_digital_mute(codec_dai, 1); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1); + break; + } + } out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: core: Allow digital mute for capture
by Mark Brown 07 Feb '13

07 Feb '13

Help avoid noise from the power up of the capture path propagating through into the start of the recording (especially noise caused by the ramp of microphone biases) by keeping the capture muted until after we've finished powering things up with DAPM in the same manner we do for playback. This allows us to take advantage of soft mute support in the hardware more effectively and is more consistent. The core code using the existing digital mute operation is updated to take advantage of this. Some additional cases in the soc-pcm code and suspend will need separate handling but these are less practically relevant than the main runtime stream start/stop case. Rather than refactor the digital mute function in every single driver a new operation is added for drivers taking advantage of this functionality, the old operation should be phased out over time. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 3 ++- sound/soc/soc-compress.c | 19 +++++++++---------- sound/soc/soc-core.c | 12 ++++++++++-- sound/soc/soc-dapm.c | 6 ++++-- sound/soc/soc-pcm.c | 7 +++---- 5 files changed, 28 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 8bcdd71..6afdaf6 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -126,7 +126,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction); struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 35726cb..b5b3db7 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,12 +116,13 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (cstream->direction == SND_COMPRESS_PLAYBACK) { cpu_dai->playback_active--; codec_dai->playback_active--; - snd_soc_dai_digital_mute(codec_dai, 1); } else { cpu_dai->capture_active--; codec_dai->capture_active--; } + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + cpu_dai->active--; codec_dai->active--; codec->active--; @@ -178,15 +179,13 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) goto out; } - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - snd_soc_dai_digital_mute(codec_dai, 0); - break; - case SNDRV_PCM_TRIGGER_STOP: - snd_soc_dai_digital_mute(codec_dai, 1); - break; - } + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_soc_dai_digital_mute(codec_dai, 0, cstream->direction); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + break; } out: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcee552..1283d86 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3547,12 +3547,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); * snd_soc_dai_digital_mute - configure DAI system or master clock. * @dai: DAI * @mute: mute enable + * @direction: stream to mute * * Mutes the DAI DAC. */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) { - if (dai->driver && dai->driver->ops->digital_mute) + if (!dai->driver) + return -ENOTSUPP; + + if (dai->driver->ops->mute_stream) + return dai->driver->ops->mute_stream(dai, mute, direction); + else if (direction == SNDRV_PCM_STREAM_PLAYBACK && + dai->driver->ops->digital_mute) return dai->driver->ops->digital_mute(dai, mute); else return -ENOTSUPP; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 17dba18..1dc5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3255,14 +3255,16 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0); + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); ret = 0; break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1); + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); if (ret != 0 && ret != -ENOTSUPP) dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); ret = 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 786c2e7..a8d72a9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -383,8 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -488,7 +487,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0, substream->stream); out: mutex_unlock(&rtd->pcm_mutex); @@ -586,7 +585,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ if (!codec->active) - snd_soc_dai_digital_mute(codec_dai, 1); + snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) -- 1.7.10.4

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Re: [alsa-devel] [PATCH 001/002] ASoC: Replace max98090 Device Driver
by Mark Brown 07 Feb '13

07 Feb '13

On Wed, Feb 06, 2013 at 11:02:33AM -0800, Jerry Wong wrote: > From: Jerry Wong <jerry.wong(a)maximintegrated.com> > > This patch removes the existing max98090 driver prior to installing a more > complete one. Applied both, thanks.

1 0

Re: [alsa-devel] [RFC] compress: add support for gapless playback
by Takashi Iwai 07 Feb '13

07 Feb '13

At Thu, 7 Feb 2013 11:37:38 +0000, Richard Fitzgerald wrote: > > > > 1) Send SNDRV_COMPRESS_NEXT_TRACK > > > 2) Send SNDRV_COMPRESS_DRAIN > > The problem would be in that case the defination of SNDRV_COMPRESS_DRAIN which > > expects the decoder to completely drain its buffers and come to complete halt. > > This would also mean the framework will treat a drained stream as stopped and > > needs a new start. Certainly we dont want that in this case. So we can't use > > SNDRV_COMPRESS_DRAIN to indicate. Yes we can put conditional check but IMO that > > would overtly complicate this. If we are not doing proper drian lets not > > call it that. > > Ok, so let's keep the partial drain but split out the NEXT_TRACK hint so we keep > the two operations separate. It's clearer to understand. NEXT_TRACK tells DSP to > prepare for data for new track, PARTIAL_DRAIN asks DSP when it has played all > data of current track. I don't think that application _must_ drain when doing > gapless, only if it cares to know when DSP reaches the join between tracks. > Possible problem - application calls NEXT_TRACK and PARTIAL_DRAIN when DSP has > already reached end of current track, so what does PARTIAL_DRAIN mean in this > case? Which track is it draining? Maybe define that it always refers to the > track _before_ the most recent NEXT_TRACK call? > > Also can we make metadata a key-pair list as Takashi suggested, with unlimited > length. Something like > > struct snd_compr_metadata_pair { > enum snd_compr_metadata_key key; > u32 value; > }; > > struct snd_compr_metadata { > int count; /* number of actual entries in following array */ > struct snd_compr_metadata_pair *pairs; > }; > > Should we union the u32 value with a u8* buf to allow for any case where the > value needs to be a buffer of binary data? Oh no, don't do it. Never. In general, - Never put types that depending on the architecture in struct passed to ioctls (e.g. long, pointer) - Never use bit fields for ABI - If 64bit value is used, think of alignment; safer to add packed attribute So, in general, use only u32, s16, or whatever, the types with explicit size. Takashi

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Re: [alsa-devel] mmotm 2013-02-06-17-13 uploaded (alsa)
by Takashi Iwai 07 Feb '13

07 Feb '13

At Wed, 06 Feb 2013 21:59:37 -0800, Randy Dunlap wrote: > > On 02/06/13 17:14, akpm(a)linux-foundation.org wrote: > > The mm-of-the-moment snapshot 2013-02-06-17-13 has been uploaded to > > > > http://www.ozlabs.org/~akpm/mmotm/ > > > > mmotm-readme.txt says > > > > README for mm-of-the-moment: > > > > http://www.ozlabs.org/~akpm/mmotm/ > > > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > > more than once a week. > > > > on i386: (from linux-next.patch) > > > CC [M] sound/pci/hda/patch_via.o > sound/pci/hda/patch_via.c: In function 'is_aa_path_mute': > sound/pci/hda/patch_via.c:360:27: error: 'struct hda_gen_spec' has no member named 'num_loopbacks' > sound/pci/hda/patch_via.c:361:17: error: 'struct hda_gen_spec' has no member named 'loopback_list' > make[4]: *** [sound/pci/hda/patch_via.o] Error 1 > > > > Full randconfig file is attached. Oh, it's again an unsatisfied CONFIG_PM ifdef. Fixed now. Thanks! Takashi

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[alsa-devel] [patch] ALSA: ice1712: fix boundary check in snd_wm8766_write()
by Dan Carpenter 07 Feb '13

07 Feb '13

The wm->regs[] array has WM8766_REG_COUNT (16) elements not WM8766_REG_RESET (31). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This is a static checker thing, I can't test it. diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index 8072ade..e473f8a 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -31,7 +31,7 @@ static void snd_wm8766_write(struct snd_wm8766 *wm, u16 addr, u16 data) { - if (addr < WM8766_REG_RESET) + if (addr < WM8766_REG_COUNT) wm->regs[addr] = data; wm->ops.write(wm, addr, data); }

2 1

[alsa-devel] [PATCH] ASoC: arizona: Fix debug logging level for FLLs and AIFs
by Mark Brown 06 Feb '13

06 Feb '13

Use _dbg for debug messages. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ba3bd03..5e77a60 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -57,14 +57,14 @@ #define arizona_fll_warn(_fll, fmt, ...) \ dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_dbg(_fll, fmt, ...) \ - dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + dev_dbg(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_aif_err(_dai, fmt, ...) \ dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_warn(_dai, fmt, ...) \ dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) #define arizona_aif_dbg(_dai, fmt, ...) \ - dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, -- 1.7.10.4

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[alsa-devel] PandaBoard ES Audio Problems
by Brent Weatherall 06 Feb '13

06 Feb '13

Hello. While developing with OpenMax and ALSA on the PandaBoard ES (OMAP 4460 armhf), I ran into a strange problem. At a high level, when configuring the built in audio device (3.5 mm jack), the device accepts ANY rate requested (via snd_pcm_set_rate_near). it accepts unsupported rates without alteration. If the rate is actually not supported, then 'snd_pcm_hw_params' fails with error '-22'. After this happens, it is not possible to reconfigure the rate without closing the device and starting over to completely re-configure from the start (calling snd_pcm_open, etc...). It is not possible to simply try another rate. Technical details below: When 'snd_pcm_set_rate_near' is called, during the min and max rate probing, the code reaches interval_inline.h:71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) During this call, line '74 return i->min;' executes. This function ALWAYS returns the same rate value requested in 'snd_pcm_set_rate_near' as the minimum possible rate. If I request a rate near 44100, the minimum rates comes back as 44100. If I request a rate near 8000, the minimum is returned as 8000 and so forth. To recap, the 'snd_pcm_set_rate_near' call ALWAYS succeeds without altering the rate passed to it the first time. The only way to find out if the rate is actually supported is to call 'snd_pcm_hw_params' and then see if that fails with err == -22 (invalid argument). Furthermore, once the failure occurs, another rate value cannot be tried unless the PCM device is closed and the entire configuration process is repeated. After the failure, subsequent calls to 'snd_pcm_set_rate_near' with ANY different value always get set back to the original bad value. Example, I request 44100 and it fails. So I try 8000 and it returns 44100 (and of course fails again if I try 'snd_pcm_hw_params'). I am not sure yet, but is the value at 'interval_inline.h:74 return i->min;' determined directly by the hardware? It seems like it is, but I don't know the code well enough yet to make that assertion. I also haven't yet examined the 'snd_interval_max' command, but my gut tells me it is behaving the same way, since after a bad rate is set, any higher or lower rate setting attempts ALWAYS come back as the original rate. How would i determine if it is hardware or maybe something bad with the hardware.software interface? I checked the PCM state value and it was correct the entire time. Here are short snippets of two gdb sessions where I found the issue and below that is test program output showing the state values as I try to configure two different rates: Code snippet in ALSA lib file interval_inline.h: 71 INTERVAL_INLINE int snd_interval_min(const snd_interval_t *i) 72 { 73 assert(!snd_interval_empty(i)); 74 return i->min; 75 } Run #1 with 44100 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $6 = 44100 Run #2 with 8000 'snd_pcm_set_rate_near' call: (gdb) s 74 return i->min; (gdb) print i->min $8 = 8000 Here is a log of the high level test program: snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_channels succeeded. cannot set buffer time (Invalid argument) cannot set buffer period time (Invalid argument) Before commit, state is: SND_PCM_STATE_OPEN cannot set parameters (Invalid argument) After failure, state is: SND_PCM_STATE_OPEN Trying to re-configure rate param. Before calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN cannot drop (Input/output error) After calling 'snd_pcm_drop', state is: SND_PCM_STATE_OPEN Trying rate 8000. snd_pcm_hw_params_set_rate_near succeeded. Rate set to 44100. snd_pcm_hw_params_set_buffer_time_near succeeded. Buffer size actually set to 22050. cannot set buffer period time (Invalid argument) cannot set parameters (Invalid argument) cannot prepare audio interface for use (Input/output error) Rate 44100 supported: FAIL Please let me know your thoughts on this and if there is a good way to prove if this is a hardware error or a software error. I am running 'Linux version 3.4.0-1487-omap4' and using the standard repositories with apt-get to install ALSA. I used 'apt-get source' to obtain 'alsa-lib-1.0.25' source files and I am running 'libasound2-dbg' to enable GDB debugging. Regards, Brent

6 17

[alsa-devel] [RFC] compress: add support for gapless playback
by Vinod Koul 06 Feb '13

06 Feb '13

From: Jeeja KP <jeeja.kp(a)intel.com> this add new API for sound compress to support gapless playback. As noted in Documentation change, we add API to send metadata of encoder and padding delay to DSP. Also add API for indicating EOF and switching to subsequent track Also bump the compress API version Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Documentation/sound/alsa/compress_offload.txt | 22 +++++++++++ include/sound/compress_driver.h | 2 + include/uapi/sound/compress_offload.h | 18 ++++++++- sound/core/compress_offload.c | 51 +++++++++++++++++++++++++ 4 files changed, 92 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 90e9b3a..071c4b6 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -145,6 +145,28 @@ Modifications include: - Addition of encoding options when required (derived from OpenMAX IL) - Addition of rateControlSupported (missing in OpenMAX AL) +Gapless Playback +================ +When playing thru an album, the decoders have the ability to skip the encoder +delay and padding and directly move from one track content to another. The end +user can perceive this as gapless playback as we dont have silence while +switching from one track to another + +The decoder needs to know the encoder delay and encoder padding. So we need to +pass this to DSP. Also DSP and userspace needs to switch from one track to +another and start using data for second track. + +The main additions are: + +- set_metadata +This routine sets the encoder delay and encoder padding. This can be used by +decoder to strip the silence + +- partial drain +This is called when end of file is reached. The userspace can inform DSP that +EOF is reached and now DSP can start skipping padding delay. Also next write +data would belong to next track + Not supported: - Support for VoIP/circuit-switched calls is not the target of this diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f2912ab..42d6d7c 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -110,6 +110,8 @@ struct snd_compr_ops { struct snd_compr_params *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_metadata)(struct snd_compr_stream *stream, + struct snd_compr_metadata *metadata); int (*trigger)(struct snd_compr_stream *stream, int cmd); int (*pointer)(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 05341a4..65ca2f1 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -30,7 +30,7 @@ #include <sound/compress_params.h> -#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0) +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1) /** * struct snd_compressed_buffer: compressed buffer * @fragment_size: size of buffer fragment in bytes @@ -122,6 +122,18 @@ struct snd_compr_codec_caps { }; /** + * struct snd_compr_metadata: compressed stream metadata + * @encoder_delay: no of samples inserted by the encoder at the beginning + * of the track + * @encoder_padding: no of samples appended by the encoder at the end + * of the track + */ +struct snd_compr_metadata { + __u32 encoder_delay; + __u32 encoder_padding; +}; + +/** * compress path ioctl definitions * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec @@ -145,6 +157,8 @@ struct snd_compr_codec_caps { struct snd_compr_codec_caps) #define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) #define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) @@ -152,10 +166,12 @@ struct snd_compr_codec_caps { #define SNDRV_COMPRESS_START _IO('C', 0x32) #define SNDRV_COMPRESS_STOP _IO('C', 0x33) #define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x35) /* * TODO * 1. add mmap support * */ #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 8 #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2928971 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -514,6 +514,37 @@ out: return retval; } +static int +snd_compr_set_metadata(struct snd_compr_stream *stream, unsigned long arg) +{ + struct snd_compr_metadata *metadata; + int retval; + + if (!stream->ops->set_metadata) + return -ENXIO; + /* + * we should allow parameter change only when stream has been + * opened not in other cases + */ + metadata = kmalloc(sizeof(*metadata), GFP_KERNEL); + if (!metadata) + return -ENOMEM; + if (copy_from_user(metadata, (void __user *)arg, + sizeof(*metadata))) { + retval = -EFAULT; + goto out; + } + + pr_debug("metadata encoder delay=%x encoder padding=%x\n", + metadata->encoder_delay, metadata->encoder_padding); + + retval = stream->ops->set_metadata(stream, metadata); + +out: + kfree(metadata); + return retval; +} + static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { @@ -594,6 +625,20 @@ static int snd_compr_drain(struct snd_compr_stream *stream) return retval; } +static int snd_compr_partial_drain(struct snd_compr_stream *stream) +{ + int retval; + if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || + stream->runtime->state == SNDRV_PCM_STATE_SETUP) + return -EPERM; + retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) + pr_err("Partial drain returned failure\n"); + else + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + return retval; +} + static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) { struct snd_compr_file *data = f->private_data; @@ -623,6 +668,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_METADATA): + retval = snd_compr_set_metadata(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_TSTAMP): retval = snd_compr_tstamp(stream, arg); break; @@ -644,6 +692,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_DRAIN): retval = snd_compr_drain(stream); break; + case _IOC_NR(SNDRV_COMPRESS_PARTIAL_DRAIN): + retval = snd_compr_partial_drain(stream); + break; } mutex_unlock(&stream->device->lock); return retval; -- 1.7.0.4

4 13

[alsa-devel] ASoC updates for v3.8
by Mark Brown 06 Feb '13

06 Feb '13

The following changes since commit 949db153b6466c6f7cad5a427ecea94985927311: Linux 3.8-rc5 (2013-01-25 11:57:28 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.8-rc6 for you to fetch changes up to 5074e8eca643dd19e5397e638c7e9ca0ba9b4e24: Merge remote-tracking branch 'asoc/fix/wm2200' into tmp (2013-02-06 10:54:35 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.8-rc7 I've been a bit slack sending the pull requests for this so there's rather more driver specific updates here than is desirable but everything here is nicely localised so hopefully we're OK. ---------------------------------------------------------------- Alexey Galakhov (1): ASoC: samsung: Fix compilation error on S3C2440 Benoît Thébaudeau (1): ASoC: tlv320aic3x: Remove mono support Charles Keepax (1): ASoC: wm_adsp: Release firmware on error Chris Rattray (1): ASoC: wm2200: correct IN2L and IN3L digital mute Dan Carpenter (1): ASoC: dwc: fix support for more than two channels Lee Jones (1): ASoC: Ux500: Fix build error pertaining to missing include file Mark Brown (13): ASoC: wm_adsp: Ensure ADSP2 DMAs are quiesced when DSP is halted ASoC: wm2000: Lock state machine updates Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/davinci' into tmp Merge remote-tracking branch 'asoc/fix/dwc' into tmp Merge remote-tracking branch 'asoc/fix/fsl' into tmp Merge remote-tracking branch 'asoc/fix/lrg' into tmp Merge remote-tracking branch 'asoc/fix/samsung' into tmp Merge remote-tracking branch 'asoc/fix/tlv320aic3x' into tmp Merge remote-tracking branch 'asoc/fix/ux500' into tmp Merge remote-tracking branch 'asoc/fix/wm2000' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Michal Bachraty (1): ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback Ryo Tsutsui (1): ASoC: arizona: Fixed a bug in FLL fractional calculation Shawn Guo (2): Revert "ASoC: fsl: fix multiple definition of init_module" ASoC: fsl: fix snd-soc-imx-pcm module build MAINTAINERS | 2 +- sound/soc/codecs/arizona.c | 16 ++++++++++++++-- sound/soc/codecs/tlv320aic3x.c | 4 ++-- sound/soc/codecs/wm2000.c | 31 ++++++++++++++++++++++++++++--- sound/soc/codecs/wm2200.c | 7 ++----- sound/soc/codecs/wm5102.c | 3 +-- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm_adsp.c | 20 ++++++++++++++------ sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/dwc/designware_i2s.c | 4 ++++ sound/soc/fsl/imx-pcm-dma.c | 21 +-------------------- sound/soc/fsl/imx-pcm-fiq.c | 22 +--------------------- sound/soc/fsl/imx-pcm.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/fsl/imx-pcm.h | 18 ++++++++++++++++++ sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/soc-dapm.c | 12 ++++++++++-- sound/soc/ux500/mop500.c | 2 +- 17 files changed, 132 insertions(+), 69 deletions(-)

2 5

[alsa-devel] hda_analyzer failed
by Raymond Yau 06 Feb '13

06 Feb '13

>> >> > File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile >> > raise ValueError, "wrong proc file format (%s)" % msg >> > ValueError: wrong proc file format (integer decode 'State of AFG node >> 0x01:' ('Node ')) >> > >> > Any ideas? >> >> This patch add lines which make the parser of hda_proc.py fail >> >> http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=f4… >> > > Sigh, this is not the first time new stuff in the proc file crashes hda_analyzer. > > Jaroslav, can you please fix hda_analyzer so that it ignores new stuff in the proc file instead of crashing, so we don't have this problem over and over again? > I don`t understand why the parser have to ignore new stuff BTW, why hda_proc query the connection list of vendor defined widget ? http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=blob;f=codecs… Node 0x23 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: -22

3 3

Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 06 Feb '13

06 Feb '13

At Wed, 6 Feb 2013 08:47:41 +0800, Daniel J Blueman wrote: > > On 6 February 2013 01:40, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 05 Feb 2013 17:34:15 +0100, > > Takashi Iwai wrote: > >> > >> At Wed, 6 Feb 2013 00:29:54 +0800, > >> Daniel J Blueman wrote: > >> > > >> > On 6 February 2013 00:16, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > > At Wed, 6 Feb 2013 00:10:30 +0800, > >> > > Daniel J Blueman wrote: > >> > >> > >> > >> Hi Takashi, > >> > >> > >> > >> The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > >> > >> arrangement on my Macbook Pro 10,1 - only the higher-frequency > >> > >> speakers work despite the front and rear channels being exposed in the > >> > >> mixer. > >> > > > >> > > Interesting. So you have a machine with 4.0 built-in speaker instead > >> > > of 2.1? Then we need to add a device-specific flag for it. Currently > >> > > the driver assumes 2.1 system blindly because majority of machines > >> > > have that. > >> > > > >> > > FWIW, the codec parser code has been totally rewritten for 3.9, so any > >> > > patch to 3.8 won't be applied to 3.9 (and vice versa)... > >> > > > >> > > Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to > >> > > cook it for 3.9 (and maybe backport to 3.8). > >> > > >> > Here's the output from the current alsa-info.sh on 3.8-rc6 with the > >> > two cited patches reverted [1]; let me know if you'd like 3.8-rc6 > >> > pure. > >> > >> Thanks. > >> > >> > The sound is pretty impressive for a laptop when the low-frequecy > >> > speakers are enabled. > >> > >> Which program are you using for testing the surrounds? > >> I'm interested in it because the commit you reverted is basically > >> providing only an additional information for the channel map, and it > >> doesn't change anything else. It implies that some applications are > >> really referring to the chmap info. > > > > Or, it might be that the mixer value is simply not set correct. > > > > To be sure, could you try again 3.8-rc6 without reversing patches, > > adjust "Speaker" and "Bass Speaker" volumes properly, and retest? > > If it still doesn't work, please take alsa-info.sh snapshot at this > > state for comparing with the previous result. > > My apologies! We do now have "Bass Speaker" which affects both bass > speakers. It was always being restored to level 0 and works when set > up. OK, good to know. > There is a "Subwoofer" slider in addition to "Balance" and "Fade" in > the GNOME mixer UI which is greyed out; presumably this is intended as > the same mixer control? I have no idea about GNOME stuff... David, do you know of this? Takashi

2 1

[alsa-devel] [PATCH 00/15] ARM build regressions in v3.8
by Arnd Bergmann 06 Feb '13

06 Feb '13

I know this comes late, but we have a number of broken configurations in ARM in v3.8 that were still building in v3.7, and I'd like to get them all fixed in the final 3.8 release. It would be nice if the respective maintainers could have a look at these patches and apply them directly when they are happy with them. The first patch in the series is strictly speaking not a build error but just a warning, but it is a particularly annoying one that came in through the latest binutils release rather than a kernel change. The same binutils update also broke the samsung and w90x900 platforms. A few of the other changes are the result of the imx multiplatform conversion. I'm not really fixing those here, just picking up the pieces. It would be much nicer if we could actually get those drivers to work again with CONFIG_MULTIPLATFORM enabled rather than just disabling them, but it may be much too late for that. At least the drivers don't seem to be too essential, as they are only built in allyesconfig but not in any of the defconfigs. Arnd Arnd Bergmann (15): ARM: compressed/head.S: work around new binutils warning ARM: mvebu: build coherency_ll.S for arch=armv7-a ARM: samsung: fix assembly syntax for new gas ARM: w90x900: fix legacy assembly syntax ASoC: fsl: fiq and dma cannot both be modules clk: export __clk_get_name drm/exynos: don't include plat/gpio-cfg.h drm/exynos: fimd and ipp are broken on multiplatform media: coda: don't build on multiplatform mfd/vexpress: export vexpress_config_func_{put,get} mtd: davinci_nand: fix OF support USB: gadget/freescale: disable non-multiplatform drivers USB: ehci: make orion and mxc bus glues coexist samples/seccomp: be less stupid about cross compiling staging/omapdrm: don't build on multiplatform arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/compressed/head.S | 12 ++++++++++++ arch/arm/mach-mvebu/coherency_ll.S | 1 + arch/arm/mach-s3c24xx/include/mach/debug-macro.S | 12 ++++++------ arch/arm/mach-s3c24xx/include/mach/entry-macro.S | 4 ++-- arch/arm/mach-s3c24xx/pm-h1940.S | 2 +- arch/arm/mach-s3c24xx/sleep-s3c2410.S | 12 ++++++------ arch/arm/mach-s3c24xx/sleep-s3c2412.S | 12 ++++++------ arch/arm/mach-w90x900/include/mach/entry-macro.S | 4 ++-- arch/arm/plat-samsung/include/plat/debug-macro.S | 18 +++++++++--------- drivers/clk/clk.c | 1 + drivers/gpu/drm/exynos/Kconfig | 4 ++-- drivers/gpu/drm/exynos/exynos_hdmi.c | 1 - drivers/media/platform/Kconfig | 2 +- drivers/mfd/vexpress-config.c | 3 ++- drivers/mtd/nand/davinci_nand.c | 2 +- drivers/staging/omapdrm/Kconfig | 2 +- drivers/usb/gadget/Kconfig | 3 ++- drivers/usb/host/ehci-hcd.c | 16 +++++++++++++++- samples/seccomp/Makefile | 2 ++ sound/soc/fsl/Kconfig | 3 +++ 21 files changed, 76 insertions(+), 42 deletions(-) -- 1.7.10.4 Cc: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: David Airlie <airlied(a)linux.ie> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: James Morris <james.l.morris(a)oracle.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> Cc: Mike Turquette <mturquette(a)linaro.org> Cc: Rob Clark <rob(a)ti.com> Cc: Russell King <rmk+kernel(a)arm.linux.org.uk> Cc: Shawn Guo <shawn.guo(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-media(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org

3 8

Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 05 Feb '13

05 Feb '13

At Wed, 6 Feb 2013 00:29:54 +0800, Daniel J Blueman wrote: > > On 6 February 2013 00:16, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 6 Feb 2013 00:10:30 +0800, > > Daniel J Blueman wrote: > >> > >> Hi Takashi, > >> > >> The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > >> arrangement on my Macbook Pro 10,1 - only the higher-frequency > >> speakers work despite the front and rear channels being exposed in the > >> mixer. > > > > Interesting. So you have a machine with 4.0 built-in speaker instead > > of 2.1? Then we need to add a device-specific flag for it. Currently > > the driver assumes 2.1 system blindly because majority of machines > > have that. > > > > FWIW, the codec parser code has been totally rewritten for 3.9, so any > > patch to 3.8 won't be applied to 3.9 (and vice versa)... > > > > Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to > > cook it for 3.9 (and maybe backport to 3.8). > > Here's the output from the current alsa-info.sh on 3.8-rc6 with the > two cited patches reverted [1]; let me know if you'd like 3.8-rc6 > pure. Thanks. > The sound is pretty impressive for a laptop when the low-frequecy > speakers are enabled. Which program are you using for testing the surrounds? I'm interested in it because the commit you reverted is basically providing only an additional information for the channel map, and it doesn't change anything else. It implies that some applications are really referring to the chmap info. Takashi

1 1

Re: [alsa-devel] 3.8-rc1 patch_cirrus 4.0 regression...
by Takashi Iwai 05 Feb '13

05 Feb '13

At Wed, 6 Feb 2013 00:10:30 +0800, Daniel J Blueman wrote: > > Hi Takashi, > > The v3.8-rc kernels have regressed from v3.7 with the quad-speaker > arrangement on my Macbook Pro 10,1 - only the higher-frequency > speakers work despite the front and rear channels being exposed in the > mixer. Interesting. So you have a machine with 4.0 built-in speaker instead of 2.1? Then we need to add a device-specific flag for it. Currently the driver assumes 2.1 system blindly because majority of machines have that. FWIW, the codec parser code has been totally rewritten for 3.9, so any patch to 3.8 won't be applied to 3.9 (and vice versa)... Could you give alsa-info.sh output on 3.8 kernel? Then I'll try to cook it for 3.9 (and maybe backport to 3.8). Takashi > > Reverting f37bc7 [1] restored the correct behaviour ([2] was reverted > to compile, but the problem still occurs with only [2] reverted). > > How can I help to debug this and find the right approach? > > Thanks, > Daniel > > --- [1] > > commit f37bc7a88d374448a1f4bba9267d308606d78bf2 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Nov 8 15:59:23 2012 +0100 > > ALSA: hda - Give standard "Bass Speaker" mixer for 2.1 speakers > > When two built-in speakers are found on the machine, we can suppose > it's rather a 2.1 speaker system with a bass output instead of > front/surround channels. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- [2] > > commit ee81abb623cb5e03c182d16871bb4fb34fdc9b4f > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Nov 8 17:12:10 2012 +0100 > > ALSA: hda - Apply a proper chmap for built-in 2.1 speakers > > When 2.1 speakers are detected, use the corresponding channel map > instead of the standard map with front+rear surrounds. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > -- > Daniel J Blueman >

1 1

[alsa-devel] [PATCH] ASoC: wm2200: Provide platform data for MICBIAS configuration
by Chris Rattray 05 Feb '13

05 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- include/sound/wm2200.h | 22 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55b..bc7ab1a 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 1, + WM2200_MBIAS_LVL_1V8 = 2, + WM2200_MBIAS_LVL_1V9 = 3, + WM2200_MBIAS_LVL_2V0 = 4, + WM2200_MBIAS_LVL_2V2 = 5, + WM2200_MBIAS_LVL_2V4 = 6, + WM2200_MBIAS_LVL_2V5 = 7, + WM2200_MBIAS_LVL_2V6 = 8, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..bbd8383 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2192,6 +2192,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2342,6 +2343,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = (wm2200->pdata.micbias[i].mb_lvl -1) + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm2200: Provide platform data for MICBIAS configuration
by Chris Rattray 05 Feb '13

05 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Change-Id: I65da7c50baf7134d20e0bfeecc601d82828a8de1 --- include/sound/wm2200.h | 22 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55b..bc7ab1a 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 1, + WM2200_MBIAS_LVL_1V8 = 2, + WM2200_MBIAS_LVL_1V9 = 3, + WM2200_MBIAS_LVL_2V0 = 4, + WM2200_MBIAS_LVL_2V2 = 5, + WM2200_MBIAS_LVL_2V4 = 6, + WM2200_MBIAS_LVL_2V5 = 7, + WM2200_MBIAS_LVL_2V6 = 8, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..bbd8383 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2192,6 +2192,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2342,6 +2343,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = (wm2200->pdata.micbias[i].mb_lvl -1) + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- 1.7.2.5

2 1

[alsa-devel] alsa wiki spam
by Eliot Blennerhassett 05 Feb '13

05 Feb '13

Hundreds of spam bot registrations dominate the recent changes: http://www.alsa-project.org/main/index.php?title=Special:RecentChanges&limi… no sign so far of spam content, but it is a bit of a worry.

2 1

[alsa-devel] [PATCH 001/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 05 Feb '13

05 Feb '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch removes the existing max98090 driver prior to installing a more complete one. --- The original author is okay to replace his recently submitted driver with this one. "... Would you mind if we began this now by replacing the existing driver with the one attached? ..." - Jerry "... But of course I agree to replace it to your code. Please send it to upstream... " - Morimoto-san sound/soc/codecs/Kconfig | 4 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/max98090.c | 577 ------------------------------------------- 3 files changed, 0 insertions(+), 583 deletions(-) delete mode 100644 sound/soc/codecs/max98090.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a84782..0e368d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,7 +44,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C - select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -268,9 +267,6 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate -config SND_SOC_MAX98090 - tristate - config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36..aa56312 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,7 +34,6 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o -snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -158,7 +157,6 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o -obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c deleted file mode 100644 index c9772ca..0000000 --- a/sound/soc/codecs/max98090.c +++ /dev/null @@ -1,577 +0,0 @@ -/* - * max98090.c -- MAX98090 ALSA SoC Audio driver - * based on Rev0p8 datasheet - * - * Copyright (C) 2012 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * Based on - * - * max98095.c - * Copyright 2011 Maxim Integrated Products - * - * https://github.com/hardkernel/linux/commit/\ - * 3417d7166b17113b3b33b0a337c74d1c7cc313df#sound/soc/codecs/max98090.c - * Copyright 2011 Maxim Integrated Products - * - * 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/i2c.h> -#include <linux/module.h> -#include <linux/regmap.h> -#include <sound/soc.h> -#include <sound/tlv.h> - -/* - * - * MAX98090 Registers Definition - * - */ - -/* RESET / STATUS / INTERRUPT REGISTERS */ -#define MAX98090_0x00_SW_RESET 0x00 -#define MAX98090_0x01_INT_STS 0x01 -#define MAX98090_0x02_JACK_STS 0x02 -#define MAX98090_0x03_INT_MASK 0x03 - -/* QUICK SETUP REGISTERS */ -#define MAX98090_0x04_SYS_CLK 0x04 -#define MAX98090_0x05_SAMPLE_RATE 0x05 -#define MAX98090_0x06_DAI_IF 0x06 -#define MAX98090_0x07_DAC_PATH 0x07 -#define MAX98090_0x08_MIC_TO_ADC 0x08 -#define MAX98090_0x09_LINE_TO_ADC 0x09 -#define MAX98090_0x0A_ANALOG_MIC_LOOP 0x0A -#define MAX98090_0x0B_ANALOG_LINE_LOOP 0x0B - -/* ANALOG INPUT CONFIGURATION REGISTERS */ -#define MAX98090_0x0D_INPUT_CONFIG 0x0D -#define MAX98090_0x0E_LINE_IN_LVL 0x0E -#define MAX98090_0x0F_LINI_IN_CFG 0x0F -#define MAX98090_0x10_MIC1_IN_LVL 0x10 -#define MAX98090_0x11_MIC2_IN_LVL 0x11 - -/* MICROPHONE CONFIGURATION REGISTERS */ -#define MAX98090_0x12_MIC_BIAS_VOL 0x12 -#define MAX98090_0x13_DIGITAL_MIC_CFG 0x13 -#define MAX98090_0x14_DIGITAL_MIC_MODE 0x14 - -/* ADC PATH AND CONFIGURATION REGISTERS */ -#define MAX98090_0x15_L_ADC_MIX 0x15 -#define MAX98090_0x16_R_ADC_MIX 0x16 -#define MAX98090_0x17_L_ADC_LVL 0x17 -#define MAX98090_0x18_R_ADC_LVL 0x18 -#define MAX98090_0x19_ADC_BIQUAD_LVL 0x19 -#define MAX98090_0x1A_ADC_SIDETONE 0x1A - -/* CLOCK CONFIGURATION REGISTERS */ -#define MAX98090_0x1B_SYS_CLK 0x1B -#define MAX98090_0x1C_CLK_MODE 0x1C -#define MAX98090_0x1D_ANY_CLK1 0x1D -#define MAX98090_0x1E_ANY_CLK2 0x1E -#define MAX98090_0x1F_ANY_CLK3 0x1F -#define MAX98090_0x20_ANY_CLK4 0x20 -#define MAX98090_0x21_MASTER_MODE 0x21 - -/* INTERFACE CONTROL REGISTERS */ -#define MAX98090_0x22_DAI_IF_FMT 0x22 -#define MAX98090_0x23_DAI_TDM_FMT1 0x23 -#define MAX98090_0x24_DAI_TDM_FMT2 0x24 -#define MAX98090_0x25_DAI_IO_CFG 0x25 -#define MAX98090_0x26_FILTER_CFG 0x26 -#define MAX98090_0x27_DAI_PLAYBACK_LVL 0x27 -#define MAX98090_0x28_EQ_PLAYBACK_LVL 0x28 - -/* HEADPHONE CONTROL REGISTERS */ -#define MAX98090_0x29_L_HP_MIX 0x29 -#define MAX98090_0x2A_R_HP_MIX 0x2A -#define MAX98090_0x2B_HP_CTR 0x2B -#define MAX98090_0x2C_L_HP_VOL 0x2C -#define MAX98090_0x2D_R_HP_VOL 0x2D - -/* SPEAKER CONFIGURATION REGISTERS */ -#define MAX98090_0x2E_L_SPK_MIX 0x2E -#define MAX98090_0x2F_R_SPK_MIX 0x2F -#define MAX98090_0x30_SPK_CTR 0x30 -#define MAX98090_0x31_L_SPK_VOL 0x31 -#define MAX98090_0x32_R_SPK_VOL 0x32 - -/* ALC CONFIGURATION REGISTERS */ -#define MAX98090_0x33_ALC_TIMING 0x33 -#define MAX98090_0x34_ALC_COMPRESSOR 0x34 -#define MAX98090_0x35_ALC_EXPANDER 0x35 -#define MAX98090_0x36_ALC_GAIN 0x36 - -/* RECEIVER AND LINE_OUTPUT REGISTERS */ -#define MAX98090_0x37_RCV_LOUT_L_MIX 0x37 -#define MAX98090_0x38_RCV_LOUT_L_CNTL 0x38 -#define MAX98090_0x39_RCV_LOUT_L_VOL 0x39 -#define MAX98090_0x3A_LOUT_R_MIX 0x3A -#define MAX98090_0x3B_LOUT_R_CNTL 0x3B -#define MAX98090_0x3C_LOUT_R_VOL 0x3C - -/* JACK DETECT AND ENABLE REGISTERS */ -#define MAX98090_0x3D_JACK_DETECT 0x3D -#define MAX98090_0x3E_IN_ENABLE 0x3E -#define MAX98090_0x3F_OUT_ENABLE 0x3F -#define MAX98090_0x40_LVL_CTR 0x40 -#define MAX98090_0x41_DSP_FILTER_ENABLE 0x41 - -/* BIAS AND POWER MODE CONFIGURATION REGISTERS */ -#define MAX98090_0x42_BIAS_CTR 0x42 -#define MAX98090_0x43_DAC_CTR 0x43 -#define MAX98090_0x44_ADC_CTR 0x44 -#define MAX98090_0x45_DEV_SHUTDOWN 0x45 - -/* REVISION ID REGISTER */ -#define MAX98090_0xFF_REV_ID 0xFF - -#define MAX98090_REG_MAX_CACHED 0x45 -#define MAX98090_REG_END 0xFF - -/* - * - * MAX98090 Registers Bit Fields - * - */ - -/* MAX98090_0x06_DAI_IF */ -#define MAX98090_DAI_IF_MASK 0x3F -#define MAX98090_RJ_M (1 << 5) -#define MAX98090_RJ_S (1 << 4) -#define MAX98090_LJ_M (1 << 3) -#define MAX98090_LJ_S (1 << 2) -#define MAX98090_I2S_M (1 << 1) -#define MAX98090_I2S_S (1 << 0) - -/* MAX98090_0x45_DEV_SHUTDOWN */ -#define MAX98090_SHDNRUN (1 << 7) - -/* codec private data */ -struct max98090_priv { - struct regmap *regmap; -}; - -static const struct reg_default max98090_reg_defaults[] = { - /* RESET / STATUS / INTERRUPT REGISTERS */ - {MAX98090_0x00_SW_RESET, 0x00}, - {MAX98090_0x01_INT_STS, 0x00}, - {MAX98090_0x02_JACK_STS, 0x00}, - {MAX98090_0x03_INT_MASK, 0x04}, - - /* QUICK SETUP REGISTERS */ - {MAX98090_0x04_SYS_CLK, 0x00}, - {MAX98090_0x05_SAMPLE_RATE, 0x00}, - {MAX98090_0x06_DAI_IF, 0x00}, - {MAX98090_0x07_DAC_PATH, 0x00}, - {MAX98090_0x08_MIC_TO_ADC, 0x00}, - {MAX98090_0x09_LINE_TO_ADC, 0x00}, - {MAX98090_0x0A_ANALOG_MIC_LOOP, 0x00}, - {MAX98090_0x0B_ANALOG_LINE_LOOP, 0x00}, - - /* ANALOG INPUT CONFIGURATION REGISTERS */ - {MAX98090_0x0D_INPUT_CONFIG, 0x00}, - {MAX98090_0x0E_LINE_IN_LVL, 0x1B}, - {MAX98090_0x0F_LINI_IN_CFG, 0x00}, - {MAX98090_0x10_MIC1_IN_LVL, 0x11}, - {MAX98090_0x11_MIC2_IN_LVL, 0x11}, - - /* MICROPHONE CONFIGURATION REGISTERS */ - {MAX98090_0x12_MIC_BIAS_VOL, 0x00}, - {MAX98090_0x13_DIGITAL_MIC_CFG, 0x00}, - {MAX98090_0x14_DIGITAL_MIC_MODE, 0x00}, - - /* ADC PATH AND CONFIGURATION REGISTERS */ - {MAX98090_0x15_L_ADC_MIX, 0x00}, - {MAX98090_0x16_R_ADC_MIX, 0x00}, - {MAX98090_0x17_L_ADC_LVL, 0x03}, - {MAX98090_0x18_R_ADC_LVL, 0x03}, - {MAX98090_0x19_ADC_BIQUAD_LVL, 0x00}, - {MAX98090_0x1A_ADC_SIDETONE, 0x00}, - - /* CLOCK CONFIGURATION REGISTERS */ - {MAX98090_0x1B_SYS_CLK, 0x00}, - {MAX98090_0x1C_CLK_MODE, 0x00}, - {MAX98090_0x1D_ANY_CLK1, 0x00}, - {MAX98090_0x1E_ANY_CLK2, 0x00}, - {MAX98090_0x1F_ANY_CLK3, 0x00}, - {MAX98090_0x20_ANY_CLK4, 0x00}, - {MAX98090_0x21_MASTER_MODE, 0x00}, - - /* INTERFACE CONTROL REGISTERS */ - {MAX98090_0x22_DAI_IF_FMT, 0x00}, - {MAX98090_0x23_DAI_TDM_FMT1, 0x00}, - {MAX98090_0x24_DAI_TDM_FMT2, 0x00}, - {MAX98090_0x25_DAI_IO_CFG, 0x00}, - {MAX98090_0x26_FILTER_CFG, 0x80}, - {MAX98090_0x27_DAI_PLAYBACK_LVL, 0x00}, - {MAX98090_0x28_EQ_PLAYBACK_LVL, 0x00}, - - /* HEADPHONE CONTROL REGISTERS */ - {MAX98090_0x29_L_HP_MIX, 0x00}, - {MAX98090_0x2A_R_HP_MIX, 0x00}, - {MAX98090_0x2B_HP_CTR, 0x00}, - {MAX98090_0x2C_L_HP_VOL, 0x1A}, - {MAX98090_0x2D_R_HP_VOL, 0x1A}, - - /* SPEAKER CONFIGURATION REGISTERS */ - {MAX98090_0x2E_L_SPK_MIX, 0x00}, - {MAX98090_0x2F_R_SPK_MIX, 0x00}, - {MAX98090_0x30_SPK_CTR, 0x00}, - {MAX98090_0x31_L_SPK_VOL, 0x2C}, - {MAX98090_0x32_R_SPK_VOL, 0x2C}, - - /* ALC CONFIGURATION REGISTERS */ - {MAX98090_0x33_ALC_TIMING, 0x00}, - {MAX98090_0x34_ALC_COMPRESSOR, 0x00}, - {MAX98090_0x35_ALC_EXPANDER, 0x00}, - {MAX98090_0x36_ALC_GAIN, 0x00}, - - /* RECEIVER AND LINE_OUTPUT REGISTERS */ - {MAX98090_0x37_RCV_LOUT_L_MIX, 0x00}, - {MAX98090_0x38_RCV_LOUT_L_CNTL, 0x00}, - {MAX98090_0x39_RCV_LOUT_L_VOL, 0x15}, - {MAX98090_0x3A_LOUT_R_MIX, 0x00}, - {MAX98090_0x3B_LOUT_R_CNTL, 0x00}, - {MAX98090_0x3C_LOUT_R_VOL, 0x15}, - - /* JACK DETECT AND ENABLE REGISTERS */ - {MAX98090_0x3D_JACK_DETECT, 0x00}, - {MAX98090_0x3E_IN_ENABLE, 0x00}, - {MAX98090_0x3F_OUT_ENABLE, 0x00}, - {MAX98090_0x40_LVL_CTR, 0x00}, - {MAX98090_0x41_DSP_FILTER_ENABLE, 0x00}, - - /* BIAS AND POWER MODE CONFIGURATION REGISTERS */ - {MAX98090_0x42_BIAS_CTR, 0x00}, - {MAX98090_0x43_DAC_CTR, 0x00}, - {MAX98090_0x44_ADC_CTR, 0x06}, - {MAX98090_0x45_DEV_SHUTDOWN, 0x00}, -}; - -static const unsigned int max98090_hp_tlv[] = { - TLV_DB_RANGE_HEAD(5), - 0x0, 0x6, TLV_DB_SCALE_ITEM(-6700, 400, 0), - 0x7, 0xE, TLV_DB_SCALE_ITEM(-4000, 300, 0), - 0xF, 0x15, TLV_DB_SCALE_ITEM(-1700, 200, 0), - 0x16, 0x1B, TLV_DB_SCALE_ITEM(-400, 100, 0), - 0x1C, 0x1F, TLV_DB_SCALE_ITEM(150, 50, 0), -}; - -static struct snd_kcontrol_new max98090_snd_controls[] = { - SOC_DOUBLE_R_TLV("Headphone Volume", MAX98090_0x2C_L_HP_VOL, - MAX98090_0x2D_R_HP_VOL, 0, 31, 0, max98090_hp_tlv), -}; - -/* Left HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x29_L_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x29_L_HP_MIX, 0, 1, 0), -}; - -/* Right HeadPhone Mixer Switch */ -static struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("DACR Switch", MAX98090_0x2A_R_HP_MIX, 1, 1, 0), - SOC_DAPM_SINGLE("DACL Switch", MAX98090_0x2A_R_HP_MIX, 0, 1, 0), -}; - -static struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - /* Output */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* PGA */ - SND_SOC_DAPM_PGA("HPL Out", MAX98090_0x3F_OUT_ENABLE, 7, 0, NULL, 0), - SND_SOC_DAPM_PGA("HPR Out", MAX98090_0x3F_OUT_ENABLE, 6, 0, NULL, 0), - - /* Mixer */ - SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, - max98090_left_hp_mixer_controls, - ARRAY_SIZE(max98090_left_hp_mixer_controls)), - - SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, - max98090_right_hp_mixer_controls, - ARRAY_SIZE(max98090_right_hp_mixer_controls)), - - /* DAC */ - SND_SOC_DAPM_DAC("DACL", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Hifi Playback", MAX98090_0x3F_OUT_ENABLE, 1, 0), -}; - -static struct snd_soc_dapm_route max98090_audio_map[] = { - /* Output */ - {"HPL", NULL, "HPL Out"}, - {"HPR", NULL, "HPR Out"}, - - /* PGA */ - {"HPL Out", NULL, "HPL Mixer"}, - {"HPR Out", NULL, "HPR Mixer"}, - - /* Mixer*/ - {"HPL Mixer", "DACR Switch", "DACR"}, - {"HPL Mixer", "DACL Switch", "DACL"}, - - {"HPR Mixer", "DACR Switch", "DACR"}, - {"HPR Mixer", "DACL Switch", "DACL"}, -}; - -static bool max98090_volatile(struct device *dev, unsigned int reg) -{ - if ((reg == MAX98090_0x01_INT_STS) || - (reg == MAX98090_0x02_JACK_STS) || - (reg > MAX98090_REG_MAX_CACHED)) - return true; - - return false; -} - -static int max98090_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - switch (params_rate(params)) { - case 96000: - val = 1 << 5; - break; - case 32000: - val = 1 << 4; - break; - case 48000: - val = 1 << 3; - break; - case 44100: - val = 1 << 2; - break; - case 16000: - val = 1 << 1; - break; - case 8000: - val = 1 << 0; - break; - default: - dev_err(codec->dev, "unsupported rate\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x05_SAMPLE_RATE, 0x03F, val); - - return 0; -} - -static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned int val; - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, 0); - - switch (freq) { - case 26000000: - val = 1 << 7; - break; - case 19200000: - val = 1 << 6; - break; - case 13000000: - val = 1 << 5; - break; - case 12288000: - val = 1 << 4; - break; - case 12000000: - val = 1 << 3; - break; - case 11289600: - val = 1 << 2; - break; - default: - dev_err(codec->dev, "Invalid master clock frequency\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x04_SYS_CLK, 0xFD, val); - - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - dev_dbg(dai->dev, "sysclk is %uHz\n", freq); - - return 0; -} - -static int max98090_dai_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct snd_soc_codec *codec = dai->codec; - int is_master; - u8 val; - - /* master/slave mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - is_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - is_master = 0; - break; - default: - dev_err(codec->dev, "unsupported clock\n"); - return -EINVAL; - } - - /* format */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_I2S: - val = (is_master) ? MAX98090_I2S_M : MAX98090_I2S_S; - break; - case SND_SOC_DAIFMT_RIGHT_J: - val = (is_master) ? MAX98090_RJ_M : MAX98090_RJ_S; - break; - case SND_SOC_DAIFMT_LEFT_J: - val = (is_master) ? MAX98090_LJ_M : MAX98090_LJ_S; - break; - default: - dev_err(codec->dev, "unsupported format\n"); - return -EINVAL; - } - snd_soc_update_bits(codec, MAX98090_0x06_DAI_IF, - MAX98090_DAI_IF_MASK, val); - - return 0; -} - -#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 -#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) - -static struct snd_soc_dai_ops max98090_dai_ops = { - .set_sysclk = max98090_dai_set_sysclk, - .set_fmt = max98090_dai_set_fmt, - .hw_params = max98090_dai_hw_params, -}; - -static struct snd_soc_dai_driver max98090_dai = { - .name = "max98090-Hifi", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = MAX98090_RATES, - .formats = MAX98090_FORMATS, - }, - .ops = &max98090_dai_ops, -}; - -static int max98090_probe(struct snd_soc_codec *codec) -{ - struct max98090_priv *priv = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; - int ret; - - codec->control_data = priv->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - /* Device active */ - snd_soc_update_bits(codec, MAX98090_0x45_DEV_SHUTDOWN, - MAX98090_SHDNRUN, MAX98090_SHDNRUN); - - return 0; -} - -static int max98090_remove(struct snd_soc_codec *codec) -{ - return 0; -} - -static struct snd_soc_codec_driver soc_codec_dev_max98090 = { - .probe = max98090_probe, - .remove = max98090_remove, - .controls = max98090_snd_controls, - .num_controls = ARRAY_SIZE(max98090_snd_controls), - .dapm_widgets = max98090_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(max98090_dapm_widgets), - .dapm_routes = max98090_audio_map, - .num_dapm_routes = ARRAY_SIZE(max98090_audio_map), -}; - -static const struct regmap_config max98090_regmap = { - .reg_bits = 8, - .val_bits = 8, - .max_register = MAX98090_REG_END, - .volatile_reg = max98090_volatile, - .cache_type = REGCACHE_RBTREE, - .reg_defaults = max98090_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(max98090_reg_defaults), -}; - -static int max98090_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct max98090_priv *priv; - struct device *dev = &i2c->dev; - unsigned int val; - int ret; - - priv = devm_kzalloc(dev, sizeof(struct max98090_priv), - GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); - if (IS_ERR(priv->regmap)) { - ret = PTR_ERR(priv->regmap); - dev_err(dev, "Failed to init regmap: %d\n", ret); - return ret; - } - - i2c_set_clientdata(i2c, priv); - - ret = regmap_read(priv->regmap, MAX98090_0xFF_REV_ID, &val); - if (ret < 0) { - dev_err(dev, "Failed to read device revision: %d\n", ret); - return ret; - } - dev_info(dev, "revision 0x%02x\n", val); - - ret = snd_soc_register_codec(dev, - &soc_codec_dev_max98090, - &max98090_dai, 1); - - return ret; -} - -static int max98090_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id max98090_i2c_id[] = { - { "max98090", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); - -static struct i2c_driver max98090_i2c_driver = { - .driver = { - .name = "max98090", - .owner = THIS_MODULE, - }, - .probe = max98090_i2c_probe, - .remove = max98090_i2c_remove, - .id_table = max98090_i2c_id, -}; -module_i2c_driver(max98090_i2c_driver); - -MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); -MODULE_AUTHOR("Peter Hsiang, Kuninori Morimoto"); -MODULE_LICENSE("GPL"); -- 1.7.0.4

2 7

[alsa-devel] [PATCH] ASoC: soc-compress: Add support for not memory mapped DSPs
by Charles Keepax 05 Feb '13

05 Feb '13

The ASoC compressed API did not implement the copy callback in its compressed ops which is required for DSPs that are not memory mapped. This patch adds a second set of compressed ops which does implement the copy callback and uses that when copy is defined in the platform compressed ops, ie. when the DSP is not memory mapped. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 36 +++++++++++++++++++++++++++++++++++- 1 files changed, 35 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea7956..f9fb112 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,6 +302,22 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int soc_compr_copy(struct snd_compr_stream *cstream, + const char __user *buf, size_t count) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + ret = platform->driver->compr_ops->copy(cstream, buf, count); + + mutex_unlock(&rtd->pcm_mutex); + return ret; +} + /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, @@ -315,10 +331,25 @@ static struct snd_compr_ops soc_compr_ops = { .get_codec_caps = soc_compr_get_codec_caps }; +/* ASoC Compress operations for non memory mapped DSPs */ +static struct snd_compr_ops soc_compr_nomap_ops = { + .open = soc_compr_open, + .free = soc_compr_free, + .set_params = soc_compr_set_params, + .get_params = soc_compr_get_params, + .trigger = soc_compr_trigger, + .pointer = soc_compr_pointer, + .copy = soc_compr_copy, + .ack = soc_compr_ack, + .get_caps = soc_compr_get_caps, + .get_codec_caps = soc_compr_get_codec_caps +}; + /* create a new compress */ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_compr *compr; @@ -335,7 +366,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -ENOMEM; } - compr->ops = &soc_compr_ops; + if (platform->driver->compr_ops && platform->driver->compr_ops->copy) + compr->ops = &soc_compr_nomap_ops; + else + compr->ops = &soc_compr_ops; mutex_init(&compr->lock); ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); if (ret < 0) { -- 1.7.2.5

3 11

[alsa-devel] [PATCH] ASoC: omap-pcm: No need to set constraint at open time
by Peter Ujfalusi 05 Feb '13

05 Feb '13

The same constraint is going to be set in the snd_dmaengine_pcm_open() function, so there is no need to set it here as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-pcm.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 47bdbd4..c722c2e 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -174,23 +174,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) static int omap_pcm_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_pcm_dma_data *dma_data; - int ret; snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); - /* Ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ret = snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, - &dma_data->dma_req); - return ret; + + return snd_dmaengine_pcm_open(substream, omap_dma_filter_fn, + &dma_data->dma_req); } static int omap_pcm_close(struct snd_pcm_substream *substream) -- 1.8.1.1

3 2

[alsa-devel] Join my network on LinkedIn
by Barry Song via LinkedIn 05 Feb '13

05 Feb '13

LinkedIn ------------ Barry Song requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Barry Accept invitation from Barry Song http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Barry Song http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/rso/196461153/a2mu/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hcsyt8h9-2d/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix phantom jacks on VT1708
by David Henningsson 05 Feb '13

05 Feb '13

The VT1708 has no unsol event capability, and polling is set using the "Jack Detect" alsamixer control. In order not to create phantom Jack controls, temporary enable jackpoll during build_controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) Sorry, the previous version of this patch the "int err" got lost. diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9641c0e..e934c49 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -661,6 +661,18 @@ static int via_init(struct hda_codec *codec) return 0; } +static int vt1708_build_controls(struct hda_codec *codec) +{ + /* In order not to create "Phantom Jack" controls, + temporary enable jackpoll */ + int err; + int old_interval = codec->jackpoll_interval; + codec->jackpoll_interval = msecs_to_jiffies(100); + err = via_build_controls(codec); + codec->jackpoll_interval = old_interval; + return err; +} + static int vt1708_build_pcms(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -723,6 +735,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; codec->patch_ops = via_patch_ops; + codec->patch_ops.build_controls = vt1708_build_controls; codec->patch_ops.build_pcms = vt1708_build_pcms; /* clear jackpoll_interval again; it's set dynamically */ -- 1.7.9.5

2 2

[alsa-devel] How exactly snd_pcm_hw_rule_add() should to be used?
by Peter Ujfalusi 05 Feb '13

05 Feb '13

Hi, I got confused (again) regarding to how to use correctly the snd_pcm_hw_rule_add(). Looking at the documentation [1] and to the code in kernel seams to be suggesting the opposite. [A] using the 5.6. example from [1]: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, -1); The [1] tells me that the SNDRV_PCM_HW_PARAM_FORMAT will be refined based on the SNDRV_PCM_HW_PARAM_CHANNELS. In 5.7. of [1] the reverse rule is placed: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_CHANNELS, -1); Which according to the example will refine the SNDRV_PCM_HW_PARAM_CHANNELS based on the SNDRV_PCM_HW_PARAM_FORMAT. However when looking at the code in the kernel the opposite seams to be what is actually happening. [B] using the same 5.6. example from [1]: snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, -1); It looks like the SNDRV_PCM_HW_PARAM_CHANNELS should be refined based on the SNDRV_PCM_HW_PARAM_FORMAT. Which one is the correct interpretation? The reason I'm asking is that I need to have a rule which would place constraint (step) on the BUFFER_SIZE based on the used RATE, but I can not use a single snd_pcm_hw_constraint_step(,,BUFFER_TIME,) because of HW limitation. Thank you, Péter [1] http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s09.html

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix phantom jacks on VT1708
by David Henningsson 05 Feb '13

05 Feb '13

The VT1708 has no unsol event capability, and polling is set using the "Jack Detect" alsamixer control. In order not to create phantom Jack controls, temporary enable jackpoll during build_controls. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9641c0e..d46c448 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -661,6 +661,17 @@ static int via_init(struct hda_codec *codec) return 0; } +static int vt1708_build_controls(struct hda_codec *codec) +{ + /* In order not to create "Phantom Jack" controls, + temporary enable jackpoll */ + int old_interval = codec->jackpoll_interval; + codec->jackpoll_interval = msecs_to_jiffies(100); + err = via_build_controls(codec); + codec->jackpoll_interval = old_interval; + return err; +} + static int vt1708_build_pcms(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -723,6 +734,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; codec->patch_ops = via_patch_ops; + codec->patch_ops.build_controls = vt1708_build_controls; codec->patch_ops.build_pcms = vt1708_build_pcms; /* clear jackpoll_interval again; it's set dynamically */ -- 1.7.9.5

1 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Use asynchronous I/O to write firmware and coefficients
by Mark Brown 04 Feb '13

04 Feb '13

Allow the regmap API to use asynchronous I/O where supported to minimise the delay between transfers, reducing firmware download times. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 82 +++++++++++++++++++++++++++++++++++--------- 1 file changed, 66 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5487a94..be45e2b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/firmware.h> +#include <linux/list.h> #include <linux/pm.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> @@ -153,6 +154,43 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +struct wm_adsp_buf { + struct list_head list; + void *buf; +}; + +static struct wm_adsp_buf *wm_adsp_buf_alloc(const void *src, size_t len, + struct list_head *list) +{ + struct wm_adsp_buf *buf = kzalloc(sizeof(*buf), GFP_KERNEL); + + if (buf == NULL) + return NULL; + + buf->buf = kmemdup(src, len, GFP_KERNEL | GFP_DMA); + if (!buf->buf) { + kfree(buf); + return NULL; + } + + if (list) + list_add_tail(&buf->list, list); + + return buf; +} + +static void wm_adsp_buf_free(struct list_head *list) +{ + while (!list_empty(list)) { + struct wm_adsp_buf *buf = list_first_entry(list, + struct wm_adsp_buf, + list); + list_del(&buf->list); + kfree(buf->buf); + kfree(buf); + } +} + #define WM_ADSP_NUM_FW 4 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { @@ -254,6 +292,7 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, static int wm_adsp_load(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); const struct firmware *firmware; struct regmap *regmap = dsp->regmap; unsigned int pos = 0; @@ -265,7 +304,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; - void *buf; + struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -420,18 +459,16 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(region->data, le32_to_cpu(region->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(region->data, + le32_to_cpu(region->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; } - ret = regmap_raw_write(regmap, reg, buf, - le32_to_cpu(region->len)); - - kfree(buf); - + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(region->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -445,12 +482,20 @@ static int wm_adsp_load(struct wm_adsp *dsp) pos += le32_to_cpu(region->len) + sizeof(*region); regions++; } - + + ret = regmap_async_complete(regmap); + if (ret != 0) { + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + goto out_fw; + } + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); out_fw: + regmap_async_complete(regmap); + wm_adsp_buf_free(&buf_list); release_firmware(firmware); out: kfree(file); @@ -655,6 +700,7 @@ out: static int wm_adsp_load_coeff(struct wm_adsp *dsp) { + LIST_HEAD(buf_list); struct regmap *regmap = dsp->regmap; struct wmfw_coeff_hdr *hdr; struct wmfw_coeff_item *blk; @@ -664,7 +710,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; - void *buf; + struct wm_adsp_buf *buf; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -776,8 +822,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { - buf = kmemdup(blk->data, le32_to_cpu(blk->len), - GFP_KERNEL | GFP_DMA); + buf = wm_adsp_buf_alloc(blk->data, + le32_to_cpu(blk->len), + &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; @@ -786,27 +833,30 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", file, blocks, le32_to_cpu(blk->len), reg); - ret = regmap_raw_write(regmap, reg, blk->data, - le32_to_cpu(blk->len)); + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write to %x in %s\n", file, blocks, reg, region_name); } - - kfree(buf); } pos += le32_to_cpu(blk->len) + sizeof(*blk); blocks++; } + ret = regmap_async_complete(regmap); + if (ret != 0) + adsp_err(dsp, "Failed to complete async write: %d\n", ret); + if (pos > firmware->size) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); out_fw: release_firmware(firmware); + wm_adsp_buf_free(&buf_list); out: kfree(file); return 0; -- 1.7.10.4

1 0

[alsa-devel] [PATCH 0/4] ASoC: atmel: fixes for working as module
by Bo Shen 04 Feb '13

04 Feb '13

The following patches fixes the audio part when working as module Bo Shen (4): ASoC: atmel_pcm: make it buildable as module ASoC: atmel_ssc_dai: remove error set private data ASoC: atmel_ssc_dai: correct sequence when unload ASoC: sam9g20_wm8731: disable clock and correct sequence when unload sound/soc/atmel/atmel-pcm.h | 6 ++++-- sound/soc/atmel/atmel_ssc_dai.c | 12 +----------- sound/soc/atmel/sam9g20_wm8731.c | 6 +++--- 3 files changed, 8 insertions(+), 16 deletions(-) -- 1.7.9.5

2 5

[alsa-devel] [PATCH] ASoC: wm2200: Provide platform data for MICBIAS configuration
by Chris Rattray 04 Feb '13

04 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- include/sound/wm2200.h | 22 +++++++++++++++++++++- sound/soc/codecs/wm2200.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 79bf55b..26ea6ae 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -12,6 +12,7 @@ #define __LINUX_SND_WM2200_H #define WM2200_GPIO_SET 0x10000 +#define WM2200_MAX_MICBIAS 2 enum wm2200_in_mode { WM2200_IN_SE = 0, @@ -25,6 +26,24 @@ enum wm2200_dmic_sup { WM2200_DMIC_SUP_MICBIAS2 = 2, }; +enum wm2200_mbias_lvl { + WM2200_MBIAS_LVL_1V5 = 0, + WM2200_MBIAS_LVL_1V8 = 1, + WM2200_MBIAS_LVL_1V9 = 2, + WM2200_MBIAS_LVL_2V0 = 3, + WM2200_MBIAS_LVL_2V2 = 4, + WM2200_MBIAS_LVL_2V4 = 5, + WM2200_MBIAS_LVL_2V5 = 6, + WM2200_MBIAS_LVL_2V6 = 7, +}; + +struct wm2200_micbias { + enum wm2200_mbias_lvl mb_lvl; /** Regulated voltage */ + unsigned int discharge:1; /** Actively discharge */ + unsigned int fast_start:1; /** Enable aggressive startup ramp rate */ + unsigned int bypass:1; /** Use bypass mode */ +}; + struct wm2200_pdata { int reset; /** GPIO controlling /RESET, if any */ int ldo_ena; /** GPIO controlling LODENA, if any */ @@ -35,7 +54,8 @@ struct wm2200_pdata { enum wm2200_in_mode in_mode[3]; enum wm2200_dmic_sup dmic_sup[3]; - int micbias_cfg[2]; /** Register value to configure MICBIAS */ + /** MICBIAS configurations */ + struct wm2200_micbias micbias[WM2200_MAX_MICBIAS]; }; #endif diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..e458252 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2192,6 +2192,7 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, struct wm2200_priv *wm2200; unsigned int reg; int ret, i; + int val; wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), GFP_KERNEL); @@ -2342,6 +2343,36 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); } + for (i = 0; i < WM2200_MAX_MICBIAS; i++) { + if (!wm2200->pdata.micbias[i].mb_lvl && + !wm2200->pdata.micbias[i].bypass) + continue; + + /* Apply default for bypass mode */ + if (!wm2200->pdata.micbias[i].mb_lvl) + wm2200->pdata.micbias[i].mb_lvl + = WM2200_MBIAS_LVL_1V5; + + val = wm2200->pdata.micbias[i].mb_lvl + << WM2200_MICB1_LVL_SHIFT; + + if (wm2200->pdata.micbias[i].discharge) + val |= WM2200_MICB1_DISCH; + + if (wm2200->pdata.micbias[i].fast_start) + val |= WM2200_MICB1_RATE; + + if (wm2200->pdata.micbias[i].bypass) + val |= WM2200_MICB1_MODE; + + regmap_update_bits(wm2200->regmap, + WM2200_MIC_BIAS_CTRL_1 + i, + WM2200_MICB1_LVL_MASK | + WM2200_MICB1_DISCH | + WM2200_MICB1_MODE | + WM2200_MICB1_RATE, val); + } + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], WM2200_IN1_MODE_MASK | -- 1.7.2.5

2 1

[alsa-devel] CA0132 DSP loader code merged to for-next branch
by Takashi Iwai 04 Feb '13

04 Feb '13

Hi Ian, FYI, right now I merged the pending CA0132 DSP stuff into for-next branch, targeted for 3.9 kernel. I changed the Kconfig to CONFIG_SND_HDA_CODEC_CA0132_DSP and leave it as default NO, in order to be compatible with the older kernels. Let's see whether the things go well with it. thanks, Takashi

1 0

[alsa-devel] [RESEND PATCH v3 0/5] ARM: OMAP4+: HDMI: Update platform devices for audio
by Ricardo Neri 04 Feb '13

04 Feb '13

Hi Mark, Tomi, Liam, Tony, I am resending this series as Mark and possibly other people missed it. This set aims to be the version 3 of my previous submission[1] and aims to address the comments that Mark and Tomi kindly provided on such submission. The creation of the platform device for the HDMI audio interface from within the OMAPDSS HDMI driver that was previously submitted[2] is resubmitted to be complemented with code to relocate to arch/arm/mach-omap2/display.c the creation of the platform devices for the HDMI ASoC codec and card drivers. This series does not break the HDMI audio functionality in any patch. Also, the names of the platform devices are changed to give them more logical and more descriptive names. As the commit commit 14840b9a83c6a56629db2ba0ec247503e975f143 Author: Ricardo Neri <ricardo.neri(a)ti.com> Date: Tue Nov 6 00:19:17 2012 -0600 OMAPDSS: HDMI: Create platform device for audio support is reverted in Tomi's git://gitorious.org/linux-omap-dss2/linux.git master branch, this series applies cleanly. Changes from v1: *Put in a single series all the patches related to platform device updates. *Now HDMI audio works correctly in every patch. *Remove reference to the TPD12S015 HDMI companion chip as the ASoC drivers are not aware of this and other chips could be used in the future. Changes from v2: *Split in two patches the renaming and the relocation of the platform devices. *Create a separate patch to pass only the address offset of the DMA data port to the audio interface platform device. *Keep the name hdmi-audio-codec to not refer to explicitly to OMAP. The codec can be made generic in a different patch series submitted to alsa-devel. BR, Ricardo [1]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg80785.html [2]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg79795.html Ricardo Neri (5): ARM: OMAP4+: HDMI: Rename platform devices for ASoC drivers ARM: OMAP4: Assign IDs to DSS HDMI devices ARM4: OMAP4+: HDMI: Relocate devices for audio codec and card ARM: OMAP4+: HDMI: Relocate the device for audio interface ARM: OMAP4+: HDMI: Refine the DMA port resource for audio arch/arm/mach-omap2/board-4430sdp.c | 9 ++--- arch/arm/mach-omap2/board-omap4panda.c | 9 ++--- arch/arm/mach-omap2/devices.c | 31 ---------------- arch/arm/mach-omap2/display.c | 31 ++++++++++++++++ drivers/video/omap2/dss/hdmi.c | 62 ++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 4 +-- sound/soc/omap/omap-hdmi.c | 5 ++- sound/soc/omap/omap-hdmi.h | 2 -- 8 files changed, 103 insertions(+), 50 deletions(-) -- 1.7.10.4

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[alsa-devel] alsa usb-audio sound corruption caused by cached vmalloc mapping
by Matthieu CASTET 04 Feb '13

04 Feb '13

Hi, on a armv5 soc, we have audio corruption while using a usb-audio sound card. After analysis, it comes from alsa vmalloc mapping. in the current kernel vmalloc mapping used by alsa are cached. This cause coherency problem on armv5 architecture because the cache are VIVT : - alsa kernel driver map a buffer with vmalloc. The memory attribute is cached - userspace map vmalloc buffer. The memory attribute is cached But because the cache are VIVT, these two mapping don't share the same cache line and won't see the same data. This corruption can be hidden by context switch because cache are flushed on context switch. On x86 or armv7, the cache are PIPT and there is no such problem. There was a commit "pcm - Call pgprot_noncached() for vmalloc'ed buffers" (c32d977b8157bf67cdf47729ce7dd054a26eb534), that made the userspace mapping not cached, but this caused issues because the kernel vmalloc mapping was still cached. Instead of revering this commit, we could have made the kernel vmalloc noncached (see attached patch). It fix the issue on armv5. But on some architecture (armv7 for example) you can't have mapping for the same physical memory cached (normal cached) and noncached (device memory). So the vmalloc and userspace mapping will conflict with the kernel direct RAM mapping. A workaround could be to use writecombine protection instead of noncached. This work because writecombine is mapped on armv7 as L_PTE_MT_BUFFERABLE (normal uncached) that is compatible with cached memory (the same trick is used for dmacoherent). But that's architecture specific may not work on others architectures. What's the best way the fix this issue on all architecture ? Matthieu

2 1

[alsa-devel] alsa bt878 driver
by grigore calugar 04 Feb '13

04 Feb '13

Please add support in "snd-bt87x" for this card: 137 -> Conceptronic CTVFMi v2 [036e:109e] Best regards.

2 1

[alsa-devel] [PATCH RFC 0/4] Driver for RME MADI FX
by Adrian Knoth 04 Feb '13

04 Feb '13

Hi! I've been developing an out-of-tree driver for the RME MADI FX. History is available at https://github.com/adiknoth/madifx but doesn't follow any kernel convention regarding commit isolation, it was basically a backup/scratchpad during development. I don't have too much feedback, yet, but I guess it makes sense to mainline it earlier than later, increase the driver's exposure and fix any problems when the card is in more widespread use. Note that this patch series is essentially RFC right now to gather developer feedback. Cheers Adrian Knoth (4): ALSA: madifx - Makefile entries for RME MADI FX ALSA: Add RME MADI FX Kconfig entries ALSA: madifx - Expose preliminary userspace interface ALSA: madifx - Add support for RME MADI FX include/uapi/sound/madifx.h | 173 ++ sound/pci/Kconfig | 21 + sound/pci/rme9652/Makefile | 2 + sound/pci/rme9652/madifx.c | 3986 +++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 4182 insertions(+) create mode 100644 include/uapi/sound/madifx.h create mode 100644 sound/pci/rme9652/madifx.c -- 1.7.10.4

2 7

[alsa-devel] [PATCH] ALSA: core: don't return uninitialized snd_compr_tstamp
by Mark Brown 04 Feb '13

04 Feb '13

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> The snd_compr_update_tstamp() can only fill in the snd_compr_tstamp if the codec implements the pointer() function. If that happened the code was previously returning uninitialized garbage in the tstamp because it wasn't initialized anywhere. This change zero-fills the tstamp in the two places it is used before calling snd_compr_update_tstamp(), and also has snd_compr_update_tstamp() return an error indication if it can't provide a tstamp. For the case of snd_compr_calc_avail() it ignores this error because we still need to return info on the available buffer space even if we can't provide tstamp info - when the tstamp is not valid all fields are now guaranteed to be zero. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2d62068 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -144,16 +144,17 @@ static int snd_compr_free(struct inode *inode, struct file *f) return 0; } -static void snd_compr_update_tstamp(struct snd_compr_stream *stream, +static int snd_compr_update_tstamp(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { if (!stream->ops->pointer) - return; + return -ENOTSUPP; stream->ops->pointer(stream, tstamp); pr_debug("dsp consumed till %d total %d bytes\n", tstamp->byte_offset, tstamp->copied_total); stream->runtime->hw_pointer = tstamp->byte_offset; stream->runtime->total_bytes_transferred = tstamp->copied_total; + return 0; } static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, @@ -161,7 +162,9 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, { long avail_calc; /*this needs to be signed variable */ + memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); + /* Still need to return avail even if tstamp can't be filled in */ /* FIXME: This needs to be different for capture stream, available is # of compressed data, for playback it's @@ -517,11 +520,14 @@ out: static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { - struct snd_compr_tstamp tstamp; + struct snd_compr_tstamp tstamp = {0}; + int ret; - snd_compr_update_tstamp(stream, &tstamp); - return copy_to_user((struct snd_compr_tstamp __user *)arg, - &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + ret = snd_compr_update_tstamp(stream, &tstamp); + if (ret == 0) + ret = copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + return ret; } static int snd_compr_pause(struct snd_compr_stream *stream) -- 1.7.10.4

4 3

[alsa-devel] [PATCH] ASOC: imx: audmux: add a integrated configuration function
by Gary Zhang 04 Feb '13

04 Feb '13

based on imx_audmux_v2_configure_port(), add a integrated interface to configure audmux port conveniently Signed-off-by: Gary Zhang <b13634(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 42 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d9..ebd38a0 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -1,5 +1,5 @@ /* - * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012-2013 Freescale Semiconductor, Inc. * Copyright 2012 Linaro Ltd. * Copyright 2009 Pengutronix, Sascha Hauer <s.hauer(a)pengutronix.de> * @@ -244,6 +244,47 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); +int imx_audmux_configure_port(unsigned int int_port, unsigned int ext_port) +{ + int ret; + + if ((int_port > 7) || (ext_port > 7)) + return -EINVAL; + + if (audmux_type != IMX31_AUDMUX) + return -EINVAL; + + if (!audmux_base) + return -ENOSYS; + + if (audmux_clk) + clk_prepare_enable(audmux_clk); + + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + pr_err("audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + pr_err("audmux external port setup failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(imx_audmux_configure_port); + static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; -- 1.7.0.4

3 5

[alsa-devel] [PATCH] ASoC: wm2200: Add Rx ANC input select control
by Chris Rattray 04 Feb '13

04 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Change-Id: Iee28d1898327633a3166dd340410a37796959c0c --- sound/soc/codecs/wm2200.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..a915c76 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1109,6 +1109,16 @@ static int wm2200_mixer_values[] = { static WM2200_MUX_CTL_DECL(name##_aux5); \ static WM2200_MUX_CTL_DECL(name##_aux6); +static const char *wm2200_rxanc_input_sel_texts[] = { + "None", "IN1", "IN2", "IN3", +}; + +static const struct soc_enum wm2200_rxanc_input_sel = + SOC_ENUM_SINGLE(WM2200_RXANC_SRC, + WM2200_IN_RXANC_SEL_SHIFT, + ARRAY_SIZE(wm2200_rxanc_input_sel_texts), + wm2200_rxanc_input_sel_texts); + static const struct snd_kcontrol_new wm2200_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL, WM2200_IN1_OSR_SHIFT, 1, 0), @@ -1168,6 +1178,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), +SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm2200: correct IN2L and IN3L digital mute
by Chris Rattray 04 Feb '13

04 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index a915c76..f5410ec 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1136,9 +1136,9 @@ SOC_DOUBLE_R_TLV("IN3 Volume", WM2200_IN3L_CONTROL, WM2200_IN3R_CONTROL, SOC_DOUBLE_R("IN1 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, +SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_2L, WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, +SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_3L, WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_1L, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix default multichannel HDMI mapping regression
by Anssi Hannula 04 Feb '13

04 Feb '13

Commit d45e6889ee69456a4d5b1bbb32252f460cd48fa9 ("ALSA: hda - Provide the proper channel mapping for generic HDMI driver") added support for custom channel maps in the HDA HDMI driver. Due to a mistake in an 'if' condition the custom map is always used even when no such map has been set. This causes incorrect channel mapping for multichannel audio by default. Pass per_pin->chmap_set to hdmi_setup_channel_mapping() as a parameter so that it can use it for detecting if a custom map has been set instead of checking if map is NULL (which is never the case). Reported-by: Staffan Lindberg <pike(a)xbmc.org> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: stable(a)vger.kernel.org --- This also affects stable 3.7, but not earlier versions. sound/pci/hda/patch_hdmi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 807a2aa..e85959f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -714,9 +714,10 @@ static void hdmi_setup_fake_chmap(unsigned char *map, int ca) static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, bool non_pcm, int ca, - int channels, unsigned char *map) + int channels, unsigned char *map, + bool chmap_set) { - if (!non_pcm && map) { + if (!non_pcm && chmap_set) { hdmi_manual_setup_channel_mapping(codec, pin_nid, channels, map); } else { @@ -905,7 +906,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, pin_nid, channels); hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap); + channels, per_pin->chmap, + per_pin->chmap_set); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -915,7 +917,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * accordingly */ if (per_pin->non_pcm != non_pcm) hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap); + channels, per_pin->chmap, + per_pin->chmap_set); } per_pin->non_pcm = non_pcm; -- 1.7.10

2 1

[alsa-devel] Loopback interface: suspend/resume leads to a kernel crash
by El boulangero 04 Feb '13

04 Feb '13

Hi, I use the loopback device all the time. It works pretty good, except when I suspend my computer to RAM. On resume, I've got a kernel call trace. Here is the procedure to reproduce this behaviour, it's pretty easy. # Insert module if needed $ modprobe snd_aloop pcm_substreams=2 # Let's see how it looks like $ aplay -l **** Liste des Périphériques Matériels PLAYBACK **** carte 0: PCH [HDA Intel PCH], périphérique 0: VT1802 Analog [VT1802 Analog] Sous-périphériques: 1/1 Sous-périphérique #0: subdevice #0 carte 0: PCH [HDA Intel PCH], périphérique 2: VT1802 HP [VT1802 HP] Sous-périphériques: 1/1 Sous-périphérique #0: subdevice #0 carte 0: PCH [HDA Intel PCH], périphérique 3: HDMI 0 [HDMI 0] Sous-périphériques: 1/1 Sous-périphérique #0: subdevice #0 carte 1: Loopback [Loopback], périphérique 0: Loopback PCM [Loopback PCM] Sous-périphériques: 2/2 Sous-périphérique #0: subdevice #0 Sous-périphérique #1: subdevice #1 carte 1: Loopback [Loopback], périphérique 1: Loopback PCM [Loopback PCM] Sous-périphériques: 2/2 Sous-périphérique #0: subdevice #0 Sous-périphérique #1: subdevice #1 # Use the loopback device $ aplay -D hw:1,0,0 sound.wave & # Suspend computer $ pm-suspend # Wake up the computer That's enough to reproduce this bug. Here are the kernel logs: [ 181.994434] PM: resume of devices complete after 3301.753 msecs [ 182.006413] Restarting tasks ... done. [ 182.006590] ------------[ cut here ]------------ [ 182.006605] kernel BUG at kernel/timer.c:908! [ 182.006619] video LNXVIDEO:00: Restoring backlight state [ 182.006621] invalid opcode: 0000 [#1] PREEMPT SMP [ 182.006640] Modules linked in: snd_aloop af_packet nfsd auth_rpcgss nfs_acl nfs lockd fscache sunrpc ipv6 loop uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev hid_generic usbhid coretemp kvm_intel kvm crc32c_intel ghash_clmulni_intel aesni_intel xts aes_x86_64 lrw i915 gf128mul i2c_algo_bit intel_agp intel_gtt drm_kms_helper ablk_helper drm cryptd iTCO_wdt snd_hda_codec_hdmi i2c_i801 psmouse iTCO_vendor_support snd_hda_codec_via snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_page_alloc iwlwifi cfg80211 snd_seq snd_seq_device snd_timer snd ehci_pci button battery ac rtc_cmos microcode video i2c_core soundcore lpc_ich evdev serio_raw mei wmi ehci_hcd rfkill ext4 crc16 jbd2 mbcache sg sr_mod sd_mod cdrom ahci libahci libata scsi_mod thermal r8169 mii xhci_hcd [ 182.006642] CPU 1 [ 182.006642] Pid: 2982, comm: aplay Not tainted 3.8.0-rc5-kdump #1 System76, Inc. Gazelle Professional/Gazelle Professional [ 182.006646] RIP: 0010:[<ffffffff81063f8f>] [<ffffffff81063f8f>] add_timer+0xf/0x20 [ 182.006647] RSP: 0018:ffff880214a93d40 EFLAGS: 00010082 [ 182.006647] RAX: 00000000fffe34c9 RBX: ffff88021570e200 RCX: 00000000fffe344c [ 182.006648] RDX: 000000000001f3ff RSI: 0000000000f42400 RDI: ffff88021492f350 [ 182.006648] RBP: ffff88021492f300 R08: 000000000001f400 R09: 0000000000000020 [ 182.006649] R10: ffff88021588f800 R11: 0000000000000206 R12: 00000000fffffffe [ 182.006649] R13: ffff880214a3cc00 R14: ffff880214bbc464 R15: ffff88021588f800 [ 182.006650] FS: 00007f3489ed0700(0000) GS:ffff88021f240000(0000) knlGS:0000000000000000 [ 182.006650] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 182.006651] CR2: 0000000000000000 CR3: 00000002148fe000 CR4: 00000000001407e0 [ 182.006652] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 182.006652] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 182.006653] Process aplay (pid: 2982, threadinfo ffff880214a92000, task ffff880215928000) [ 182.006653] Stack: [ 182.006654] ffffffffa0d0947c 00000000000000b4 00000b2c00000003 0000000000003e80 [ 182.006655] ffffffffa02fbe80 ffff880214a3cc00 0000000000000003 ffff880214a3cc00 [ 182.006656] 0000000000000000 0000000000000fa0 ffffffffa02f01b0 0000000000003e80 [ 182.006656] Call Trace: [ 182.006659] [<ffffffffa0d0947c>] ? loopback_trigger+0x20c/0x2e0 [snd_aloop] [ 182.006664] [<ffffffffa02f01b0>] ? snd_pcm_action_single+0x30/0x80 [snd_pcm] [ 182.006667] [<ffffffffa02f86d1>] ? snd_pcm_lib_write1+0x391/0x400 [snd_pcm] [ 182.006669] [<ffffffffa02f5ab0>] ? snd_pcm_lib_readv_transfer+0x170/0x170 [snd_pcm] [ 182.006672] [<ffffffffa02f3f05>] ? snd_pcm_playback_ioctl1+0x255/0x270 [snd_pcm] [ 182.006674] [<ffffffffa02f3f47>] ? snd_pcm_playback_ioctl+0x27/0x30 [snd_pcm] [ 182.006676] [<ffffffff8115f43f>] ? do_vfs_ioctl+0x8f/0x530 [ 182.006678] [<ffffffff8114e4de>] ? vfs_read+0xfe/0x180 [ 182.006679] [<ffffffff8115f980>] ? sys_ioctl+0xa0/0xc0 [ 182.006681] [<ffffffff814591ad>] ? system_call_fastpath+0x1a/0x1f [ 182.006691] Code: e9 4c fe ff ff 4d 89 f4 e9 d4 fe ff ff 66 66 66 66 66 66 2e 0f 1f 84 00 00 00 00 00 48 83 3f 00 75 09 48 8b 77 10 e9 a1 fd ff ff <0f> 0b 66 66 66 66 66 66 2e 0f 1f 84 00 00 00 00 00 48 83 ec 08 [ 182.006692] RIP [<ffffffff81063f8f>] add_timer+0xf/0x20 [ 182.006692] RSP <ffff880214a93d40> I tried to change a few things in the snd_aloop driver, but this code is way beyond my understanding. I can't fix that myself. Feel free to contact me if you need more information. Thanks ! Regards, Arnaud

2 4

[alsa-devel] [PATCH] UAC2 clock selector - fixed?
by Louis Gorenfeld 03 Feb '13

03 Feb '13

Hi, I'd been struggling with a UAC2 device that I couldn't set the clock on. I finally got in touch with a driver author who pointed me to some official documentation (the USB Device Class Definition for Audio Devices). Where the ALSA code always sends a 16-bit packet to control mixer settings, the docs say that the clock selector control uses layout 1 which is 8-bit. By changing the routine to send an 8-bit packet, that control now works. This is the patch that fixes the problem for me (for ALSA 1.0.25). Is this proper? --- a/linux-3.4/sound/usb/mixer.c 2012-05-20 15:29:13.000000000 -0700 +++ b/vanilla-3.4/sound/usb/mixer.c 2013-01-02 17:06:11.009660059 -0800 @@ -422,8 +422,13 @@ if (cval->mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + /* audio class v2 controls are often but not always 2 bytes in size */ + /* For example, the clock selector (see 5.2.5.2.1 in the Universal Serial + * Bus Device Class Definition for Audio Devices document) */ + if (cval->control == UAC2_CX_CLOCK_SELECTOR) + val_len = sizeof(__u8); + else + val_len = sizeof(__u16); /* FIXME */ if (request != UAC_SET_CUR) { Thanks much! Louis

2 1

[alsa-devel] LAC 2013: the Linux Audio Conference - Deadline Extension
by IOhannes m zmoelnig - LAC 2013 Organizing Team 02 Feb '13

02 Feb '13

sorry for >< please >> << Hi all, The Linux Audio Conference submissions deadline has been extended! It is now February 17th, 2013 (23:59 HAST) So, if you were considering to submit a paper but couldn't make up your mind yet, here is your chance to become active! Never forget that this conference lives through the people participating in it. FEBRUARY 17th is the new deadline for all submission types: papers, music, installations, workshop proposals. Check out the link below more info: http://lac.linuxaudio.org/2013/participation Please spread this information to anyone who might be interested. If you have any questions, drop us a line at lac(a)linuxaudio.org We are looking forward to seeing you in Graz in May! Thanks and happy last-minute music-and-paper-submissions, on behalf of the LAC2013 organization team, IOhannes m zmoelnig --- LAC 2013: the Linux Audio Conference May 9-12, 2013 @ IEM, Graz/Austria lac(a)linuxaudio.org http://lac.iem.at

1 0

[alsa-devel] Lenovo ThinkCentre A58 - surround 5.1 problem
by Raymond Yau 02 Feb '13

02 Feb '13

> I already tried so many options in alsa-mixer - as far as they where possible to set. As we say im Germany: "I'm out of my latin" ;-) > > Any Ideas how to go on? > >> > >> > I'm running a Lenovo ThinkCentre A58 with a ALC662 rev. 1 chip on >> board. The mainboard has 3 jacks at the back of the PC and 2 ones at >> it's front (mic and headphones). >> > >> > The problem is, that the connectors are not automatically arranged to >> playback 5.1 surround sound. I must manually set the "model" to >> "asusmode4" to get 5.1 and then run a script as root, to set the >> backside jacks to output and to Center/LFE and rear speaker. Also I have >> to mute the internal speaker. >> > >> > I've tried to patch the module manually, but this went wrong somehow. >> I don't know how to proceed. >> > >> > So maybe you can help me and I can help you to complete the ALSA code >> for this special mainboard/PC - hopefully. >> > >> > So long, I would thank you for your feedback. >> > >> For a desktop with/without internal speaker, three audio jacks at rear panel Green lineout jack , blue and pink jacks should have higher priority to allocate DAC (audio output node 0x02, 0x03 and 0x04) than the headphone and the internal speaker to support surround 5.1 This mean that in function parse_output_paths() to prevent the assignment of dac speaker - bool fill_hardwired = true, fill_mio_first = true; + bool fill_hardwired = false, fill_mio_first = true; Or use the existence of internal Mic to determine the value of fill_hardwired

1 0

[alsa-devel] New one on board - a damn rooky :">
by Rüdi 02 Feb '13

02 Feb '13

Hello @ all, I'm Rüdiger from Germany. I'm no programmer, that's my biggest fault here. But because of a "private" problem with ALSA and Intel HDA I want to help whereever I can - at least with that special problem. I'm running a Lenovo ThinkCentre A58 with a ALC662 rev. 1 chip on board. The mainboard has 3 jacks at the back of the PC and 2 ones at it's front (mic and headphones). The problem is, that the connectors are not automatically arranged to playback 5.1 surround sound. I must manually set the "model" to "asusmode4" to get 5.1 and then run a script as root, to set the backside jacks to output and to Center/LFE and rear speaker. Also I have to mute the internal speaker. I've tried to patch the module manually, but this went wrong somehow. I don't know how to proceed. So maybe you can help me and I can help you to complete the ALSA code for this special mainboard/PC - hopefully. So long, I would thank you for your feedback. Regards, Rüdiger

2 2

[alsa-devel] [PATCH - my module 1/1] Fix a memory access violation resulting from improper error propogation.
by allan＠allanwirth.com 01 Feb '13

01 Feb '13

From: Allan Wirth <allan(a)allanwirth.com> Fixes an issue where a variable is used undeclared, which can cause seg faults on some systems if the configuration file is not formatted properly. Signed-off-by: Allan Wirth <allan(a)allanwirth.com> diff --git a/src/conf.c b/src/conf.c index ecaa339..4ba5863 100644 --- a/src/conf.c +++ b/src/conf.c @@ -3530,7 +3530,7 @@ int snd_config_hook_load(snd_config_t *root, snd_config_t *config, snd_config_t if (err < 0) goto _err; } - } else if (config_file_open(root, fi[idx].name) < 0) + } else if ((err = config_file_open(root, fi[idx].name)) < 0) goto _err; } *dst = NULL; -- 1.7.10.4

2 1

[alsa-devel] [git pull] another usb-audio bugfix
by Clemens Ladisch 01 Feb '13

01 Feb '13

The following change since commit 949db153b6466c6f7cad5a427ecea94985927311: Linux 3.8-rc5 (2013-01-25 11:57:28 -0800) is available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git usb-audio-fix for you to fetch changes up to 7da58046482fceb17c4a0d4afefd9507ec56de7f: ALSA: usb-audio: fix Roland A-PRO support (2013-01-31 21:21:59 +0100) ---------------------------------------------------------------- Clemens Ladisch (1): ALSA: usb-audio: fix Roland A-PRO support sound/usb/quirks-table.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) ---------------------------------------------------------------- ALSA: usb-audio: fix Roland A-PRO support The quirk for the Roland/Cakewalk A-PRO keyboards accidentally used the wrong interface number, which prevented the driver from attaching to the device. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> Cc: 2.6.37+ <stable(a)vger.kernel.org> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 64d25a7..820580a 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1750,7 +1750,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Roland", */ /* .product_name = "A-PRO", */ - .ifnum = 1, + .ifnum = 0, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0003,

2 1

[alsa-devel] [PATCH 002/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 01 Feb '13

01 Feb '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch completes the replacement of the existing max98090 driver, by installing a more complete driver. Signed-off-by: Jerry Wong <jerry.wong(a)maximintegrated.com> Tested-by: Matthew Mowdy <matthew.mowdy(a)maximintegrated.com> Reviewed-by: Ralph Birt <ralph.birt(a)maximintegrated.com> --- Mark, I have rebased and remade the patches. Jerry include/sound/max98090.h | 29 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98090.c | 2398 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98090.h | 1549 ++++++++++++++++++++++++++++ 5 files changed, 3982 insertions(+), 0 deletions(-) create mode 100755 include/sound/max98090.h create mode 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h diff --git a/include/sound/max98090.h b/include/sound/max98090.h new file mode 100755 index 0000000..95efb13 --- /dev/null +++ b/include/sound/max98090.h @@ -0,0 +1,29 @@ +/* + * Platform data for MAX98090 + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SOUND_MAX98090_PDATA_H__ +#define __SOUND_MAX98090_PDATA_H__ + +/* codec platform data */ +struct max98090_pdata { + + /* Analog/digital microphone configuration: + * 0 = analog microphone input (normal setting) + * 1 = digital microphone input + */ + unsigned int digmic_left_mode:1; + unsigned int digmic_right_mode:1; + unsigned int digmic_3_mode:1; + unsigned int digmic_4_mode:1; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0e368d4..3a84782 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -267,6 +268,9 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate +config SND_SOC_MAX98090 + tristate + config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aa56312..f6e8e36 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o +snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -157,6 +158,7 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c new file mode 100755 index 0000000..590ac94 --- /dev/null +++ b/sound/soc/codecs/max98090.c @@ -0,0 +1,2398 @@ +/* + * max98090.c -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/max98090.h> +#include "max98090.h" + +#include <linux/version.h> + +#define DEBUG +#define EXTMIC_METHOD +#define EXTMIC_METHOD_TEST + +/* Allows for sparsely populated register maps */ +static struct reg_default max98090_reg[] = { + { 0x00, 0x00 }, /* 00 Software Reset */ + { 0x03, 0x04 }, /* 03 Interrupt Masks */ + { 0x04, 0x00 }, /* 04 System Clock Quick */ + { 0x05, 0x00 }, /* 05 Sample Rate Quick */ + { 0x06, 0x00 }, /* 06 DAI Interface Quick */ + { 0x07, 0x00 }, /* 07 DAC Path Quick */ + { 0x08, 0x00 }, /* 08 Mic/Direct to ADC Quick */ + { 0x09, 0x00 }, /* 09 Line to ADC Quick */ + { 0x0A, 0x00 }, /* 0A Analog Mic Loop Quick */ + { 0x0B, 0x00 }, /* 0B Analog Line Loop Quick */ + { 0x0C, 0x00 }, /* 0C Reserved */ + { 0x0D, 0x00 }, /* 0D Input Config */ + { 0x0E, 0x1B }, /* 0E Line Input Level */ + { 0x0F, 0x00 }, /* 0F Line Config */ + + { 0x10, 0x14 }, /* 10 Mic1 Input Level */ + { 0x11, 0x14 }, /* 11 Mic2 Input Level */ + { 0x12, 0x00 }, /* 12 Mic Bias Voltage */ + { 0x13, 0x00 }, /* 13 Digital Mic Config */ + { 0x14, 0x00 }, /* 14 Digital Mic Mode */ + { 0x15, 0x00 }, /* 15 Left ADC Mixer */ + { 0x16, 0x00 }, /* 16 Right ADC Mixer */ + { 0x17, 0x03 }, /* 17 Left ADC Level */ + { 0x18, 0x03 }, /* 18 Right ADC Level */ + { 0x19, 0x00 }, /* 19 ADC Biquad Level */ + { 0x1A, 0x00 }, /* 1A ADC Sidetone */ + { 0x1B, 0x00 }, /* 1B System Clock */ + { 0x1C, 0x00 }, /* 1C Clock Mode */ + { 0x1D, 0x00 }, /* 1D Any Clock 1 */ + { 0x1E, 0x00 }, /* 1E Any Clock 2 */ + { 0x1F, 0x00 }, /* 1F Any Clock 3 */ + + { 0x20, 0x00 }, /* 20 Any Clock 4 */ + { 0x21, 0x00 }, /* 21 Master Mode */ + { 0x22, 0x00 }, /* 22 Interface Format */ + { 0x23, 0x00 }, /* 23 TDM Format 1*/ + { 0x24, 0x00 }, /* 24 TDM Format 2*/ + { 0x25, 0x00 }, /* 25 I/O Configuration */ + { 0x26, 0x80 }, /* 26 Filter Config */ + { 0x27, 0x00 }, /* 27 DAI Playback Level */ + { 0x28, 0x00 }, /* 28 EQ Playback Level */ + { 0x29, 0x00 }, /* 29 Left HP Mixer */ + { 0x2A, 0x00 }, /* 2A Right HP Mixer */ + { 0x2B, 0x00 }, /* 2B HP Control */ + { 0x2C, 0x1A }, /* 2C Left HP Volume */ + { 0x2D, 0x1A }, /* 2D Right HP Volume */ + { 0x2E, 0x00 }, /* 2E Left Spk Mixer */ + { 0x2F, 0x00 }, /* 2F Right Spk Mixer */ + + { 0x30, 0x00 }, /* 30 Spk Control */ + { 0x31, 0x2C }, /* 31 Left Spk Volume */ + { 0x32, 0x2C }, /* 32 Right Spk Volume */ + { 0x33, 0x00 }, /* 33 ALC Timing */ + { 0x34, 0x00 }, /* 34 ALC Compressor */ + { 0x35, 0x00 }, /* 35 ALC Expander */ + { 0x36, 0x00 }, /* 36 ALC Gain */ + { 0x37, 0x00 }, /* 37 Rcv/Line OutL Mixer */ + { 0x38, 0x00 }, /* 38 Rcv/Line OutL Control */ + { 0x39, 0x15 }, /* 39 Rcv/Line OutL Volume */ + { 0x3A, 0x00 }, /* 3A Line OutR Mixer */ + { 0x3B, 0x00 }, /* 3B Line OutR Control */ + { 0x3C, 0x15 }, /* 3C Line OutR Volume */ + { 0x3D, 0x00 }, /* 3D Jack Detect */ + { 0x3E, 0x00 }, /* 3E Input Enable */ + { 0x3F, 0x00 }, /* 3F Output Enable */ + + { 0x40, 0x00 }, /* 40 Level Control */ + { 0x41, 0x00 }, /* 41 DSP Filter Enable */ + { 0x42, 0x00 }, /* 42 Bias Control */ + { 0x43, 0x00 }, /* 43 DAC Control */ + { 0x44, 0x06 }, /* 44 ADC Control */ + { 0x45, 0x00 }, /* 45 Device Shutdown */ + { 0x46, 0x00 }, /* 46 Equalizer Band 1 Coefficient B0 */ + { 0x47, 0x00 }, /* 47 Equalizer Band 1 Coefficient B0 */ + { 0x48, 0x00 }, /* 48 Equalizer Band 1 Coefficient B0 */ + { 0x49, 0x00 }, /* 49 Equalizer Band 1 Coefficient B1 */ + { 0x4A, 0x00 }, /* 4A Equalizer Band 1 Coefficient B1 */ + { 0x4B, 0x00 }, /* 4B Equalizer Band 1 Coefficient B1 */ + { 0x4C, 0x00 }, /* 4C Equalizer Band 1 Coefficient B2 */ + { 0x4D, 0x00 }, /* 4D Equalizer Band 1 Coefficient B2 */ + { 0x4E, 0x00 }, /* 4E Equalizer Band 1 Coefficient B2 */ + { 0x4F, 0x00 }, /* 4F Equalizer Band 1 Coefficient A1 */ + + { 0x50, 0x00 }, /* 50 Equalizer Band 1 Coefficient A1 */ + { 0x51, 0x00 }, /* 51 Equalizer Band 1 Coefficient A1 */ + { 0x52, 0x00 }, /* 52 Equalizer Band 1 Coefficient A2 */ + { 0x53, 0x00 }, /* 53 Equalizer Band 1 Coefficient A2 */ + { 0x54, 0x00 }, /* 54 Equalizer Band 1 Coefficient A2 */ + { 0x55, 0x00 }, /* 55 Equalizer Band 2 Coefficient B0 */ + { 0x56, 0x00 }, /* 56 Equalizer Band 2 Coefficient B0 */ + { 0x57, 0x00 }, /* 57 Equalizer Band 2 Coefficient B0 */ + { 0x58, 0x00 }, /* 58 Equalizer Band 2 Coefficient B1 */ + { 0x59, 0x00 }, /* 59 Equalizer Band 2 Coefficient B1 */ + { 0x5A, 0x00 }, /* 5A Equalizer Band 2 Coefficient B1 */ + { 0x5B, 0x00 }, /* 5B Equalizer Band 2 Coefficient B2 */ + { 0x5C, 0x00 }, /* 5C Equalizer Band 2 Coefficient B2 */ + { 0x5D, 0x00 }, /* 5D Equalizer Band 2 Coefficient B2 */ + { 0x5E, 0x00 }, /* 5E Equalizer Band 2 Coefficient A1 */ + { 0x5F, 0x00 }, /* 5F Equalizer Band 2 Coefficient A1 */ + + { 0x60, 0x00 }, /* 60 Equalizer Band 2 Coefficient A1 */ + { 0x61, 0x00 }, /* 61 Equalizer Band 2 Coefficient A2 */ + { 0x62, 0x00 }, /* 62 Equalizer Band 2 Coefficient A2 */ + { 0x63, 0x00 }, /* 63 Equalizer Band 2 Coefficient A2 */ + { 0x64, 0x00 }, /* 64 Equalizer Band 3 Coefficient B0 */ + { 0x65, 0x00 }, /* 65 Equalizer Band 3 Coefficient B0 */ + { 0x66, 0x00 }, /* 66 Equalizer Band 3 Coefficient B0 */ + { 0x67, 0x00 }, /* 67 Equalizer Band 3 Coefficient B1 */ + { 0x68, 0x00 }, /* 68 Equalizer Band 3 Coefficient B1 */ + { 0x69, 0x00 }, /* 69 Equalizer Band 3 Coefficient B1 */ + { 0x6A, 0x00 }, /* 6A Equalizer Band 3 Coefficient B2 */ + { 0x6B, 0x00 }, /* 6B Equalizer Band 3 Coefficient B2 */ + { 0x6C, 0x00 }, /* 6C Equalizer Band 3 Coefficient B2 */ + { 0x6D, 0x00 }, /* 6D Equalizer Band 3 Coefficient A1 */ + { 0x6E, 0x00 }, /* 6E Equalizer Band 3 Coefficient A1 */ + { 0x6F, 0x00 }, /* 6F Equalizer Band 3 Coefficient A1 */ + + { 0x70, 0x00 }, /* 70 Equalizer Band 3 Coefficient A2 */ + { 0x71, 0x00 }, /* 71 Equalizer Band 3 Coefficient A2 */ + { 0x72, 0x00 }, /* 72 Equalizer Band 3 Coefficient A2 */ + { 0x73, 0x00 }, /* 73 Equalizer Band 4 Coefficient B0 */ + { 0x74, 0x00 }, /* 74 Equalizer Band 4 Coefficient B0 */ + { 0x75, 0x00 }, /* 75 Equalizer Band 4 Coefficient B0 */ + { 0x76, 0x00 }, /* 76 Equalizer Band 4 Coefficient B1 */ + { 0x77, 0x00 }, /* 77 Equalizer Band 4 Coefficient B1 */ + { 0x78, 0x00 }, /* 78 Equalizer Band 4 Coefficient B1 */ + { 0x79, 0x00 }, /* 79 Equalizer Band 4 Coefficient B2 */ + { 0x7A, 0x00 }, /* 7A Equalizer Band 4 Coefficient B2 */ + { 0x7B, 0x00 }, /* 7B Equalizer Band 4 Coefficient B2 */ + { 0x7C, 0x00 }, /* 7C Equalizer Band 4 Coefficient A1 */ + { 0x7D, 0x00 }, /* 7D Equalizer Band 4 Coefficient A1 */ + { 0x7E, 0x00 }, /* 7E Equalizer Band 4 Coefficient A1 */ + { 0x7F, 0x00 }, /* 7F Equalizer Band 4 Coefficient A2 */ + + { 0x80, 0x00 }, /* 80 Equalizer Band 4 Coefficient A2 */ + { 0x81, 0x00 }, /* 81 Equalizer Band 4 Coefficient A2 */ + { 0x82, 0x00 }, /* 82 Equalizer Band 5 Coefficient B0 */ + { 0x83, 0x00 }, /* 83 Equalizer Band 5 Coefficient B0 */ + { 0x84, 0x00 }, /* 84 Equalizer Band 5 Coefficient B0 */ + { 0x85, 0x00 }, /* 85 Equalizer Band 5 Coefficient B1 */ + { 0x86, 0x00 }, /* 86 Equalizer Band 5 Coefficient B1 */ + { 0x87, 0x00 }, /* 87 Equalizer Band 5 Coefficient B1 */ + { 0x88, 0x00 }, /* 88 Equalizer Band 5 Coefficient B2 */ + { 0x89, 0x00 }, /* 89 Equalizer Band 5 Coefficient B2 */ + { 0x8A, 0x00 }, /* 8A Equalizer Band 5 Coefficient B2 */ + { 0x8B, 0x00 }, /* 8B Equalizer Band 5 Coefficient A1 */ + { 0x8C, 0x00 }, /* 8C Equalizer Band 5 Coefficient A1 */ + { 0x8D, 0x00 }, /* 8D Equalizer Band 5 Coefficient A1 */ + { 0x8E, 0x00 }, /* 8E Equalizer Band 5 Coefficient A2 */ + { 0x8F, 0x00 }, /* 8F Equalizer Band 5 Coefficient A2 */ + + { 0x90, 0x00 }, /* 90 Equalizer Band 5 Coefficient A2 */ + { 0x91, 0x00 }, /* 91 Equalizer Band 6 Coefficient B0 */ + { 0x92, 0x00 }, /* 92 Equalizer Band 6 Coefficient B0 */ + { 0x93, 0x00 }, /* 93 Equalizer Band 6 Coefficient B0 */ + { 0x94, 0x00 }, /* 94 Equalizer Band 6 Coefficient B1 */ + { 0x95, 0x00 }, /* 95 Equalizer Band 6 Coefficient B1 */ + { 0x96, 0x00 }, /* 96 Equalizer Band 6 Coefficient B1 */ + { 0x97, 0x00 }, /* 97 Equalizer Band 6 Coefficient B2 */ + { 0x98, 0x00 }, /* 98 Equalizer Band 6 Coefficient B2 */ + { 0x99, 0x00 }, /* 99 Equalizer Band 6 Coefficient B2 */ + { 0x9A, 0x00 }, /* 9A Equalizer Band 6 Coefficient A1 */ + { 0x9B, 0x00 }, /* 9B Equalizer Band 6 Coefficient A1 */ + { 0x9C, 0x00 }, /* 9C Equalizer Band 6 Coefficient A1 */ + { 0x9D, 0x00 }, /* 9D Equalizer Band 6 Coefficient A2 */ + { 0x9E, 0x00 }, /* 9E Equalizer Band 6 Coefficient A2 */ + { 0x9F, 0x00 }, /* 9F Equalizer Band 6 Coefficient A2 */ + + { 0xA0, 0x00 }, /* A0 Equalizer Band 7 Coefficient B0 */ + { 0xA1, 0x00 }, /* A1 Equalizer Band 7 Coefficient B0 */ + { 0xA2, 0x00 }, /* A2 Equalizer Band 7 Coefficient B0 */ + { 0xA3, 0x00 }, /* A3 Equalizer Band 7 Coefficient B1 */ + { 0xA4, 0x00 }, /* A4 Equalizer Band 7 Coefficient B1 */ + { 0xA5, 0x00 }, /* A5 Equalizer Band 7 Coefficient B1 */ + { 0xA6, 0x00 }, /* A6 Equalizer Band 7 Coefficient B2 */ + { 0xA7, 0x00 }, /* A7 Equalizer Band 7 Coefficient B2 */ + { 0xA8, 0x00 }, /* A8 Equalizer Band 7 Coefficient B2 */ + { 0xA9, 0x00 }, /* A9 Equalizer Band 7 Coefficient A1 */ + { 0xAA, 0x00 }, /* AA Equalizer Band 7 Coefficient A1 */ + { 0xAB, 0x00 }, /* AB Equalizer Band 7 Coefficient A1 */ + { 0xAC, 0x00 }, /* AC Equalizer Band 7 Coefficient A2 */ + { 0xAD, 0x00 }, /* AD Equalizer Band 7 Coefficient A2 */ + { 0xAE, 0x00 }, /* AE Equalizer Band 7 Coefficient A2 */ + { 0xAF, 0x00 }, /* AF ADC Biquad Coefficient B0 */ + + { 0xB0, 0x00 }, /* B0 ADC Biquad Coefficient B0 */ + { 0xB1, 0x00 }, /* B1 ADC Biquad Coefficient B0 */ + { 0xB2, 0x00 }, /* B2 ADC Biquad Coefficient B1 */ + { 0xB3, 0x00 }, /* B3 ADC Biquad Coefficient B1 */ + { 0xB4, 0x00 }, /* B4 ADC Biquad Coefficient B1 */ + { 0xB5, 0x00 }, /* B5 ADC Biquad Coefficient B2 */ + { 0xB6, 0x00 }, /* B6 ADC Biquad Coefficient B2 */ + { 0xB7, 0x00 }, /* B7 ADC Biquad Coefficient B2 */ + { 0xB8, 0x00 }, /* B8 ADC Biquad Coefficient A1 */ + { 0xB9, 0x00 }, /* B9 ADC Biquad Coefficient A1 */ + { 0xBA, 0x00 }, /* BA ADC Biquad Coefficient A1 */ + { 0xBB, 0x00 }, /* BB ADC Biquad Coefficient A2 */ + { 0xBC, 0x00 }, /* BC ADC Biquad Coefficient A2 */ + { 0xBD, 0x00 }, /* BD ADC Biquad Coefficient A2 */ + { 0xBE, 0x00 }, /* BE Digital Mic 3 Volume */ + { 0xBF, 0x00 }, /* BF Digital Mic 4 Volume */ + + { 0xC0, 0x00 }, /* C0 Digital Mic 34 Biquad Pre Atten */ + { 0xC1, 0x00 }, /* C1 Record TDM Slot */ + { 0xC2, 0x00 }, /* C2 Sample Rate */ + { 0xC3, 0x00 }, /* C3 Digital Mic 34 Biquad Coefficient C3 */ + { 0xC4, 0x00 }, /* C4 Digital Mic 34 Biquad Coefficient C4 */ + { 0xC5, 0x00 }, /* C5 Digital Mic 34 Biquad Coefficient C5 */ + { 0xC6, 0x00 }, /* C6 Digital Mic 34 Biquad Coefficient C6 */ + { 0xC7, 0x00 }, /* C7 Digital Mic 34 Biquad Coefficient C7 */ + { 0xC8, 0x00 }, /* C8 Digital Mic 34 Biquad Coefficient C8 */ + { 0xC9, 0x00 }, /* C9 Digital Mic 34 Biquad Coefficient C9 */ + { 0xCA, 0x00 }, /* CA Digital Mic 34 Biquad Coefficient CA */ + { 0xCB, 0x00 }, /* CB Digital Mic 34 Biquad Coefficient CB */ + { 0xCC, 0x00 }, /* CC Digital Mic 34 Biquad Coefficient CC */ + { 0xCD, 0x00 }, /* CD Digital Mic 34 Biquad Coefficient CD */ + { 0xCE, 0x00 }, /* CE Digital Mic 34 Biquad Coefficient CE */ + { 0xCF, 0x00 }, /* CF Digital Mic 34 Biquad Coefficient CF */ + + { 0xD0, 0x00 }, /* D0 Digital Mic 34 Biquad Coefficient D0 */ + { 0xD1, 0x00 }, /* D1 Digital Mic 34 Biquad Coefficient D1 */ +}; + +static bool max98090_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_REVISION_ID: + return true; + default: + return false; + } +} + +static bool max98090_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_INTERRUPT_S: + case M98090_REG_RESERVED: + case M98090_REG_LINE_INPUT_CONFIG: + case M98090_REG_LINE_INPUT_LEVEL: + case M98090_REG_INPUT_MODE: + case M98090_REG_MIC1_INPUT_LEVEL: + case M98090_REG_MIC2_INPUT_LEVEL: + case M98090_REG_MIC_BIAS_VOLTAGE: + case M98090_REG_DIGITAL_MIC_ENABLE: + case M98090_REG_DIGITAL_MIC_CONFIG: + case M98090_REG_LEFT_ADC_MIXER: + case M98090_REG_RIGHT_ADC_MIXER: + case M98090_REG_LEFT_ADC_LEVEL: + case M98090_REG_RIGHT_ADC_LEVEL: + case M98090_REG_ADC_BIQUAD_LEVEL: + case M98090_REG_ADC_SIDETONE: + case M98090_REG_SYSTEM_CLOCK: + case M98090_REG_CLOCK_MODE: + case M98090_REG_CLOCK_RATIO_NI_MSB: + case M98090_REG_CLOCK_RATIO_NI_LSB: + case M98090_REG_CLOCK_RATIO_MI_MSB: + case M98090_REG_CLOCK_RATIO_MI_LSB: + case M98090_REG_MASTER_MODE: + case M98090_REG_INTERFACE_FORMAT: + case M98090_REG_TDM_CONTROL: + case M98090_REG_TDM_FORMAT: + case M98090_REG_IO_CONFIGURATION: + case M98090_REG_FILTER_CONFIG: + case M98090_REG_DAI_PLAYBACK_LEVEL: + case M98090_REG_DAI_PLAYBACK_LEVEL_EQ: + case M98090_REG_LEFT_HP_MIXER: + case M98090_REG_RIGHT_HP_MIXER: + case M98090_REG_HP_CONTROL: + case M98090_REG_LEFT_HP_VOLUME: + case M98090_REG_RIGHT_HP_VOLUME: + case M98090_REG_LEFT_SPK_MIXER: + case M98090_REG_RIGHT_SPK_MIXER: + case M98090_REG_SPK_CONTROL: + case M98090_REG_LEFT_SPK_VOLUME: + case M98090_REG_RIGHT_SPK_VOLUME: + case M98090_REG_DRC_TIMING: + case M98090_REG_DRC_COMPRESSOR: + case M98090_REG_DRC_EXPANDER: + case M98090_REG_DRC_GAIN: + case M98090_REG_RCV_LOUTL_MIXER: + case M98090_REG_RCV_LOUTL_CONTROL: + case M98090_REG_RCV_LOUTL_VOLUME: + case M98090_REG_LOUTR_MIXER: + case M98090_REG_LOUTR_CONTROL: + case M98090_REG_LOUTR_VOLUME: + case M98090_REG_JACK_DETECT: + case M98090_REG_INPUT_ENABLE: + case M98090_REG_OUTPUT_ENABLE: + case M98090_REG_LEVEL_CONTROL: + case M98090_REG_DSP_FILTER_ENABLE: + case M98090_REG_BIAS_CONTROL: + case M98090_REG_DAC_CONTROL: + case M98090_REG_ADC_CONTROL: + case M98090_REG_DEVICE_SHUTDOWN: + case M98090_REG_EQUALIZER_BASE ... M98090_REG_EQUALIZER_BASE + 0x68: + case M98090_REG_RECORD_BIQUAD_BASE ... M98090_REG_RECORD_BIQUAD_BASE + 0x0E: + case M98090_REG_DMIC3_VOLUME: + case M98090_REG_DMIC4_VOLUME: + case M98090_REG_DMIC34_BQ_PREATTEN: + case M98090_REG_RECORD_TDM_SLOT: + case M98090_REG_SAMPLE_RATE: + case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + return true; + default: + return false; + } +} + +static int max98090_reset(struct max98090_priv *max98090) +{ + int ret; + + /* Reset the codec by writing to this write-only reset register */ + ret = regmap_write(max98090->regmap, M98090_REG_SOFTWARE_RESET, + M98090_SWRESET_MASK); + if (ret < 0) { + dev_err(max98090->codec->dev, + "Failed to reset codec: %d\n", ret); + return ret; + } + + msleep(20); + return ret; +} + +static const unsigned int max98090_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_mic_tlv, 0, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_line_single_ended_tlv, + -600, 600, 0); + +static const unsigned int max98090_line_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-600, 300, 0), + 4, 5, TLV_DB_SCALE_ITEM(1400, 600, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_avg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_av_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_dv_tlv, -1500, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_sidetone_tlv, -6050, 200, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_alc_tlv, -1500, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcmakeup_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alccomp_tlv, -3100, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_drcexp_tlv, -6600, 100, 0); + +static const unsigned int max98090_mixout_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(-1200, 250, 0), + 2, 3, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +static const unsigned int max98090_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98090_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 4, TLV_DB_SCALE_ITEM(-4800, 400, 0), + 5, 10, TLV_DB_SCALE_ITEM(-2900, 300, 0), + 11, 14, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 15, 29, TLV_DB_SCALE_ITEM(-500, 100, 0), + 30, 39, TLV_DB_SCALE_ITEM(950, 50, 0), +}; + +static const unsigned int max98090_rcv_lout_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + if (val >= 1) { + /* If on, return the volume */ + val = val - 1; + *select = val; + } else { + /* If off, return last stored value */ + val = *select; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + *select = sel; + + /* Setting a volume is only valid if it is already On */ + if (val >= 1) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, mc->reg, + mask << mc->shift, + sel << mc->shift); + + return 0; +} + +static const char * max98090_perf_pwr_text[] = + { "High Performance", "Low Power" }; +static const char * max98090_pwr_perf_text[] = + { "Low Power", "High Performance" }; + +static const struct soc_enum max98090_vcmbandgap_enum = + SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; + +static const struct soc_enum max98090_osr128_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, + ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); + +static const char *max98090_mode_text[] = { "Voice", "Music" }; + +static const struct soc_enum max98090_mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const struct soc_enum max98090_filter_dmic34mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const char * max98090_drcatk_text[] = + { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; + +static const struct soc_enum max98090_drcatk_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, + ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); + +static const char * max98090_drcrls_text[] = + { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; + +static const struct soc_enum max98090_drcrls_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, + ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); + +static const char * max98090_alccmp_text[] = + { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; + +static const struct soc_enum max98090_alccmp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, + ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); + +static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; + +static const struct soc_enum max98090_drcexp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, + ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text); + +static const struct soc_enum max98090_dac_perfmode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT, + ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); + +static const struct soc_enum max98090_dachp_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_adchp_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct snd_kcontrol_new max98090_snd_controls[] = { + SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum), + + SOC_SINGLE("DMIC MIC Comp Filter Config", M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0), + + SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", + M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + M98090_MIC_PA1EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_EXT_TLV("MIC2 Boost Volume", + M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + M98090_MIC_PA2EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_TLV("MIC1 Volume", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PGAM1_SHIFT, M98090_MIC_PGAM1_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_TLV("MIC2 Volume", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PGAM2_SHIFT, M98090_MIC_PGAM2_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG135_SHIFT, 0, + M98090_MIXG135_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG246_SHIFT, 0, + M98090_MIXG246_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINAPGA_SHIFT, 0, M98090_LINAPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINBPGA_SHIFT, 0, M98090_LINBPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE("LINEA Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFA_SHIFT, M98090_EXTBUFA_NUM - 1, 0), + SOC_SINGLE("LINEB Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFB_SHIFT, M98090_EXTBUFB_NUM - 1, 0), + + SOC_SINGLE_TLV("ADCL Boost Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVLG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("ADCR Boost Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVRG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("ADCL Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVL_SHIFT, M98090_AVL_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("ADCR Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVR_SHIFT, M98090_AVR_NUM - 1, 1, + max98090_av_tlv), + + SOC_ENUM("ADC Oversampling Rate", max98090_osr128_enum), + SOC_SINGLE("ADC Quantizer Dither", M98090_REG_ADC_CONTROL, + M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0), + SOC_ENUM("ADC High Performance Mode", max98090_adchp_enum), + + SOC_SINGLE("DAC Mono Mode", M98090_REG_IO_CONFIGURATION, + M98090_DMONO_SHIFT, M98090_DMONO_NUM - 1, 0), + SOC_SINGLE("SDIN Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, M98090_SDIEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, M98090_SDOEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Hi-Z Mode", M98090_REG_IO_CONFIGURATION, + M98090_HIZOFF_SHIFT, M98090_HIZOFF_NUM - 1, 1), + SOC_ENUM("Filter Mode", max98090_mode_enum), + SOC_SINGLE("Record Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, M98090_AHPF_NUM - 1, 0), + SOC_SINGLE("Playback Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_DHPF_SHIFT, M98090_DHPF_NUM - 1, 0), + SOC_SINGLE_TLV("Digital BQ Volume", M98090_REG_ADC_BIQUAD_LEVEL, + M98090_AVBQ_SHIFT, M98090_AVBQ_NUM - 1, 1, max98090_dv_tlv), + SOC_SINGLE_EXT_TLV("Digital Sidetone Volume", + M98090_REG_ADC_SIDETONE, M98090_DVST_SHIFT, + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + SOC_SINGLE_TLV("Digital Coarse Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVG_SHIFT, M98090_DVG_NUM - 1, 0, + max98090_dvg_tlv), + SOC_SINGLE_TLV("Digital Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DV_SHIFT, M98090_DV_NUM - 1, 1, + max98090_dv_tlv), + SND_SOC_BYTES("EQ Coefficients", M98090_REG_EQUALIZER_BASE, 105), + SOC_SINGLE("Digital EQ 3 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 5 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 7 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ Clipping Detection", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_EQCLPN_SHIFT, M98090_EQCLPN_NUM - 1, + 1), + SOC_SINGLE_TLV("Digital EQ Volume", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_DVEQ_SHIFT, M98090_DVEQ_NUM - 1, 1, + max98090_dv_tlv), + + SOC_SINGLE("ALC Enable", M98090_REG_DRC_TIMING, + M98090_DRCEN_SHIFT, M98090_DRCEN_NUM - 1, 0), + SOC_ENUM("ALC Attack Time", max98090_drcatk_enum), + SOC_ENUM("ALC Release Time", max98090_drcrls_enum), + SOC_SINGLE_TLV("ALC Make Up Volume", M98090_REG_DRC_GAIN, + M98090_DRCG_SHIFT, M98090_DRCG_NUM - 1, 0, + max98090_alcmakeup_tlv), + SOC_ENUM("ALC Compression Ratio", max98090_alccmp_enum), + SOC_ENUM("ALC Expansion Ratio", max98090_drcexp_enum), + SOC_SINGLE_TLV("ALC Compression Threshold Volume", + M98090_REG_DRC_COMPRESSOR, M98090_DRCTHC_SHIFT, + M98090_DRCTHC_NUM - 1, 1, max98090_alccomp_tlv), + SOC_SINGLE_TLV("ALC Expansion Threshold Volume", + M98090_REG_DRC_EXPANDER, M98090_DRCTHE_SHIFT, + M98090_DRCTHE_NUM - 1, 1, max98090_drcexp_tlv), + + SOC_ENUM("DAC HP Playback Performance Mode", + max98090_dac_perfmode_enum), + SOC_ENUM("DAC High Performance Mode", max98090_dachp_enum), + + SOC_SINGLE_TLV("Headphone Left Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPLG_SHIFT, + M98090_MIXHPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Headphone Right Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPRG_SHIFT, + M98090_MIXHPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Speaker Left Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPLG_SHIFT, + M98090_MIXSPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Speaker Right Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPRG_SHIFT, + M98090_MIXSPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Receiver Left Mixer Volume", + M98090_REG_RCV_LOUTL_CONTROL, M98090_MIXRCVLG_SHIFT, + M98090_MIXRCVLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Receiver Right Mixer Volume", + M98090_REG_LOUTR_CONTROL, M98090_MIXRCVRG_SHIFT, + M98090_MIXRCVRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_DOUBLE_R_TLV("Headphone Volume", M98090_REG_LEFT_HP_VOLUME, + M98090_REG_RIGHT_HP_VOLUME, M98090_HPVOLL_SHIFT, + M98090_HPVOLL_NUM - 1, 0, max98090_hp_tlv), + + SOC_DOUBLE_R_RANGE_TLV("Speaker Volume", + M98090_REG_LEFT_SPK_VOLUME, M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPVOLL_SHIFT, 24, M98090_SPVOLL_NUM - 1 + 24, + 0, max98090_spk_tlv), + + SOC_DOUBLE_R_TLV("Receiver Volume", M98090_REG_RCV_LOUTL_VOLUME, + M98090_REG_LOUTR_VOLUME, M98090_RCVLVOL_SHIFT, + M98090_RCVLVOL_NUM - 1, 0, max98090_rcv_lout_tlv), + + SOC_SINGLE("Headphone Left Switch", M98090_REG_LEFT_HP_VOLUME, + M98090_HPLM_SHIFT, 1, 1), + SOC_SINGLE("Headphone Right Switch", M98090_REG_RIGHT_HP_VOLUME, + M98090_HPRM_SHIFT, 1, 1), + + SOC_SINGLE("Speaker Left Switch", M98090_REG_LEFT_SPK_VOLUME, + M98090_SPLM_SHIFT, 1, 1), + SOC_SINGLE("Speaker Right Switch", M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPRM_SHIFT, 1, 1), + + SOC_SINGLE("Receiver Left Switch", M98090_REG_RCV_LOUTL_VOLUME, + M98090_RCVLM_SHIFT, 1, 1), + SOC_SINGLE("Receiver Right Switch", M98090_REG_LOUTR_VOLUME, + M98090_RCVRM_SHIFT, 1, 1), + + SOC_SINGLE("Zero-Crossing Detection", M98090_REG_LEVEL_CONTROL, + M98090_ZDENN_SHIFT, M98090_ZDENN_NUM - 1, 1), + SOC_SINGLE("Enhanced Vol Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VS2ENN_SHIFT, M98090_VS2ENN_NUM - 1, 1), + SOC_SINGLE("Volume Adjustment Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VSENN_SHIFT, M98090_VSENN_NUM - 1, 1), + + SND_SOC_BYTES("Biquad Coefficients", M98090_REG_RECORD_BIQUAD_BASE, 15), + SOC_SINGLE("Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0), +}; + +static const struct snd_kcontrol_new max98091_snd_controls[] = { + + SOC_SINGLE("DMIC34 Zeropad", M98090_REG_SAMPLE_RATE, + M98090_DMIC34_ZEROPAD_SHIFT, + M98090_DMIC34_ZEROPAD_NUM - 1, 0), + + SOC_ENUM("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum), + SOC_SINGLE("DMIC34 DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34HPF_SHIFT, + M98090_FLT_DMIC34HPF_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC3 Boost Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3G_SHIFT, M98090_DMIC_AV3G_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("DMIC4 Boost Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4G_SHIFT, M98090_DMIC_AV4G_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("DMIC3 Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3_SHIFT, M98090_DMIC_AV3_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("DMIC4 Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4_SHIFT, M98090_DMIC_AV4_NUM - 1, 1, + max98090_av_tlv), + + SND_SOC_BYTES("DMIC34 Biquad Coefficients", + M98090_REG_DMIC34_BIQUAD_BASE, 15), + SOC_SINGLE("DMIC34 Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC34 BQ PreAttenuation Volume", + M98090_REG_DMIC34_BQ_PREATTEN, M98090_AV34BQ_SHIFT, + M98090_AV34BQ_NUM - 1, 1, max98090_dv_tlv), +}; + +static int max98090_micinput_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, w->reg); + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = (val & M98090_MIC_PA1EN_MASK) >> M98090_MIC_PA1EN_SHIFT; + else + val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; + + + if (val >= 1) { + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { + max98090->pa1en = val - 1; /* Update for volatile */ + } else { + max98090->pa2en = val - 1; /* Update for volatile */ + } + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to most recently selected volume */ + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = max98090->pa1en + 1; + else + val = max98090->pa2en + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = 0; + break; + default: + return -EINVAL; + } + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA1EN_MASK, + val << M98090_MIC_PA1EN_SHIFT); + else + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA2EN_MASK, + val << M98090_MIC_PA2EN_SHIFT); + + return 0; +} + +static const char *mic1_mux_text[] = { "IN12", "IN56" }; + +static const struct soc_enum mic1_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT, + ARRAY_SIZE(mic1_mux_text), mic1_mux_text); + +static const struct snd_kcontrol_new max98090_mic1_mux = + SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); + +static const char *mic2_mux_text[] = { "IN34", "IN56" }; + +static const struct soc_enum mic2_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT, + ARRAY_SIZE(mic2_mux_text), mic2_mux_text); + +static const struct snd_kcontrol_new max98090_mic2_mux = + SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); + +static const char * max98090_micpre_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_pa1en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const struct soc_enum max98090_pa2en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +/* LINEA mixer switch */ +static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN1SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN3 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN3SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN5 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN5SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN34DIFF_SHIFT, 1, 0), +}; + +/* LINEB mixer switch */ +static const struct snd_kcontrol_new max98090_lineb_mixer_controls[] = { + SOC_DAPM_SINGLE("IN2 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN2SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN4 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN4SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN6 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN6SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN56DIFF_SHIFT, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98090_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC2_SHIFT, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC2_SHIFT, 1, 0), +}; + +static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; + +static const struct soc_enum ltenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct soc_enum ltenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct snd_kcontrol_new max98090_ltenl_mux = + SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); + +static const struct snd_kcontrol_new max98090_ltenr_mux = + SOC_DAPM_ENUM("LTENR Mux", ltenr_mux_enum); + +static const char *lben_mux_text[] = { "Normal", "Loopback" }; + +static const struct soc_enum lbenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct soc_enum lbenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct snd_kcontrol_new max98090_lbenl_mux = + SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); + +static const struct snd_kcontrol_new max98090_lbenr_mux = + SOC_DAPM_ENUM("LBENR Mux", lbenr_mux_enum); + +static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" }; + +static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" }; + +static const struct soc_enum stenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT, + ARRAY_SIZE(stenl_mux_text), stenl_mux_text); + +static const struct soc_enum stenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT, + ARRAY_SIZE(stenr_mux_text), stenr_mux_text); + +static const struct snd_kcontrol_new max98090_stenl_mux = + SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum); + +static const struct snd_kcontrol_new max98090_stenr_mux = + SOC_DAPM_ENUM("STENR Mux", stenr_mux_enum); + +/* Left speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC2_SHIFT, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC2_SHIFT, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC2_SHIFT, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC2_SHIFT, 1, 0), +}; + +/* Left receiver mixer switch */ +static const struct snd_kcontrol_new max98090_left_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC2_SHIFT, 1, 0), +}; + +/* Right receiver mixer switch */ +static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC2_SHIFT, 1, 0), +}; + +static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; + +static const struct soc_enum linmod_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT, + ARRAY_SIZE(linmod_mux_text), linmod_mux_text); + +static const struct snd_kcontrol_new max98090_linmod_mux = + SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); + +static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; + +/* + * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable + */ +static const struct soc_enum mixhplsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhplsel_mux = + SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); + +static const struct soc_enum mixhprsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhprsel_mux = + SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); + +static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMICL"), + SND_SOC_DAPM_INPUT("DMICR"), + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN12"), + SND_SOC_DAPM_INPUT("IN34"), + SND_SOC_DAPM_INPUT("IN56"), + + SND_SOC_DAPM_SUPPLY("MICBIAS", M98090_REG_INPUT_ENABLE, + M98090_MBEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDIEN", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICL_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICR_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, 0, NULL, 0), + +/* + * Note: Sysclk and misc power supplies are taken care of by SHDN + */ + + SND_SOC_DAPM_MUX("MIC1 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic1_mux), + + SND_SOC_DAPM_MUX("MIC2 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic2_mux), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PA2EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("LINEA Mixer", SND_SOC_NOPM, 0, 0, + &max98090_linea_mixer_controls[0], + ARRAY_SIZE(max98090_linea_mixer_controls)), + + SND_SOC_DAPM_MIXER("LINEB Mixer", SND_SOC_NOPM, 0, 0, + &max98090_lineb_mixer_controls[0], + ARRAY_SIZE(max98090_lineb_mixer_controls)), + + SND_SOC_DAPM_PGA("LINEA Input", M98090_REG_INPUT_ENABLE, + M98090_LINEAEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("LINEB Input", M98090_REG_INPUT_ENABLE, + M98090_LINEBEN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_adc_mixer_controls[0], + ARRAY_SIZE(max98090_left_adc_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_adc_mixer_controls[0], + ARRAY_SIZE(max98090_right_adc_mixer_controls)), + + SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0), + SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", "HiFi Capture", 1, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("LBENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenl_mux), + + SND_SOC_DAPM_MUX("LBENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenr_mux), + + SND_SOC_DAPM_MUX("LTENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenl_mux), + + SND_SOC_DAPM_MUX("LTENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenr_mux), + + SND_SOC_DAPM_MUX("STENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenl_mux), + + SND_SOC_DAPM_MUX("STENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenr_mux), + + SND_SOC_DAPM_AIF_IN("AIFINL", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", "HiFi Playback", 1, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DACL", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DALEN_SHIFT, 0), + SND_SOC_DAPM_DAC("DACR", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DAREN_SHIFT, 0), + + SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_hp_mixer_controls[0], + ARRAY_SIZE(max98090_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_hp_mixer_controls[0], + ARRAY_SIZE(max98090_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_left_rcv_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_right_rcv_mixer_controls)), + + SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, + M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + + SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("RCV Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("RCV Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RCVL"), + SND_SOC_DAPM_OUTPUT("RCVR"), +}; + +static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + + SND_SOC_DAPM_SUPPLY("DMIC3_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC3_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC4_SHIFT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { + + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, + + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, + {"DMICL", NULL, "AHPF"}, + {"DMICR", NULL, "AHPF"}, + + /* MIC1 input mux */ + {"MIC1 Mux", "IN12", "IN12"}, + {"MIC1 Mux", "IN56", "IN56"}, + + /* MIC2 input mux */ + {"MIC2 Mux", "IN34", "IN34"}, + {"MIC2 Mux", "IN56", "IN56"}, + + {"MIC1 Input", NULL, "MIC1 Mux"}, + {"MIC2 Input", NULL, "MIC2 Mux"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "IN12 Switch", "IN12"}, + {"Left ADC Mixer", "IN34 Switch", "IN34"}, + {"Left ADC Mixer", "IN56 Switch", "IN56"}, + {"Left ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Left ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "IN12 Switch", "IN12"}, + {"Right ADC Mixer", "IN34 Switch", "IN34"}, + {"Right ADC Mixer", "IN56 Switch", "IN56"}, + {"Right ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Right ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Line A input mixer */ + {"LINEA Mixer", "IN1 Switch", "IN1"}, + {"LINEA Mixer", "IN3 Switch", "IN3"}, + {"LINEA Mixer", "IN5 Switch", "IN5"}, + {"LINEA Mixer", "IN34 Switch", "IN34"}, + + /* Line B input mixer */ + {"LINEB Mixer", "IN2 Switch", "IN2"}, + {"LINEB Mixer", "IN4 Switch", "IN4"}, + {"LINEB Mixer", "IN6 Switch", "IN6"}, + {"LINEB Mixer", "IN56 Switch", "IN56"}, + + {"LINEA Input", NULL, "LINEA Mixer"}, + {"LINEB Input", NULL, "LINEB Mixer"}, + + /* Inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + {"ADCL", NULL, "SHDN"}, + {"ADCR", NULL, "SHDN"}, + + {"LBENL Mux", "Normal", "ADCL"}, + {"LBENL Mux", "Normal", "DMICL"}, + {"LBENL Mux", "Loopback", "LTENL Mux"}, + {"LBENR Mux", "Normal", "ADCR"}, + {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Loopback", "LTENR Mux"}, + + {"AIFOUTL", NULL, "LBENL Mux"}, + {"AIFOUTR", NULL, "LBENR Mux"}, + {"AIFOUTL", NULL, "SHDN"}, + {"AIFOUTR", NULL, "SHDN"}, + {"AIFOUTL", NULL, "SDOEN"}, + {"AIFOUTR", NULL, "SDOEN"}, + + {"LTENL Mux", "Normal", "AIFINL"}, + {"LTENL Mux", "Loopthrough", "LBENL Mux"}, + {"LTENR Mux", "Normal", "AIFINR"}, + {"LTENR Mux", "Loopthrough", "LBENR Mux"}, + + {"DACL", NULL, "LTENL Mux"}, + {"DACR", NULL, "LTENR Mux"}, + + {"STENL Mux", "Sidetone Left", "ADCL"}, + {"STENL Mux", "Sidetone Left", "DMICL"}, + {"STENR Mux", "Sidetone Right", "ADCR"}, + {"STENR Mux", "Sidetone Right", "DMICR"}, + {"DACL", "NULL", "STENL Mux"}, + {"DACR", "NULL", "STENL Mux"}, + + {"AIFINL", NULL, "SHDN"}, + {"AIFINR", NULL, "SHDN"}, + {"AIFINL", NULL, "SDIEN"}, + {"AIFINR", NULL, "SDIEN"}, + {"DACL", NULL, "SHDN"}, + {"DACR", NULL, "SHDN"}, + + /* Left headphone output mixer */ + {"Left Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Left Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Left Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right headphone output mixer */ + {"Right Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Right Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Right Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left speaker output mixer */ + {"Left Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Left Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Left Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right speaker output mixer */ + {"Right Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Right Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Right Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left Receiver output mixer */ + {"Left Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Left Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Left Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right Receiver output mixer */ + {"Right Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Right Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Right Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + {"MIXHPLSEL Mux", "HP Mixer", "Left Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Left Out", NULL, "DACL"}, + {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + + {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Right Out", NULL, "DACR"}, + {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + + {"SPK Left Out", NULL, "Left Speaker Mixer"}, + {"SPK Right Out", NULL, "Right Speaker Mixer"}, + {"RCV Left Out", NULL, "Left Receiver Mixer"}, + + {"LINMOD Mux", "Left and Right", "Right Receiver Mixer"}, + {"LINMOD Mux", "Left Only", "Left Receiver Mixer"}, + {"RCV Right Out", NULL, "LINMOD Mux"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RCVL", NULL, "RCV Left Out"}, + {"RCVR", NULL, "RCV Right Out"}, + +}; + +static const struct snd_soc_dapm_route max98091_dapm_routes[] = { + + /* DMIC inputs */ + {"DMIC3", NULL, "DMIC3_ENA"}, + {"DMIC4", NULL, "DMIC4_ENA"}, + {"DMIC3", NULL, "AHPF"}, + {"DMIC4", NULL, "AHPF"}, + +}; + +static int max98090_add_widgets(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_add_codec_controls(codec, max98090_snd_controls, + ARRAY_SIZE(max98090_snd_controls)); + + if (max98090->devtype == MAX98091) { + snd_soc_add_codec_controls(codec, max98091_snd_controls, + ARRAY_SIZE(max98091_snd_controls)); + } + + snd_soc_dapm_new_controls(dapm, max98090_dapm_widgets, + ARRAY_SIZE(max98090_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98090_dapm_routes, + ARRAY_SIZE(max98090_dapm_routes)); + + if (max98090->devtype == MAX98091) { + snd_soc_dapm_new_controls(dapm, max98091_dapm_widgets, + ARRAY_SIZE(max98091_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, + ARRAY_SIZE(max98091_dapm_routes)); + + } + + return 0; +} + +static const int pclk_rates[] = { + 12000000, 12000000, 13000000, 13000000, + 16000000, 16000000, 19200000, 19200000 +}; + +static const int lrclk_rates[] = { + 8000, 16000, 8000, 16000, + 8000, 16000, 8000, 16000 +}; + +static const int user_pclk_rates[] = { + 13000000, 13000000 +}; + +static const int user_lrclk_rates[] = { + 44100, 48000 +}; + +static const unsigned long long ni_value[] = { + 3528, 768 +}; + +static const unsigned long long mi_value[] = { + 8125, 1625 +}; + +static void max98090_configure_bclk(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned long long ni; + int i; + + if (!max98090->sysclk) { + dev_err(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!max98090->bclk || !max98090->lrclk) { + dev_err(codec->dev, "No audio clocks configured\n"); + return; + } + + /* Skip configuration when operating as slave */ + if (!(snd_soc_read(codec, M98090_REG_MASTER_MODE) & + M98090_MAS_MASK)) { + return; + } + + /* Check for supported PCLK to LRCLK ratios */ + for (i = 0; i < ARRAY_SIZE(pclk_rates); i++) { + if ((pclk_rates[i] == max98090->sysclk) && + (lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found supported PCLK to LRCLK rates 0x%x\n", + i + 0x8); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, + (i + 0x8) << M98090_FREQ_SHIFT); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + return; + } + } + + /* Check for user calculated MI and NI ratios */ + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { + if ((user_pclk_rates[i] == max98090->sysclk) && + (user_lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found user supported PCLK to LRCLK rates\n"); + dev_dbg(codec->dev, "i %d ni %lld mi %lld\n", + i, ni_value[i], mi_value[i]); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, + 1 << M98090_USE_M1_SHIFT); + + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, + ni_value[i] & 0xFF); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_MSB, + (mi_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_LSB, + mi_value[i] & 0xFF); + + return; + } + } + + /* + * Calculate based on MI = 65536 (not as good as either method above) + */ + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + + /* + * Configure NI when operating as master + * Note: There is a small, but significant audio quality improvement + * by calculating ni and mi. + */ + ni = 65536ULL * (max98090->lrclk < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)max98090->lrclk; + do_div(ni, (unsigned long long int)max98090->sysclk); + dev_info(codec->dev, "No better method found\n"); + dev_info(codec->dev, "Calculating ni %lld with mi 65536\n", ni); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, ni & 0xFF); +} + +static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + u8 regval; + + max98090->dai_fmt = fmt; + cdata = &max98090->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Set to slave mode PLL - MAS mode off */ + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_MSB, 0x00); + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_LSB, 0x00); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + if (max98090->tdm_slots == 4) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_64; + } else if (max98090->tdm_slots == 3) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_48; + } else { + /* Few TDM slots, or No TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_32; + } + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + snd_soc_write(codec, M98090_REG_MASTER_MODE, regval); + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98090_DLY_MASK; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regval |= M98090_RJ_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Not supported mode */ + default: + dev_err(codec->dev, "DAI format unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98090_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98090_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98090_BCI_MASK|M98090_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + /* + * This accommodates an inverted logic in the MAX98090 chip + * for Bit Clock Invert (BCI). The inverted logic is only + * seen for the case of TDM mode. The remaining cases have + * normal logic. + */ + if (max98090->tdm_slots > 1) { + regval ^= M98090_BCI_MASK; + } + + snd_soc_write(codec, + M98090_REG_INTERFACE_FORMAT, regval); + } + + return 0; +} + +static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + cdata = &max98090->dai[0]; + + if (slots < 0 || slots > 4) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_width = slot_width; + + if (max98090->tdm_slots > 1) { + /* SLOTL SLOTR SLOTDLY */ + snd_soc_write(codec, M98090_REG_TDM_FORMAT, + 0 << M98090_TDM_SLOTL_SHIFT | + 1 << M98090_TDM_SLOTR_SHIFT | + 0 << M98090_TDM_SLOTDLY_SHIFT); + + /* FSW TDM */ + snd_soc_update_bits(codec, M98090_REG_TDM_CONTROL, + M98090_TDM_MASK, + M98090_TDM_MASK); + } + + /* + * Normally advisable to set TDM first, but this permits either order + */ + cdata->fmt = 0; + max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + + return 0; +} + +static int max98090_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { + /* + * Set to normal bias level. + */ + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, + M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + } + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + /* Set internal pull-up to lowest power mode */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + regcache_mark_dirty(max98090->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const int comp_pclk_rates[] = { + 11289600, 12288000, 12000000, 13000000, 19200000 +}; + +static const int dmic_micclk[] = { + 2, 2, 2, 2, 4, 2 +}; + +static const int comp_lrclk_rates[] = { + 8000, 16000, 32000, 44100, 48000, 96000 +}; + +static const int dmic_comp[6][6] = { + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 1, 1, 1}, + {7, 8, 3, 1, 2, 2}, + {7, 8, 3, 3, 3, 3} +}; + +static int max98090_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int i, j; + + cdata = &max98090->dai[0]; + max98090->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + max98090->bclk *= 2; + + max98090->lrclk = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98090_REG_INTERFACE_FORMAT, + M98090_WS_MASK, 0); + break; + default: + return -EINVAL; + } + + max98090_configure_bclk(codec); + + cdata->rate = max98090->lrclk; + + /* Update filter mode */ + if (max98090->lrclk < 24000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, M98090_MODE_MASK); + + /* Update sample rate mode */ + if (max98090->lrclk < 50000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, M98090_DHF_MASK); + + /* Check for supported PCLK to LRCLK ratios */ + for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) { + if (comp_pclk_rates[j] == max98090->sysclk) { + break; + } + } + + for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { + if (max98090->lrclk <= (comp_lrclk_rates[i] + + comp_lrclk_rates[i + 1]) / 2) { + break; + } + } + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_ENABLE, + M98090_MICCLK_MASK, + dmic_micclk[j] << M98090_MICCLK_SHIFT); + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_MASK, + dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT); + + return 0; +} + +/* + * PLL / Sysclk + */ +static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + /* Requested clock frequency is already setup */ + if (freq == max98090->sysclk) + return 0; + + /* Setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 40MHz).. + * 0x03 (when master clk is 40MHz to 60MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV1); + } else if ((freq >= 20000000) && (freq < 40000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV2); + } else if ((freq >= 40000000) && (freq < 60000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV4); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + max98090->sysclk = freq; + + max98090_configure_bclk(codec); + + return 0; +} + +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int regval; + + regval = mute ? M98090_DVM_MASK : 0; + snd_soc_update_bits(codec, M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVM_MASK, regval); + + return 0; +} + +static void max98090_jack_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = container_of(work, + struct max98090_priv, + jack_work.work); + struct snd_soc_codec *codec = max98090->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int status = 0; + int reg; + + /* Read a second time */ + if (max98090->jack_state == M98090_JACK_STATE_NO_HEADSET) { + + /* Strong pull up allows mic detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, 0); + + msleep(50); + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + /* Weak pull up allows only insertion detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + } else { + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + } + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { + case M98090_LSNS_MASK | M98090_JKSNS_MASK: + dev_dbg(codec->dev, "No Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + status |= 0; + + break; + + case 0: + if (max98090->jack_state == + M98090_JACK_STATE_HEADSET) { + + dev_dbg(codec->dev, + "Headset Button Down Detected\n"); + + /* + * max98090_headset_button_event(codec) + * could be defined, then called here. + */ + + status |= SND_JACK_HEADSET; + status |= SND_JACK_BTN_0; + + break; + } + + /* Line is reported as Headphone */ + /* Nokia Headset is reported as Headphone */ + /* Mono Headphone is reported as Headphone */ + dev_dbg(codec->dev, "Headphone Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADPHONE; + + status |= SND_JACK_HEADPHONE; + + break; + + case M98090_JKSNS_MASK: + dev_dbg(codec->dev, "Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADSET; + + status |= SND_JACK_HEADSET; + + break; + + default: + dev_dbg(codec->dev, "Unrecognized Jack Status\n"); + break; + } + + snd_soc_jack_report(max98090->jack, status, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + snd_soc_dapm_sync(dapm); +} + +static irqreturn_t max98090_interrupt(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int mask; + unsigned int active; + + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); + + ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_INTERRUPT_S: %d\n", + ret); + return IRQ_NONE; + } + + ret = regmap_read(max98090->regmap, M98090_REG_DEVICE_STATUS, &active); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_DEVICE_STATUS: %d\n", + ret); + return IRQ_NONE; + } + + dev_dbg(codec->dev, "active=0x%02x mask=0x%02x -> active=0x%02x\n", + active, mask, active & mask); + + active &= mask; + + if (!active) + return IRQ_NONE; + + if (active & M98090_CLD_MASK) { + dev_err(codec->dev, "M98090_CLD_MASK\n"); + } + + if (active & M98090_SLD_MASK) { + dev_dbg(codec->dev, "M98090_SLD_MASK\n"); + } + + if (active & M98090_ULK_MASK) { + dev_err(codec->dev, "M98090_ULK_MASK\n"); + } + + if (active & M98090_JDET_MASK) { + dev_dbg(codec->dev, "M98090_JDET_MASK\n"); + + pm_wakeup_event(codec->dev, 100); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + } + + if (active & M98090_DRCACT_MASK) { + dev_dbg(codec->dev, "M98090_DRCACT_MASK\n"); + } + + if (active & M98090_DRCCLP_MASK) { + dev_err(codec->dev, "M98090_DRCCLP_MASK\n"); + } + + return IRQ_HANDLED; +} + +/** + * max98090_mic_detect - Enable microphone detection via the MAX98090 IRQ + * + * @codec: MAX98090 codec + * @jack: jack to report detection events on + * + * Enable microphone detection via IRQ on the MAX98090. If GPIOs are + * being used to bring out signals to the processor then only platform + * data configuration is needed for MAX98090 and processor GPIOs should + * be configured using snd_soc_jack_add_gpios() instead. + * + * If no jack is supplied detection will be disabled. + */ +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "max98090_mic_detect\n"); + + max98090->jack = jack; + if (jack) { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 1 << M98090_IJDET_SHIFT); + } else { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 0); + } + + /* Send an initial empty report */ + snd_soc_jack_report(max98090->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + + return 0; +} +EXPORT_SYMBOL_GPL(max98090_mic_detect); + +#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98090_dai_ops = { + .set_sysclk = max98090_dai_set_sysclk, + .set_fmt = max98090_dai_set_fmt, + .set_tdm_slot = max98090_set_tdm_slot, + .hw_params = max98090_dai_hw_params, + .digital_mute = max98090_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98090_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .ops = &max98090_dai_ops, +} +}; + +static void max98090_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_pdata *pdata = max98090->pdata; + + if (!pdata) { + dev_err(codec->dev, "No platform data\n"); + return; + } + +} + +static int max98090_probe(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int ret = 0; + + dev_dbg(codec->dev, "max98090_probe\n"); + + max98090->codec = codec; + + codec->control_data = max98090->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Reset the codec, the DSP core, and disable all interrupts */ + max98090_reset(max98090); + + /* Initialize private data */ + + max98090->sysclk = (unsigned)-1; + + cdata = &max98090->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + max98090->lin_state = 0; + max98090->pa1en = 0; + max98090->pa2en = 0; + max98090->extmic_mux = 0; + + ret = snd_soc_read(codec, M98090_REG_REVISION_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + + if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { + max98090->devtype = MAX98090; + dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); + } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { + max98090->devtype = MAX98091; + dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); + } else { + max98090->devtype = MAX98090; + dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); + } + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + + /* Enable jack detection */ + snd_soc_write(codec, M98090_REG_JACK_DETECT, + M98090_JDETEN_MASK | M98090_JDEB_25MS); + + /* Register for interrupts */ + dev_dbg(codec->dev, "irq = %d\n", max98090->irq); + + ret = request_threaded_irq(max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING, + "max98090_interrupt", codec); + if (ret < 0) { + dev_err(codec->dev, "request_irq failed: %d\n", + ret); + } + + /* + * Clear any old interrupts. + * An old interrupt ocurring prior to installing the ISR + * can keep a new interrupt from generating a trigger. + */ + snd_soc_read(codec, M98090_REG_DEVICE_STATUS); + + /* High Performance is default */ + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_DACHP_MASK, + 1 << M98090_DACHP_SHIFT); + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_PERFMODE_MASK, + 0 << M98090_PERFMODE_SHIFT); + snd_soc_update_bits(codec, M98090_REG_ADC_CONTROL, + M98090_ADCHP_MASK, + 1 << M98090_ADCHP_SHIFT); + + /* Turn on VCM bandgap reference */ + snd_soc_write(codec, M98090_REG_BIAS_CONTROL, + M98090_VCM_MODE_MASK); + + max98090_handle_pdata(codec); + + max98090_add_widgets(codec); + +err_access: + return ret; +} + +static int max98090_remove(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&max98090->jack_work); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98090 = { + .probe = max98090_probe, + .remove = max98090_remove, + .set_bias_level = max98090_set_bias_level, +}; + +static const struct regmap_config max98090_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX98090_MAX_REGISTER, + .reg_defaults = max98090_reg, + .num_reg_defaults = ARRAY_SIZE(max98090_reg), + .volatile_reg = max98090_volatile_register, + .readable_reg = max98090_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98090_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98090_priv *max98090; + int ret; + + pr_debug("max98090_i2c_probe\n"); + + max98090 = devm_kzalloc(&i2c->dev, sizeof(struct max98090_priv), + GFP_KERNEL); + if (max98090 == NULL) + return -ENOMEM; + + max98090->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98090); + max98090->control_data = i2c; + max98090->pdata = i2c->dev.platform_data; + max98090->irq = i2c->irq; + + max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + if (IS_ERR(max98090->regmap)) { + ret = PTR_ERR(max98090->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98090, max98090_dai, + ARRAY_SIZE(max98090_dai)); + if (ret < 0) + regmap_exit(max98090->regmap); + +err_enable: + return ret; +} + +static int max98090_i2c_remove(struct i2c_client *client) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); + snd_soc_unregister_codec(&client->dev); + regmap_exit(max98090->regmap); + return 0; +} + +static int max98090_runtime_resume(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, false); + + regcache_sync(max98090->regmap); + + return 0; +} + +static int max98090_runtime_suspend(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, true); + + return 0; +} + +static struct dev_pm_ops max98090_pm = { + SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + max98090_runtime_resume, NULL) +}; + +static const struct i2c_device_id max98090_i2c_id[] = { + { "max98090", MAX98090 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); + +static struct i2c_driver max98090_i2c_driver = { + .driver = { + .name = "max98090", + .owner = THIS_MODULE, + .pm = &max98090_pm, + }, + .probe = max98090_i2c_probe, + .remove = max98090_i2c_remove, + .id_table = max98090_i2c_id, +}; + +module_i2c_driver(max98090_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); +MODULE_AUTHOR("Peter Hsiang, Jesse Marroqin, Jerry Wong"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h new file mode 100755 index 0000000..6c9fa34 --- /dev/null +++ b/sound/soc/codecs/max98090.h @@ -0,0 +1,1549 @@ +/* + * max98090.h -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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 _MAX98090_H +#define _MAX98090_H + +#include <linux/version.h> + +/* One can override the Linux version here with an explicit version number */ +#define M98090_LINUX_VERSION LINUX_VERSION_CODE + +/* + * MAX98090 Register Definitions + */ + +#define M98090_REG_SOFTWARE_RESET 0x00 +#define M98090_REG_DEVICE_STATUS 0x01 +#define M98090_REG_JACK_STATUS 0x02 +#define M98090_REG_INTERRUPT_S 0x03 +#define M98090_REG_QUICK_SYSTEM_CLOCK 0x04 +#define M98090_REG_QUICK_SAMPLE_RATE 0x05 +#define M98090_REG_DAI_INTERFACE 0x06 +#define M98090_REG_DAC_PATH 0x07 +#define M98090_REG_MIC_DIRECT_TO_ADC 0x08 +#define M98090_REG_LINE_TO_ADC 0x09 +#define M98090_REG_ANALOG_MIC_LOOP 0x0A +#define M98090_REG_ANALOG_LINE_LOOP 0x0B +#define M98090_REG_RESERVED 0x0C +#define M98090_REG_LINE_INPUT_CONFIG 0x0D +#define M98090_REG_LINE_INPUT_LEVEL 0x0E +#define M98090_REG_INPUT_MODE 0x0F +#define M98090_REG_MIC1_INPUT_LEVEL 0x10 +#define M98090_REG_MIC2_INPUT_LEVEL 0x11 +#define M98090_REG_MIC_BIAS_VOLTAGE 0x12 +#define M98090_REG_DIGITAL_MIC_ENABLE 0x13 +#define M98090_REG_DIGITAL_MIC_CONFIG 0x14 +#define M98090_REG_LEFT_ADC_MIXER 0x15 +#define M98090_REG_RIGHT_ADC_MIXER 0x16 +#define M98090_REG_LEFT_ADC_LEVEL 0x17 +#define M98090_REG_RIGHT_ADC_LEVEL 0x18 +#define M98090_REG_ADC_BIQUAD_LEVEL 0x19 +#define M98090_REG_ADC_SIDETONE 0x1A +#define M98090_REG_SYSTEM_CLOCK 0x1B +#define M98090_REG_CLOCK_MODE 0x1C +#define M98090_REG_CLOCK_RATIO_NI_MSB 0x1D +#define M98090_REG_CLOCK_RATIO_NI_LSB 0x1E +#define M98090_REG_CLOCK_RATIO_MI_MSB 0x1F +#define M98090_REG_CLOCK_RATIO_MI_LSB 0x20 +#define M98090_REG_MASTER_MODE 0x21 +#define M98090_REG_INTERFACE_FORMAT 0x22 +#define M98090_REG_TDM_CONTROL 0x23 +#define M98090_REG_TDM_FORMAT 0x24 +#define M98090_REG_IO_CONFIGURATION 0x25 +#define M98090_REG_FILTER_CONFIG 0x26 +#define M98090_REG_DAI_PLAYBACK_LEVEL 0x27 +#define M98090_REG_DAI_PLAYBACK_LEVEL_EQ 0x28 +#define M98090_REG_LEFT_HP_MIXER 0x29 +#define M98090_REG_RIGHT_HP_MIXER 0x2A +#define M98090_REG_HP_CONTROL 0x2B +#define M98090_REG_LEFT_HP_VOLUME 0x2C +#define M98090_REG_RIGHT_HP_VOLUME 0x2D +#define M98090_REG_LEFT_SPK_MIXER 0x2E +#define M98090_REG_RIGHT_SPK_MIXER 0x2F +#define M98090_REG_SPK_CONTROL 0x30 +#define M98090_REG_LEFT_SPK_VOLUME 0x31 +#define M98090_REG_RIGHT_SPK_VOLUME 0x32 +#define M98090_REG_DRC_TIMING 0x33 +#define M98090_REG_DRC_COMPRESSOR 0x34 +#define M98090_REG_DRC_EXPANDER 0x35 +#define M98090_REG_DRC_GAIN 0x36 +#define M98090_REG_RCV_LOUTL_MIXER 0x37 +#define M98090_REG_RCV_LOUTL_CONTROL 0x38 +#define M98090_REG_RCV_LOUTL_VOLUME 0x39 +#define M98090_REG_LOUTR_MIXER 0x3A +#define M98090_REG_LOUTR_CONTROL 0x3B +#define M98090_REG_LOUTR_VOLUME 0x3C +#define M98090_REG_JACK_DETECT 0x3D +#define M98090_REG_INPUT_ENABLE 0x3E +#define M98090_REG_OUTPUT_ENABLE 0x3F +#define M98090_REG_LEVEL_CONTROL 0x40 +#define M98090_REG_DSP_FILTER_ENABLE 0x41 +#define M98090_REG_BIAS_CONTROL 0x42 +#define M98090_REG_DAC_CONTROL 0x43 +#define M98090_REG_ADC_CONTROL 0x44 +#define M98090_REG_DEVICE_SHUTDOWN 0x45 +#define M98090_REG_EQUALIZER_BASE 0x46 +#define M98090_REG_RECORD_BIQUAD_BASE 0xAF +#define M98090_REG_DMIC3_VOLUME 0xBE +#define M98090_REG_DMIC4_VOLUME 0xBF +#define M98090_REG_DMIC34_BQ_PREATTEN 0xC0 +#define M98090_REG_RECORD_TDM_SLOT 0xC1 +#define M98090_REG_SAMPLE_RATE 0xC2 +#define M98090_REG_DMIC34_BIQUAD_BASE 0xC3 +#define M98090_REG_REVISION_ID 0xFF + +#define M98090_REG_CNT (0xFF+1) +#define MAX98090_MAX_REGISTER 0xFF + +/* MAX98090 Register Bit Fields */ + +/* + * M98090_REG_SOFTWARE_RESET + */ +#define M98090_SWRESET_MASK (1<<7) +#define M98090_SWRESET_SHIFT 7 +#define M98090_SWRESET_WIDTH 1 + +/* + * M98090_REG_DEVICE_STATUS + */ +#define M98090_CLD_MASK (1<<7) +#define M98090_CLD_SHIFT 7 +#define M98090_CLD_WIDTH 1 +#define M98090_SLD_MASK (1<<6) +#define M98090_SLD_SHIFT 6 +#define M98090_SLD_WIDTH 1 +#define M98090_ULK_MASK (1<<5) +#define M98090_ULK_SHIFT 5 +#define M98090_ULK_WIDTH 1 +#define M98090_JDET_MASK (1<<2) +#define M98090_JDET_SHIFT 2 +#define M98090_JDET_WIDTH 1 +#define M98090_DRCACT_MASK (1<<1) +#define M98090_DRCACT_SHIFT 1 +#define M98090_DRCACT_WIDTH 1 +#define M98090_DRCCLP_MASK (1<<0) +#define M98090_DRCCLP_SHIFT 0 +#define M98090_DRCCLP_WIDTH 1 + +/* + * M98090_REG_JACK_STATUS + */ +#define M98090_LSNS_MASK (1<<2) +#define M98090_LSNS_SHIFT 2 +#define M98090_LSNS_WIDTH 1 +#define M98090_JKSNS_MASK (1<<1) +#define M98090_JKSNS_SHIFT 1 +#define M98090_JKSNS_WIDTH 1 + +/* + * M98090_REG_INTERRUPT_S + */ +#define M98090_ICLD_MASK (1<<7) +#define M98090_ICLD_SHIFT 7 +#define M98090_ICLD_WIDTH 1 +#define M98090_ISLD_MASK (1<<6) +#define M98090_ISLD_SHIFT 6 +#define M98090_ISLD_WIDTH 1 +#define M98090_IULK_MASK (1<<5) +#define M98090_IULK_SHIFT 5 +#define M98090_IULK_WIDTH 1 +#define M98090_IJDET_MASK (1<<2) +#define M98090_IJDET_SHIFT 2 +#define M98090_IJDET_WIDTH 1 +#define M98090_IDRCACT_MASK (1<<1) +#define M98090_IDRCACT_SHIFT 1 +#define M98090_IDRCACT_WIDTH 1 +#define M98090_IDRCCLP_MASK (1<<0) +#define M98090_IDRCCLP_SHIFT 0 +#define M98090_IDRCCLP_WIDTH 1 + +/* + * M98090_REG_QUICK_SYSTEM_CLOCK + */ +#define M98090_26M_MASK (1<<7) +#define M98090_26M_SHIFT 7 +#define M98090_26M_WIDTH 1 +#define M98090_19P2M_MASK (1<<6) +#define M98090_19P2M_SHIFT 6 +#define M98090_19P2M_WIDTH 1 +#define M98090_13M_MASK (1<<5) +#define M98090_13M_SHIFT 5 +#define M98090_13M_WIDTH 1 +#define M98090_12P288M_MASK (1<<4) +#define M98090_12P288M_SHIFT 4 +#define M98090_12P288M_WIDTH 1 +#define M98090_12M_MASK (1<<3) +#define M98090_12M_SHIFT 3 +#define M98090_12M_WIDTH 1 +#define M98090_11P2896M_MASK (1<<2) +#define M98090_11P2896M_SHIFT 2 +#define M98090_11P2896M_WIDTH 1 +#define M98090_256FS_MASK (1<<0) +#define M98090_256FS_SHIFT 0 +#define M98090_256FS_WIDTH 1 +#define M98090_CLK_ALL_SHIFT 0 +#define M98090_CLK_ALL_WIDTH 8 +#define M98090_CLK_ALL_NUM (1<<M98090_CLK_ALL_WIDTH) + +/* + * M98090_REG_QUICK_SAMPLE_RATE + */ +#define M98090_SR_96K_MASK (1<<5) +#define M98090_SR_96K_SHIFT 5 +#define M98090_SR_96K_WIDTH 1 +#define M98090_SR_32K_MASK (1<<4) +#define M98090_SR_32K_SHIFT 4 +#define M98090_SR_32K_WIDTH 1 +#define M98090_SR_48K_MASK (1<<3) +#define M98090_SR_48K_SHIFT 3 +#define M98090_SR_48K_WIDTH 1 +#define M98090_SR_44K1_MASK (1<<2) +#define M98090_SR_44K1_SHIFT 2 +#define M98090_SR_44K1_WIDTH 1 +#define M98090_SR_16K_MASK (1<<1) +#define M98090_SR_16K_SHIFT 1 +#define M98090_SR_16K_WIDTH 1 +#define M98090_SR_8K_MASK (1<<0) +#define M98090_SR_8K_SHIFT 0 +#define M98090_SR_8K_WIDTH 1 +#define M98090_SR_MASK 0x3F +#define M98090_SR_ALL_SHIFT 0 +#define M98090_SR_ALL_WIDTH 8 +#define M98090_SR_ALL_NUM (1<<M98090_SR_ALL_WIDTH) + +/* + * M98090_REG_DAI_INTERFACE + */ +#define M98090_RJ_M_MASK (1<<5) +#define M98090_RJ_M_SHIFT 5 +#define M98090_RJ_M_WIDTH 1 +#define M98090_RJ_S_MASK (1<<4) +#define M98090_RJ_S_SHIFT 4 +#define M98090_RJ_S_WIDTH 1 +#define M98090_LJ_M_MASK (1<<3) +#define M98090_LJ_M_SHIFT 3 +#define M98090_LJ_M_WIDTH 1 +#define M98090_LJ_S_MASK (1<<2) +#define M98090_LJ_S_SHIFT 2 +#define M98090_LJ_S_WIDTH 1 +#define M98090_I2S_M_MASK (1<<1) +#define M98090_I2S_M_SHIFT 1 +#define M98090_I2S_M_WIDTH 1 +#define M98090_I2S_S_MASK (1<<0) +#define M98090_I2S_S_SHIFT 0 +#define M98090_I2S_S_WIDTH 1 +#define M98090_DAI_ALL_SHIFT 0 +#define M98090_DAI_ALL_WIDTH 8 +#define M98090_DAI_ALL_NUM (1<<M98090_DAI_ALL_WIDTH) + +/* + * M98090_REG_DAC_PATH + */ +#define M98090_DIG2_HP_MASK (1<<7) +#define M98090_DIG2_HP_SHIFT 7 +#define M98090_DIG2_HP_WIDTH 1 +#define M98090_DIG2_EAR_MASK (1<<6) +#define M98090_DIG2_EAR_SHIFT 6 +#define M98090_DIG2_EAR_WIDTH 1 +#define M98090_DIG2_SPK_MASK (1<<5) +#define M98090_DIG2_SPK_SHIFT 5 +#define M98090_DIG2_SPK_WIDTH 1 +#define M98090_DIG2_LOUT_MASK (1<<4) +#define M98090_DIG2_LOUT_SHIFT 4 +#define M98090_DIG2_LOUT_WIDTH 1 +#define M98090_DIG2_ALL_SHIFT 0 +#define M98090_DIG2_ALL_WIDTH 8 +#define M98090_DIG2_ALL_NUM (1<<M98090_DIG2_ALL_WIDTH) + +/* + * M98090_REG_MIC_DIRECT_TO_ADC + */ +#define M98090_IN12_MIC1_MASK (1<<7) +#define M98090_IN12_MIC1_SHIFT 7 +#define M98090_IN12_MIC1_WIDTH 1 +#define M98090_IN34_MIC2_MASK (1<<6) +#define M98090_IN34_MIC2_SHIFT 6 +#define M98090_IN34_MIC2_WIDTH 1 +#define M98090_IN56_MIC1_MASK (1<<5) +#define M98090_IN56_MIC1_SHIFT 5 +#define M98090_IN56_MIC1_WIDTH 1 +#define M98090_IN56_MIC2_MASK (1<<4) +#define M98090_IN56_MIC2_SHIFT 4 +#define M98090_IN56_MIC2_WIDTH 1 +#define M98090_IN12_DADC_MASK (1<<3) +#define M98090_IN12_DADC_SHIFT 3 +#define M98090_IN12_DADC_WIDTH 1 +#define M98090_IN34_DADC_MASK (1<<2) +#define M98090_IN34_DADC_SHIFT 2 +#define M98090_IN34_DADC_WIDTH 1 +#define M98090_IN56_DADC_MASK (1<<1) +#define M98090_IN56_DADC_SHIFT 1 +#define M98090_IN56_DADC_WIDTH 1 +#define M98090_MIC_ALL_SHIFT 0 +#define M98090_MIC_ALL_WIDTH 8 +#define M98090_MIC_ALL_NUM (1<<M98090_MIC_ALL_WIDTH) + +/* + * M98090_REG_LINE_TO_ADC + */ +#define M98090_IN12S_AB_MASK (1<<7) +#define M98090_IN12S_AB_SHIFT 7 +#define M98090_IN12S_AB_WIDTH 1 +#define M98090_IN34S_AB_MASK (1<<6) +#define M98090_IN34S_AB_SHIFT 6 +#define M98090_IN34S_AB_WIDTH 1 +#define M98090_IN56S_AB_MASK (1<<5) +#define M98090_IN56S_AB_SHIFT 5 +#define M98090_IN56S_AB_WIDTH 1 +#define M98090_IN34D_A_MASK (1<<4) +#define M98090_IN34D_A_SHIFT 4 +#define M98090_IN34D_A_WIDTH 1 +#define M98090_IN56D_B_MASK (1<<3) +#define M98090_IN56D_B_SHIFT 3 +#define M98090_IN56D_B_WIDTH 1 +#define M98090_LINE_ALL_SHIFT 0 +#define M98090_LINE_ALL_WIDTH 8 +#define M98090_LINE_ALL_NUM (1<<M98090_LINE_ALL_WIDTH) + +/* + * M98090_REG_ANALOG_MIC_LOOP + */ +#define M98090_IN12_M1HPL_MASK (1<<7) +#define M98090_IN12_M1HPL_SHIFT 7 +#define M98090_IN12_M1HPL_WIDTH 1 +#define M98090_IN12_M1SPKL_MASK (1<<6) +#define M98090_IN12_M1SPKL_SHIFT 6 +#define M98090_IN12_M1SPKL_WIDTH 1 +#define M98090_IN12_M1EAR_MASK (1<<5) +#define M98090_IN12_M1EAR_SHIFT 5 +#define M98090_IN12_M1EAR_WIDTH 1 +#define M98090_IN12_M1LOUTL_MASK (1<<4) +#define M98090_IN12_M1LOUTL_SHIFT 4 +#define M98090_IN12_M1LOUTL_WIDTH 1 +#define M98090_IN34_M2HPR_MASK (1<<3) +#define M98090_IN34_M2HPR_SHIFT 3 +#define M98090_IN34_M2HPR_WIDTH 1 +#define M98090_IN34_M2SPKR_MASK (1<<2) +#define M98090_IN34_M2SPKR_SHIFT 2 +#define M98090_IN34_M2SPKR_WIDTH 1 +#define M98090_IN34_M2EAR_MASK (1<<1) +#define M98090_IN34_M2EAR_SHIFT 1 +#define M98090_IN34_M2EAR_WIDTH 1 +#define M98090_IN34_M2LOUTR_MASK (1<<0) +#define M98090_IN34_M2LOUTR_SHIFT 0 +#define M98090_IN34_M2LOUTR_WIDTH 1 +#define M98090_AMIC_ALL_SHIFT 0 +#define M98090_AMIC_ALL_WIDTH 8 +#define M98090_AMIC_ALL_NUM (1<<M98090_AMIC_ALL_WIDTH) + +/* + * M98090_REG_ANALOG_LINE_LOOP + */ +#define M98090_IN12S_ABHP_MASK (1<<7) +#define M98090_IN12S_ABHP_SHIFT 7 +#define M98090_IN12S_ABHP_WIDTH 1 +#define M98090_IN34D_ASPKL_MASK (1<<6) +#define M98090_IN34D_ASPKL_SHIFT 6 +#define M98090_IN34D_ASPKL_WIDTH 1 +#define M98090_IN34D_AEAR_MASK (1<<5) +#define M98090_IN34D_AEAR_SHIFT 5 +#define M98090_IN34D_AEAR_WIDTH 1 +#define M98090_IN12S_ABLOUT_MASK (1<<4) +#define M98090_IN12S_ABLOUT_SHIFT 4 +#define M98090_IN12S_ABLOUT_WIDTH 1 +#define M98090_IN34S_ABHP_MASK (1<<3) +#define M98090_IN34S_ABHP_SHIFT 3 +#define M98090_IN34S_ABHP_WIDTH 1 +#define M98090_IN56D_BSPKR_MASK (1<<2) +#define M98090_IN56D_BSPKR_SHIFT 2 +#define M98090_IN56D_BSPKR_WIDTH 1 +#define M98090_IN56D_BEAR_MASK (1<<1) +#define M98090_IN56D_BEAR_SHIFT 1 +#define M98090_IN56D_BEAR_WIDTH 1 +#define M98090_IN34S_ABLOUT_MASK (1<<0) +#define M98090_IN34S_ABLOUT_SHIFT 0 +#define M98090_IN34S_ABLOUT_WIDTH 1 +#define M98090_ALIN_ALL_SHIFT 0 +#define M98090_ALIN_ALL_WIDTH 8 +#define M98090_ALIN_ALL_NUM (1<<M98090_ALIN_ALL_WIDTH) + +/* + * M98090_REG_RESERVED + */ + +/* + * M98090_REG_LINE_INPUT_CONFIG + */ +#define M98090_IN34DIFF_MASK (1<<7) +#define M98090_IN34DIFF_SHIFT 7 +#define M98090_IN34DIFF_WIDTH 1 +#define M98090_IN56DIFF_MASK (1<<6) +#define M98090_IN56DIFF_SHIFT 6 +#define M98090_IN56DIFF_WIDTH 1 +#define M98090_IN1SEEN_MASK (1<<5) +#define M98090_IN1SEEN_SHIFT 5 +#define M98090_IN1SEEN_WIDTH 1 +#define M98090_IN2SEEN_MASK (1<<4) +#define M98090_IN2SEEN_SHIFT 4 +#define M98090_IN2SEEN_WIDTH 1 +#define M98090_IN3SEEN_MASK (1<<3) +#define M98090_IN3SEEN_SHIFT 3 +#define M98090_IN3SEEN_WIDTH 1 +#define M98090_IN4SEEN_MASK (1<<2) +#define M98090_IN4SEEN_SHIFT 2 +#define M98090_IN4SEEN_WIDTH 1 +#define M98090_IN5SEEN_MASK (1<<1) +#define M98090_IN5SEEN_SHIFT 1 +#define M98090_IN5SEEN_WIDTH 1 +#define M98090_IN6SEEN_MASK (1<<0) +#define M98090_IN6SEEN_SHIFT 0 +#define M98090_IN6SEEN_WIDTH 1 + +/* + * M98090_REG_LINE_INPUT_LEVEL + */ +#define M98090_MIXG135_MASK (1<<7) +#define M98090_MIXG135_SHIFT 7 +#define M98090_MIXG135_WIDTH 1 +#define M98090_MIXG135_NUM (1<<M98090_MIXG135_WIDTH) +#define M98090_MIXG246_MASK (1<<6) +#define M98090_MIXG246_SHIFT 6 +#define M98090_MIXG246_WIDTH 1 +#define M98090_MIXG246_NUM (1<<M98090_MIXG246_WIDTH) +#define M98090_LINAPGA_MASK (7<<3) +#define M98090_LINAPGA_SHIFT 3 +#define M98090_LINAPGA_WIDTH 3 +#define M98090_LINAPGA_NUM 6 +#define M98090_LINBPGA_MASK (7<<0) +#define M98090_LINBPGA_SHIFT 0 +#define M98090_LINBPGA_WIDTH 3 +#define M98090_LINBPGA_NUM 6 + +/* + * M98090_REG_INPUT_MODE + */ +#define M98090_EXTBUFA_MASK (1<<7) +#define M98090_EXTBUFA_SHIFT 7 +#define M98090_EXTBUFA_WIDTH 1 +#define M98090_EXTBUFA_NUM (1<<M98090_EXTBUFA_WIDTH) +#define M98090_EXTBUFB_MASK (1<<6) +#define M98090_EXTBUFB_SHIFT 6 +#define M98090_EXTBUFB_WIDTH 1 +#define M98090_EXTBUFB_NUM (1<<M98090_EXTBUFB_WIDTH) +#define M98090_EXTMIC_MASK (3<<0) +#define M98090_EXTMIC_SHIFT 0 +#define M98090_EXTMIC1_SHIFT 0 +#define M98090_EXTMIC2_SHIFT 1 +#define M98090_EXTMIC_WIDTH 2 +#define M98090_EXTMIC_NONE (0<<0) +#define M98090_EXTMIC_MIC1 (1<<0) +#define M98090_EXTMIC_MIC2 (2<<0) + +/* + * M98090_REG_MIC1_INPUT_LEVEL + */ +#define M98090_MIC_PA1EN_MASK (3<<5) +#define M98090_MIC_PA1EN_SHIFT 5 +#define M98090_MIC_PA1EN_WIDTH 2 +#define M98090_MIC_PA1EN_NUM 3 +#define M98090_MIC_PGAM1_MASK (31<<0) +#define M98090_MIC_PGAM1_SHIFT 0 +#define M98090_MIC_PGAM1_WIDTH 5 +#define M98090_MIC_PGAM1_NUM 21 + +/* + * M98090_REG_MIC2_INPUT_LEVEL + */ +#define M98090_MIC_PA2EN_MASK (3<<5) +#define M98090_MIC_PA2EN_SHIFT 5 +#define M98090_MIC_PA2EN_WIDTH 2 +#define M98090_MIC_PA2EN_NUM 3 +#define M98090_MIC_PGAM2_MASK (31<<0) +#define M98090_MIC_PGAM2_SHIFT 0 +#define M98090_MIC_PGAM2_WIDTH 5 +#define M98090_MIC_PGAM2_NUM 21 + +/* + * M98090_REG_MIC_BIAS_VOLTAGE + */ +#define M98090_MBVSEL_MASK (3<<0) +#define M98090_MBVSEL_SHIFT 0 +#define M98090_MBVSEL_WIDTH 2 +#define M98090_MBVSEL_2V8 (3<<0) +#define M98090_MBVSEL_2V55 (2<<0) +#define M98090_MBVSEL_2V4 (1<<0) +#define M98090_MBVSEL_2V2 (0<<0) + +/* + * M98090_REG_DIGITAL_MIC_ENABLE + */ +#define M98090_MICCLK_MASK (7<<4) +#define M98090_MICCLK_SHIFT 4 +#define M98090_MICCLK_WIDTH 3 +#define M98090_DIGMIC4_MASK (1<<3) +#define M98090_DIGMIC4_SHIFT 3 +#define M98090_DIGMIC4_WIDTH 1 +#define M98090_DIGMIC4_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DIGMIC3_MASK (1<<2) +#define M98090_DIGMIC3_SHIFT 2 +#define M98090_DIGMIC3_WIDTH 1 +#define M98090_DIGMIC3_NUM (1<<M98090_DIGMIC3_WIDTH) +#define M98090_DIGMICR_MASK (1<<1) +#define M98090_DIGMICR_SHIFT 1 +#define M98090_DIGMICR_WIDTH 1 +#define M98090_DIGMICR_NUM (1<<M98090_DIGMICR_WIDTH) +#define M98090_DIGMICL_MASK (1<<0) +#define M98090_DIGMICL_SHIFT 0 +#define M98090_DIGMICL_WIDTH 1 +#define M98090_DIGMICL_NUM (1<<M98090_DIGMICL_WIDTH) + +/* + * M98090_REG_DIGITAL_MIC_CONFIG + */ +#define M98090_DMIC_COMP_MASK (15<<4) +#define M98090_DMIC_COMP_SHIFT 4 +#define M98090_DMIC_COMP_WIDTH 4 +#define M98090_DMIC_COMP_NUM (1<<M98090_DMIC_COMP_WIDTH) +#define M98090_DMIC_FREQ_MASK (3<<0) +#define M98090_DMIC_FREQ_SHIFT 0 +#define M98090_DMIC_FREQ_WIDTH 2 + +/* + * M98090_REG_LEFT_ADC_MIXER + */ +#define M98090_MIXADL_MIC2_MASK (1<<6) +#define M98090_MIXADL_MIC2_SHIFT 6 +#define M98090_MIXADL_MIC2_WIDTH 1 +#define M98090_MIXADL_MIC1_MASK (1<<5) +#define M98090_MIXADL_MIC1_SHIFT 5 +#define M98090_MIXADL_MIC1_WIDTH 1 +#define M98090_MIXADL_LINEB_MASK (1<<4) +#define M98090_MIXADL_LINEB_SHIFT 4 +#define M98090_MIXADL_LINEB_WIDTH 1 +#define M98090_MIXADL_LINEA_MASK (1<<3) +#define M98090_MIXADL_LINEA_SHIFT 3 +#define M98090_MIXADL_LINEA_WIDTH 1 +#define M98090_MIXADL_IN65DIFF_MASK (1<<2) +#define M98090_MIXADL_IN65DIFF_SHIFT 2 +#define M98090_MIXADL_IN65DIFF_WIDTH 1 +#define M98090_MIXADL_IN34DIFF_MASK (1<<1) +#define M98090_MIXADL_IN34DIFF_SHIFT 1 +#define M98090_MIXADL_IN34DIFF_WIDTH 1 +#define M98090_MIXADL_IN12DIFF_MASK (1<<0) +#define M98090_MIXADL_IN12DIFF_SHIFT 0 +#define M98090_MIXADL_IN12DIFF_WIDTH 1 +#define M98090_MIXADL_MASK (255<<0) +#define M98090_MIXADL_SHIFT 0 +#define M98090_MIXADL_WIDTH 8 + +/* + * M98090_REG_RIGHT_ADC_MIXER + */ +#define M98090_MIXADR_MIC2_MASK (1<<6) +#define M98090_MIXADR_MIC2_SHIFT 6 +#define M98090_MIXADR_MIC2_WIDTH 1 +#define M98090_MIXADR_MIC1_MASK (1<<5) +#define M98090_MIXADR_MIC1_SHIFT 5 +#define M98090_MIXADR_MIC1_WIDTH 1 +#define M98090_MIXADR_LINEB_MASK (1<<4) +#define M98090_MIXADR_LINEB_SHIFT 4 +#define M98090_MIXADR_LINEB_WIDTH 1 +#define M98090_MIXADR_LINEA_MASK (1<<3) +#define M98090_MIXADR_LINEA_SHIFT 3 +#define M98090_MIXADR_LINEA_WIDTH 1 +#define M98090_MIXADR_IN65DIFF_MASK (1<<2) +#define M98090_MIXADR_IN65DIFF_SHIFT 2 +#define M98090_MIXADR_IN65DIFF_WIDTH 1 +#define M98090_MIXADR_IN34DIFF_MASK (1<<1) +#define M98090_MIXADR_IN34DIFF_SHIFT 1 +#define M98090_MIXADR_IN34DIFF_WIDTH 1 +#define M98090_MIXADR_IN12DIFF_MASK (1<<0) +#define M98090_MIXADR_IN12DIFF_SHIFT 0 +#define M98090_MIXADR_IN12DIFF_WIDTH 1 +#define M98090_MIXADR_MASK (255<<0) +#define M98090_MIXADR_SHIFT 0 +#define M98090_MIXADR_WIDTH 8 + +/* + * M98090_REG_LEFT_ADC_LEVEL + */ +#define M98090_AVLG_MASK (7<<4) +#define M98090_AVLG_SHIFT 4 +#define M98090_AVLG_WIDTH 3 +#define M98090_AVLG_NUM (1<<M98090_AVLG_WIDTH) +#define M98090_AVL_MASK (15<<0) +#define M98090_AVL_SHIFT 0 +#define M98090_AVL_WIDTH 4 +#define M98090_AVL_NUM (1<<M98090_AVL_WIDTH) + +/* + * M98090_REG_RIGHT_ADC_LEVEL + */ +#define M98090_AVRG_MASK (7<<4) +#define M98090_AVRG_SHIFT 4 +#define M98090_AVRG_WIDTH 3 +#define M98090_AVRG_NUM (1<<M98090_AVRG_WIDTH) +#define M98090_AVR_MASK (15<<0) +#define M98090_AVR_SHIFT 0 +#define M98090_AVR_WIDTH 4 +#define M98090_AVR_NUM (1<<M98090_AVR_WIDTH) + +/* + * M98090_REG_ADC_BIQUAD_LEVEL + */ +#define M98090_AVBQ_MASK (15<<0) +#define M98090_AVBQ_SHIFT 0 +#define M98090_AVBQ_WIDTH 4 +#define M98090_AVBQ_NUM (1<<M98090_AVBQ_WIDTH) + +/* + * M98090_REG_ADC_SIDETONE + */ +#define M98090_DSTSR_MASK (1<<7) +#define M98090_DSTSR_SHIFT 7 +#define M98090_DSTSR_WIDTH 1 +#define M98090_DSTSL_MASK (1<<6) +#define M98090_DSTSL_SHIFT 6 +#define M98090_DSTSL_WIDTH 1 +#define M98090_DVST_MASK (31<<0) +#define M98090_DVST_SHIFT 0 +#define M98090_DVST_WIDTH 5 +#define M98090_DVST_NUM 31 + +/* + * M98090_REG_SYSTEM_CLOCK + */ +#define M98090_PSCLK_MASK (3<<4) +#define M98090_PSCLK_SHIFT 4 +#define M98090_PSCLK_WIDTH 2 +#define M98090_PSCLK_DISABLED (0<<4) +#define M98090_PSCLK_DIV1 (1<<4) +#define M98090_PSCLK_DIV2 (2<<4) +#define M98090_PSCLK_DIV4 (3<<4) + +/* + * M98090_REG_CLOCK_MODE + */ +#define M98090_FREQ_MASK (15<<4) +#define M98090_FREQ_SHIFT 4 +#define M98090_FREQ_WIDTH 4 +#define M98090_USE_M1_MASK (1<<0) +#define M98090_USE_M1_SHIFT 0 +#define M98090_USE_M1_WIDTH 1 +#define M98090_USE_M1_NUM (1<<M98090_USE_M1_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_NI_MSB + */ +#define M98090_NI_HI_MASK (127<<0) +#define M98090_NI_HI_SHIFT 0 +#define M98090_NI_HI_WIDTH 7 +#define M98090_NI_HI_NUM (1<<M98090_NI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_NI_LSB + */ +#define M98090_NI_LO_MASK (255<<0) +#define M98090_NI_LO_SHIFT 0 +#define M98090_NI_LO_WIDTH 8 +#define M98090_NI_LO_NUM (1<<M98090_NI_LO_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_MI_MSB + */ +#define M98090_MI_HI_MASK (255<<0) +#define M98090_MI_HI_SHIFT 0 +#define M98090_MI_HI_WIDTH 8 +#define M98090_MI_HI_NUM (1<<M98090_MI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_MI_LSB + */ +#define M98090_MI_LO_MASK (255<<0) +#define M98090_MI_LO_SHIFT 0 +#define M98090_MI_LO_WIDTH 8 +#define M98090_MI_LO_NUM (1<<M98090_MI_LO_WIDTH) + +/* + * M98090_REG_MASTER_MODE + */ +#define M98090_MAS_MASK (1<<7) +#define M98090_MAS_SHIFT 7 +#define M98090_MAS_WIDTH 1 +#define M98090_BSEL_MASK (1<<0) +#define M98090_BSEL_SHIFT 0 +#define M98090_BSEL_WIDTH 1 +#define M98090_BSEL_32 (1<<0) +#define M98090_BSEL_48 (2<<0) +#define M98090_BSEL_64 (3<<0) + +/* + * M98090_REG_INTERFACE_FORMAT + */ +#define M98090_RJ_MASK (1<<5) +#define M98090_RJ_SHIFT 5 +#define M98090_RJ_WIDTH 1 +#define M98090_WCI_MASK (1<<4) +#define M98090_WCI_SHIFT 4 +#define M98090_WCI_WIDTH 1 +#define M98090_BCI_MASK (1<<3) +#define M98090_BCI_SHIFT 3 +#define M98090_BCI_WIDTH 1 +#define M98090_DLY_MASK (1<<2) +#define M98090_DLY_SHIFT 2 +#define M98090_DLY_WIDTH 1 +#define M98090_WS_MASK (3<<0) +#define M98090_WS_SHIFT 0 +#define M98090_WS_WIDTH 2 +#define M98090_WS_NUM (1<<M98090_WS_WIDTH) + +/* + * M98090_REG_TDM_CONTROL + */ +#define M98090_FSW_MASK (1<<1) +#define M98090_FSW_SHIFT 1 +#define M98090_FSW_WIDTH 1 +#define M98090_TDM_MASK (1<<0) +#define M98090_TDM_SHIFT 0 +#define M98090_TDM_WIDTH 1 +#define M98090_TDM_NUM (1<<M98090_TDM_WIDTH) + +/* + * M98090_REG_TDM_FORMAT + */ +#define M98090_TDM_SLOTL_MASK (3<<6) +#define M98090_TDM_SLOTL_SHIFT 6 +#define M98090_TDM_SLOTL_WIDTH 2 +#define M98090_TDM_SLOTL_NUM (1<<M98090_TDM_SLOTL_WIDTH) +#define M98090_TDM_SLOTR_MASK (3<<4) +#define M98090_TDM_SLOTR_SHIFT 4 +#define M98090_TDM_SLOTR_WIDTH 2 +#define M98090_TDM_SLOTR_NUM (1<<M98090_TDM_SLOTR_WIDTH) +#define M98090_TDM_SLOTDLY_MASK (15<<0) +#define M98090_TDM_SLOTDLY_SHIFT 0 +#define M98090_TDM_SLOTDLY_WIDTH 4 +#define M98090_TDM_SLOTDLY_NUM (1<<M98090_TDM_SLOTDLY_WIDTH) + +/* + * M98090_REG_IO_CONFIGURATION + */ +#define M98090_LTEN_MASK (1<<5) +#define M98090_LTEN_SHIFT 5 +#define M98090_LTEN_WIDTH 1 +#define M98090_LTEN_NUM (1<<M98090_LTEN_WIDTH) +#define M98090_LBEN_MASK (1<<4) +#define M98090_LBEN_SHIFT 4 +#define M98090_LBEN_WIDTH 1 +#define M98090_LBEN_NUM (1<<M98090_LBEN_WIDTH) +#define M98090_DMONO_MASK (1<<3) +#define M98090_DMONO_SHIFT 3 +#define M98090_DMONO_WIDTH 1 +#define M98090_DMONO_NUM (1<<M98090_DMONO_WIDTH) +#define M98090_HIZOFF_MASK (1<<2) +#define M98090_HIZOFF_SHIFT 2 +#define M98090_HIZOFF_WIDTH 1 +#define M98090_HIZOFF_NUM (1<<M98090_HIZOFF_WIDTH) +#define M98090_SDOEN_MASK (1<<1) +#define M98090_SDOEN_SHIFT 1 +#define M98090_SDOEN_WIDTH 1 +#define M98090_SDOEN_NUM (1<<M98090_SDOEN_WIDTH) +#define M98090_SDIEN_MASK (1<<0) +#define M98090_SDIEN_SHIFT 0 +#define M98090_SDIEN_WIDTH 1 +#define M98090_SDIEN_NUM (1<<M98090_SDIEN_WIDTH) + +/* + * M98090_REG_FILTER_CONFIG + */ +#define M98090_MODE_MASK (1<<7) +#define M98090_MODE_SHIFT 7 +#define M98090_MODE_WIDTH 1 +#define M98090_AHPF_MASK (1<<6) +#define M98090_AHPF_SHIFT 6 +#define M98090_AHPF_WIDTH 1 +#define M98090_AHPF_NUM (1<<M98090_AHPF_WIDTH) +#define M98090_DHPF_MASK (1<<5) +#define M98090_DHPF_SHIFT 5 +#define M98090_DHPF_WIDTH 1 +#define M98090_DHPF_NUM (1<<M98090_DHPF_WIDTH) +#define M98090_DHF_MASK (1<<4) +#define M98090_DHF_SHIFT 4 +#define M98090_DHF_WIDTH 1 +#define M98090_FLT_DMIC34MODE_MASK (1<<3) +#define M98090_FLT_DMIC34MODE_SHIFT 3 +#define M98090_FLT_DMIC34MODE_WIDTH 1 +#define M98090_FLT_DMIC34HPF_MASK (1<<2) +#define M98090_FLT_DMIC34HPF_SHIFT 2 +#define M98090_FLT_DMIC34HPF_WIDTH 1 +#define M98090_FLT_DMIC34HPF_NUM (1<<M98090_FLT_DMIC34HPF_WIDTH) + +/* + * M98090_REG_DAI_PLAYBACK_LEVEL + */ +#define M98090_DVM_MASK (1<<7) +#define M98090_DVM_SHIFT 7 +#define M98090_DVM_WIDTH 1 +#define M98090_DVG_MASK (3<<4) +#define M98090_DVG_SHIFT 4 +#define M98090_DVG_WIDTH 2 +#define M98090_DVG_NUM (1<<M98090_DVG_WIDTH) +#define M98090_DV_MASK (15<<0) +#define M98090_DV_SHIFT 0 +#define M98090_DV_WIDTH 4 +#define M98090_DV_NUM (1<<M98090_DV_WIDTH) + +/* + * M98090_REG_DAI_PLAYBACK_LEVEL_EQ + */ +#define M98090_EQCLPN_MASK (1<<4) +#define M98090_EQCLPN_SHIFT 4 +#define M98090_EQCLPN_WIDTH 1 +#define M98090_EQCLPN_NUM (1<<M98090_EQCLPN_WIDTH) +#define M98090_DVEQ_MASK (15<<0) +#define M98090_DVEQ_SHIFT 0 +#define M98090_DVEQ_WIDTH 4 +#define M98090_DVEQ_NUM (1<<M98090_DVEQ_WIDTH) + +/* + * M98090_REG_LEFT_HP_MIXER + */ +#define M98090_MIXHPL_MIC2_MASK (1<<5) +#define M98090_MIXHPL_MIC2_SHIFT 5 +#define M98090_MIXHPL_MIC2_WIDTH 1 +#define M98090_MIXHPL_MIC1_MASK (1<<4) +#define M98090_MIXHPL_MIC1_SHIFT 4 +#define M98090_MIXHPL_MIC1_WIDTH 1 +#define M98090_MIXHPL_LINEB_MASK (1<<3) +#define M98090_MIXHPL_LINEB_SHIFT 3 +#define M98090_MIXHPL_LINEB_WIDTH 1 +#define M98090_MIXHPL_LINEA_MASK (1<<2) +#define M98090_MIXHPL_LINEA_SHIFT 2 +#define M98090_MIXHPL_LINEA_WIDTH 1 +#define M98090_MIXHPL_DACR_MASK (1<<1) +#define M98090_MIXHPL_DACR_SHIFT 1 +#define M98090_MIXHPL_DACR_WIDTH 1 +#define M98090_MIXHPL_DACL_MASK (1<<0) +#define M98090_MIXHPL_DACL_SHIFT 0 +#define M98090_MIXHPL_DACL_WIDTH 1 +#define M98090_MIXHPL_MASK (63<<0) +#define M98090_MIXHPL_SHIFT 0 +#define M98090_MIXHPL_WIDTH 6 + +/* + * M98090_REG_RIGHT_HP_MIXER + */ +#define M98090_MIXHPR_MIC2_MASK (1<<5) +#define M98090_MIXHPR_MIC2_SHIFT 5 +#define M98090_MIXHPR_MIC2_WIDTH 1 +#define M98090_MIXHPR_MIC1_MASK (1<<4) +#define M98090_MIXHPR_MIC1_SHIFT 4 +#define M98090_MIXHPR_MIC1_WIDTH 1 +#define M98090_MIXHPR_LINEB_MASK (1<<3) +#define M98090_MIXHPR_LINEB_SHIFT 3 +#define M98090_MIXHPR_LINEB_WIDTH 1 +#define M98090_MIXHPR_LINEA_MASK (1<<2) +#define M98090_MIXHPR_LINEA_SHIFT 2 +#define M98090_MIXHPR_LINEA_WIDTH 1 +#define M98090_MIXHPR_DACR_MASK (1<<1) +#define M98090_MIXHPR_DACR_SHIFT 1 +#define M98090_MIXHPR_DACR_WIDTH 1 +#define M98090_MIXHPR_DACL_MASK (1<<0) +#define M98090_MIXHPR_DACL_SHIFT 0 +#define M98090_MIXHPR_DACL_WIDTH 1 +#define M98090_MIXHPR_MASK (63<<0) +#define M98090_MIXHPR_SHIFT 0 +#define M98090_MIXHPR_WIDTH 6 + +/* + * M98090_REG_HP_CONTROL + */ +#define M98090_MIXHPRSEL_MASK (1<<5) +#define M98090_MIXHPRSEL_SHIFT 5 +#define M98090_MIXHPRSEL_WIDTH 1 +#define M98090_MIXHPLSEL_MASK (1<<4) +#define M98090_MIXHPLSEL_SHIFT 4 +#define M98090_MIXHPLSEL_WIDTH 1 +#define M98090_MIXHPRG_MASK (3<<2) +#define M98090_MIXHPRG_SHIFT 2 +#define M98090_MIXHPRG_WIDTH 2 +#define M98090_MIXHPRG_NUM (1<<M98090_MIXHPRG_WIDTH) +#define M98090_MIXHPLG_MASK (3<<0) +#define M98090_MIXHPLG_SHIFT 0 +#define M98090_MIXHPLG_WIDTH 2 +#define M98090_MIXHPLG_NUM (1<<M98090_MIXHPLG_WIDTH) + +/* + * M98090_REG_LEFT_HP_VOLUME + */ +#define M98090_HPLM_MASK (1<<7) +#define M98090_HPLM_SHIFT 7 +#define M98090_HPLM_WIDTH 1 +#define M98090_HPVOLL_MASK (31<<0) +#define M98090_HPVOLL_SHIFT 0 +#define M98090_HPVOLL_WIDTH 5 +#define M98090_HPVOLL_NUM (1<<M98090_HPVOLL_WIDTH) + +/* + * M98090_REG_RIGHT_HP_VOLUME + */ +#define M98090_HPRM_MASK (1<<7) +#define M98090_HPRM_SHIFT 7 +#define M98090_HPRM_WIDTH 1 +#define M98090_HPVOLR_MASK (31<<0) +#define M98090_HPVOLR_SHIFT 0 +#define M98090_HPVOLR_WIDTH 5 +#define M98090_HPVOLR_NUM (1<<M98090_HPVOLR_WIDTH) + +/* + * M98090_REG_LEFT_SPK_MIXER + */ +#define M98090_MIXSPL_MIC2_MASK (1<<5) +#define M98090_MIXSPL_MIC2_SHIFT 5 +#define M98090_MIXSPL_MIC2_WIDTH 1 +#define M98090_MIXSPL_MIC1_MASK (1<<4) +#define M98090_MIXSPL_MIC1_SHIFT 4 +#define M98090_MIXSPL_MIC1_WIDTH 1 +#define M98090_MIXSPL_LINEB_MASK (1<<3) +#define M98090_MIXSPL_LINEB_SHIFT 3 +#define M98090_MIXSPL_LINEB_WIDTH 1 +#define M98090_MIXSPL_LINEA_MASK (1<<2) +#define M98090_MIXSPL_LINEA_SHIFT 2 +#define M98090_MIXSPL_LINEA_WIDTH 1 +#define M98090_MIXSPL_DACR_MASK (1<<1) +#define M98090_MIXSPL_DACR_SHIFT 1 +#define M98090_MIXSPL_DACR_WIDTH 1 +#define M98090_MIXSPL_DACL_MASK (1<<0) +#define M98090_MIXSPL_DACL_SHIFT 0 +#define M98090_MIXSPL_DACL_WIDTH 1 +#define M98090_MIXSPL_MASK (63<<0) +#define M98090_MIXSPL_SHIFT 0 +#define M98090_MIXSPL_WIDTH 6 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 + + +/* + * M98090_REG_RIGHT_SPK_MIXER + */ +#define M98090_SPK_SLAVE_MASK (1<<6) +#define M98090_SPK_SLAVE_SHIFT 6 +#define M98090_SPK_SLAVE_WIDTH 1 +#define M98090_MIXSPR_MIC2_MASK (1<<5) +#define M98090_MIXSPR_MIC2_SHIFT 5 +#define M98090_MIXSPR_MIC2_WIDTH 1 +#define M98090_MIXSPR_MIC1_MASK (1<<4) +#define M98090_MIXSPR_MIC1_SHIFT 4 +#define M98090_MIXSPR_MIC1_WIDTH 1 +#define M98090_MIXSPR_LINEB_MASK (1<<3) +#define M98090_MIXSPR_LINEB_SHIFT 3 +#define M98090_MIXSPR_LINEB_WIDTH 1 +#define M98090_MIXSPR_LINEA_MASK (1<<2) +#define M98090_MIXSPR_LINEA_SHIFT 2 +#define M98090_MIXSPR_LINEA_WIDTH 1 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 +#define M98090_MIXSPR_DACL_MASK (1<<0) +#define M98090_MIXSPR_DACL_SHIFT 0 +#define M98090_MIXSPR_DACL_WIDTH 1 +#define M98090_MIXSPR_MASK (63<<0) +#define M98090_MIXSPR_SHIFT 0 +#define M98090_MIXSPR_WIDTH 6 + +/* + * M98090_REG_SPK_CONTROL + */ +#define M98090_MIXSPRG_MASK (3<<2) +#define M98090_MIXSPRG_SHIFT 2 +#define M98090_MIXSPRG_WIDTH 2 +#define M98090_MIXSPRG_NUM (1<<M98090_MIXSPRG_WIDTH) +#define M98090_MIXSPLG_MASK (3<<0) +#define M98090_MIXSPLG_SHIFT 0 +#define M98090_MIXSPLG_WIDTH 2 +#define M98090_MIXSPLG_NUM (1<<M98090_MIXSPLG_WIDTH) + +/* + * M98090_REG_LEFT_SPK_VOLUME + */ +#define M98090_SPLM_MASK (1<<7) +#define M98090_SPLM_SHIFT 7 +#define M98090_SPLM_WIDTH 1 +#define M98090_SPVOLL_MASK (63<<0) +#define M98090_SPVOLL_SHIFT 0 +#define M98090_SPVOLL_WIDTH 6 +#define M98090_SPVOLL_NUM 40 + +/* + * M98090_REG_RIGHT_SPK_VOLUME + */ +#define M98090_SPRM_MASK (1<<7) +#define M98090_SPRM_SHIFT 7 +#define M98090_SPRM_WIDTH 1 +#define M98090_SPVOLR_MASK (63<<0) +#define M98090_SPVOLR_SHIFT 0 +#define M98090_SPVOLR_WIDTH 6 +#define M98090_SPVOLR_NUM 40 + +/* + * M98090_REG_DRC_TIMING + */ +#define M98090_DRCEN_MASK (1<<7) +#define M98090_DRCEN_SHIFT 7 +#define M98090_DRCEN_WIDTH 1 +#define M98090_DRCEN_NUM (1<<M98090_DRCEN_WIDTH) +#define M98090_DRCRLS_MASK (7<<4) +#define M98090_DRCRLS_SHIFT 4 +#define M98090_DRCRLS_WIDTH 3 +#define M98090_DRCATK_MASK (7<<0) +#define M98090_DRCATK_SHIFT 0 +#define M98090_DRCATK_WIDTH 3 + +/* + * M98090_REG_DRC_COMPRESSOR + */ +#define M98090_DRCCMP_MASK (7<<5) +#define M98090_DRCCMP_SHIFT 5 +#define M98090_DRCCMP_WIDTH 3 +#define M98090_DRCTHC_MASK (31<<0) +#define M98090_DRCTHC_SHIFT 0 +#define M98090_DRCTHC_WIDTH 5 +#define M98090_DRCTHC_NUM (1<<M98090_DRCTHC_WIDTH) + +/* + * M98090_REG_DRC_EXPANDER + */ +#define M98090_DRCEXP_MASK (7<<5) +#define M98090_DRCEXP_SHIFT 5 +#define M98090_DRCEXP_WIDTH 3 +#define M98090_DRCTHE_MASK (31<<0) +#define M98090_DRCTHE_SHIFT 0 +#define M98090_DRCTHE_WIDTH 5 +#define M98090_DRCTHE_NUM (1<<M98090_DRCTHE_WIDTH) + +/* + * M98090_REG_DRC_GAIN + */ +#define M98090_DRCG_MASK (31<<0) +#define M98090_DRCG_SHIFT 0 +#define M98090_DRCG_WIDTH 5 +#define M98090_DRCG_NUM 13 + +/* + * M98090_REG_RCV_LOUTL_MIXER + */ +#define M98090_MIXRCVL_MIC2_MASK (1<<5) +#define M98090_MIXRCVL_MIC2_SHIFT 5 +#define M98090_MIXRCVL_MIC2_WIDTH 1 +#define M98090_MIXRCVL_MIC1_MASK (1<<4) +#define M98090_MIXRCVL_MIC1_SHIFT 4 +#define M98090_MIXRCVL_MIC1_WIDTH 1 +#define M98090_MIXRCVL_LINEB_MASK (1<<3) +#define M98090_MIXRCVL_LINEB_SHIFT 3 +#define M98090_MIXRCVL_LINEB_WIDTH 1 +#define M98090_MIXRCVL_LINEA_MASK (1<<2) +#define M98090_MIXRCVL_LINEA_SHIFT 2 +#define M98090_MIXRCVL_LINEA_WIDTH 1 +#define M98090_MIXRCVL_DACR_MASK (1<<1) +#define M98090_MIXRCVL_DACR_SHIFT 1 +#define M98090_MIXRCVL_DACR_WIDTH 1 +#define M98090_MIXRCVL_DACL_MASK (1<<0) +#define M98090_MIXRCVL_DACL_SHIFT 0 +#define M98090_MIXRCVL_DACL_WIDTH 1 +#define M98090_MIXRCVL_MASK (63<<0) +#define M98090_MIXRCVL_SHIFT 0 +#define M98090_MIXRCVL_WIDTH 6 + +/* + * M98090_REG_RCV_LOUTL_CONTROL + */ +#define M98090_MIXRCVLG_MASK (3<<0) +#define M98090_MIXRCVLG_SHIFT 0 +#define M98090_MIXRCVLG_WIDTH 2 +#define M98090_MIXRCVLG_NUM (1<<M98090_MIXRCVLG_WIDTH) + +/* + * M98090_REG_RCV_LOUTL_VOLUME + */ +#define M98090_RCVLM_MASK (1<<7) +#define M98090_RCVLM_SHIFT 7 +#define M98090_RCVLM_WIDTH 1 +#define M98090_RCVLVOL_MASK (31<<0) +#define M98090_RCVLVOL_SHIFT 0 +#define M98090_RCVLVOL_WIDTH 5 +#define M98090_RCVLVOL_NUM (1<<M98090_RCVLVOL_WIDTH) + +/* + * M98090_REG_LOUTR_MIXER + */ +#define M98090_LINMOD_MASK (1<<7) +#define M98090_LINMOD_SHIFT 7 +#define M98090_LINMOD_WIDTH 1 +#define M98090_MIXRCVR_MIC2_MASK (1<<5) +#define M98090_MIXRCVR_MIC2_SHIFT 5 +#define M98090_MIXRCVR_MIC2_WIDTH 1 +#define M98090_MIXRCVR_MIC1_MASK (1<<4) +#define M98090_MIXRCVR_MIC1_SHIFT 4 +#define M98090_MIXRCVR_MIC1_WIDTH 1 +#define M98090_MIXRCVR_LINEB_MASK (1<<3) +#define M98090_MIXRCVR_LINEB_SHIFT 3 +#define M98090_MIXRCVR_LINEB_WIDTH 1 +#define M98090_MIXRCVR_LINEA_MASK (1<<2) +#define M98090_MIXRCVR_LINEA_SHIFT 2 +#define M98090_MIXRCVR_LINEA_WIDTH 1 +#define M98090_MIXRCVR_DACR_MASK (1<<1) +#define M98090_MIXRCVR_DACR_SHIFT 1 +#define M98090_MIXRCVR_DACR_WIDTH 1 +#define M98090_MIXRCVR_DACL_MASK (1<<0) +#define M98090_MIXRCVR_DACL_SHIFT 0 +#define M98090_MIXRCVR_DACL_WIDTH 1 +#define M98090_MIXRCVR_MASK (63<<0) +#define M98090_MIXRCVR_SHIFT 0 +#define M98090_MIXRCVR_WIDTH 6 + +/* + * M98090_REG_LOUTR_CONTROL + */ +#define M98090_MIXRCVRG_MASK (3<<0) +#define M98090_MIXRCVRG_SHIFT 0 +#define M98090_MIXRCVRG_WIDTH 2 +#define M98090_MIXRCVRG_NUM (1<<M98090_MIXRCVRG_WIDTH) + +/* + * M98090_REG_LOUTR_VOLUME + */ +#define M98090_RCVRM_MASK (1<<7) +#define M98090_RCVRM_SHIFT 7 +#define M98090_RCVRM_WIDTH 1 +#define M98090_RCVRVOL_MASK (31<<0) +#define M98090_RCVRVOL_SHIFT 0 +#define M98090_RCVRVOL_WIDTH 5 +#define M98090_RCVRVOL_NUM (1<<M98090_RCVRVOL_WIDTH) + +/* + * M98090_REG_JACK_DETECT + */ +#define M98090_JDETEN_MASK (1<<7) +#define M98090_JDETEN_SHIFT 7 +#define M98090_JDETEN_WIDTH 1 +#define M98090_JDWK_MASK (1<<6) +#define M98090_JDWK_SHIFT 6 +#define M98090_JDWK_WIDTH 1 +#define M98090_JDEB_MASK (3<<0) +#define M98090_JDEB_SHIFT 0 +#define M98090_JDEB_WIDTH 2 +#define M98090_JDEB_25MS (0<<0) +#define M98090_JDEB_50MS (1<<0) +#define M98090_JDEB_100MS (2<<0) +#define M98090_JDEB_200MS (3<<0) + +/* + * M98090_REG_INPUT_ENABLE + */ +#define M98090_MBEN_MASK (1<<4) +#define M98090_MBEN_SHIFT 4 +#define M98090_MBEN_WIDTH 1 +#define M98090_LINEAEN_MASK (1<<3) +#define M98090_LINEAEN_SHIFT 3 +#define M98090_LINEAEN_WIDTH 1 +#define M98090_LINEBEN_MASK (1<<2) +#define M98090_LINEBEN_SHIFT 2 +#define M98090_LINEBEN_WIDTH 1 +#define M98090_ADREN_MASK (1<<1) +#define M98090_ADREN_SHIFT 1 +#define M98090_ADREN_WIDTH 1 +#define M98090_ADLEN_MASK (1<<0) +#define M98090_ADLEN_SHIFT 0 +#define M98090_ADLEN_WIDTH 1 + +/* + * M98090_REG_OUTPUT_ENABLE + */ +#define M98090_HPREN_MASK (1<<7) +#define M98090_HPREN_SHIFT 7 +#define M98090_HPREN_WIDTH 1 +#define M98090_HPLEN_MASK (1<<6) +#define M98090_HPLEN_SHIFT 6 +#define M98090_HPLEN_WIDTH 1 +#define M98090_SPREN_MASK (1<<5) +#define M98090_SPREN_SHIFT 5 +#define M98090_SPREN_WIDTH 1 +#define M98090_SPLEN_MASK (1<<4) +#define M98090_SPLEN_SHIFT 4 +#define M98090_SPLEN_WIDTH 1 +#define M98090_RCVLEN_MASK (1<<3) +#define M98090_RCVLEN_SHIFT 3 +#define M98090_RCVLEN_WIDTH 1 +#define M98090_RCVREN_MASK (1<<2) +#define M98090_RCVREN_SHIFT 2 +#define M98090_RCVREN_WIDTH 1 +#define M98090_DAREN_MASK (1<<1) +#define M98090_DAREN_SHIFT 1 +#define M98090_DAREN_WIDTH 1 +#define M98090_DALEN_MASK (1<<0) +#define M98090_DALEN_SHIFT 0 +#define M98090_DALEN_WIDTH 1 + +/* + * M98090_REG_LEVEL_CONTROL + */ +#define M98090_ZDENN_MASK (1<<2) +#define M98090_ZDENN_SHIFT 2 +#define M98090_ZDENN_WIDTH 1 +#define M98090_ZDENN_NUM (1<<M98090_ZDENN_WIDTH) +#define M98090_VS2ENN_MASK (1<<1) +#define M98090_VS2ENN_SHIFT 1 +#define M98090_VS2ENN_WIDTH 1 +#define M98090_VS2ENN_NUM (1<<M98090_VS2ENN_WIDTH) +#define M98090_VSENN_MASK (1<<0) +#define M98090_VSENN_SHIFT 0 +#define M98090_VSENN_WIDTH 1 +#define M98090_VSENN_NUM (1<<M98090_VSENN_WIDTH) + +/* + * M98090_REG_DSP_FILTER_ENABLE + */ +#define M98090_DMIC34BQEN_MASK (1<<4) +#define M98090_DMIC34BQEN_SHIFT 4 +#define M98090_DMIC34BQEN_WIDTH 1 +#define M98090_DMIC34BQEN_NUM (1<<M98090_DMIC34BQEN_WIDTH) +#define M98090_ADCBQEN_MASK (1<<3) +#define M98090_ADCBQEN_SHIFT 3 +#define M98090_ADCBQEN_WIDTH 1 +#define M98090_ADCBQEN_NUM (1<<M98090_ADCBQEN_WIDTH) +#define M98090_EQ3BANDEN_MASK (1<<2) +#define M98090_EQ3BANDEN_SHIFT 2 +#define M98090_EQ3BANDEN_WIDTH 1 +#define M98090_EQ3BANDEN_NUM (1<<M98090_EQ3BANDEN_WIDTH) +#define M98090_EQ5BANDEN_MASK (1<<1) +#define M98090_EQ5BANDEN_SHIFT 1 +#define M98090_EQ5BANDEN_WIDTH 1 +#define M98090_EQ5BANDEN_NUM (1<<M98090_EQ5BANDEN_WIDTH) +#define M98090_EQ7BANDEN_MASK (1<<0) +#define M98090_EQ7BANDEN_SHIFT 0 +#define M98090_EQ7BANDEN_WIDTH 1 +#define M98090_EQ7BANDEN_NUM (1<<M98090_EQ7BANDEN_WIDTH) + +/* + * M98090_REG_BIAS_CONTROL + */ +#define M98090_VCM_MODE_MASK (1<<0) +#define M98090_VCM_MODE_SHIFT 0 +#define M98090_VCM_MODE_WIDTH 1 +#define M98090_VCM_MODE_NUM (1<<M98090_VCM_MODE_WIDTH) + +/* + * M98090_REG_DAC_CONTROL + */ +#define M98090_PERFMODE_MASK (1<<1) +#define M98090_PERFMODE_SHIFT 1 +#define M98090_PERFMODE_WIDTH 1 +#define M98090_PERFMODE_NUM (1<<M98090_PERFMODE_WIDTH) +#define M98090_DACHP_MASK (1<<0) +#define M98090_DACHP_SHIFT 0 +#define M98090_DACHP_WIDTH 1 +#define M98090_DACHP_NUM (1<<M98090_DACHP_WIDTH) + +/* + * M98090_REG_ADC_CONTROL + */ +#define M98090_OSR128_MASK (1<<2) +#define M98090_OSR128_SHIFT 2 +#define M98090_OSR128_WIDTH 1 +#define M98090_ADCDITHER_MASK (1<<1) +#define M98090_ADCDITHER_SHIFT 1 +#define M98090_ADCDITHER_WIDTH 1 +#define M98090_ADCDITHER_NUM (1<<M98090_ADCDITHER_WIDTH) +#define M98090_ADCHP_MASK (1<<0) +#define M98090_ADCHP_SHIFT 0 +#define M98090_ADCHP_WIDTH 1 +#define M98090_ADCHP_NUM (1<<M98090_ADCHP_WIDTH) + +/* + * M98090_REG_DEVICE_SHUTDOWN + */ +#define M98090_SHDNN_MASK (1<<7) +#define M98090_SHDNN_SHIFT 7 +#define M98090_SHDNN_WIDTH 1 + +/* + * M98090_REG_EQUALIZER_BASE + */ +#define M98090_B0_1_HI_MASK (255<<0) +#define M98090_B0_1_HI_SHIFT 0 +#define M98090_B0_1_HI_WIDTH 8 +#define M98090_B0_1_MID_MASK (255<<0) +#define M98090_B0_1_MID_SHIFT 0 +#define M98090_B0_1_MID_WIDTH 8 +#define M98090_B0_1_LO_MASK (255<<0) +#define M98090_B0_1_LO_SHIFT 0 +#define M98090_B0_1_LO_WIDTH 8 +#define M98090_B1_1_HI_MASK (255<<0) +#define M98090_B1_1_HI_SHIFT 0 +#define M98090_B1_1_HI_WIDTH 8 +#define M98090_B1_1_MID_MASK (255<<0) +#define M98090_B1_1_MID_SHIFT 0 +#define M98090_B1_1_MID_WIDTH 8 +#define M98090_B1_1_LO_MASK (255<<0) +#define M98090_B1_1_LO_SHIFT 0 +#define M98090_B1_1_LO_WIDTH 8 +#define M98090_B2_1_HI_MASK (255<<0) +#define M98090_B2_1_HI_SHIFT 0 +#define M98090_B2_1_HI_WIDTH 8 +#define M98090_B2_1_MID_MASK (255<<0) +#define M98090_B2_1_MID_SHIFT 0 +#define M98090_B2_1_MID_WIDTH 8 +#define M98090_B2_1_LO_MASK (255<<0) +#define M98090_B2_1_LO_SHIFT 0 +#define M98090_B2_1_LO_WIDTH 8 +#define M98090_A1_1_HI_MASK (255<<0) +#define M98090_A1_1_HI_SHIFT 0 +#define M98090_A1_1_HI_WIDTH 8 +#define M98090_A1_1_MID_MASK (255<<0) +#define M98090_A1_1_MID_SHIFT 0 +#define M98090_A1_1_MID_WIDTH 8 +#define M98090_A1_1_LO_MASK (255<<0) +#define M98090_A1_1_LO_SHIFT 0 +#define M98090_A1_1_LO_WIDTH 8 +#define M98090_A2_1_HI_MASK (255<<0) +#define M98090_A2_1_HI_SHIFT 0 +#define M98090_A2_1_HI_WIDTH 8 +#define M98090_A2_1_MID_MASK (255<<0) +#define M98090_A2_1_MID_SHIFT 0 +#define M98090_A2_1_MID_WIDTH 8 +#define M98090_A2_1_LO_MASK (255<<0) +#define M98090_A2_1_LO_SHIFT 0 +#define M98090_A2_1_LO_WIDTH 8 + +#define M98090_COEFS_PER_BAND 5 +#define M98090_COEFS_BLK_SZ (M98090_COEFS_PER_BAND * 3) +#define M98090_COEFS_MAX_SZ (M98090_COEFS_BLK_SZ * 7) + +/* + * M98090_REG_RECORD_BIQUAD_BASE + */ +#define M98090_REC_B0_HI_MASK (255<<0) +#define M98090_REC_B0_HI_SHIFT 0 +#define M98090_REC_B0_HI_WIDTH 8 +#define M98090_REC_B0_MID_MASK (255<<0) +#define M98090_REC_B0_MID_SHIFT 0 +#define M98090_REC_B0_MID_WIDTH 8 +#define M98090_REC_B0_LO_MASK (255<<0) +#define M98090_REC_B0_LO_SHIFT 0 +#define M98090_REC_B0_LO_WIDTH 8 +#define M98090_REC_B1_HI_MASK (255<<0) +#define M98090_REC_B1_HI_SHIFT 0 +#define M98090_REC_B1_HI_WIDTH 8 +#define M98090_REC_B1_MID_MASK (255<<0) +#define M98090_REC_B1_MID_SHIFT 0 +#define M98090_REC_B1_MID_WIDTH 8 +#define M98090_REC_B1_LO_MASK (255<<0) +#define M98090_REC_B1_LO_SHIFT 0 +#define M98090_REC_B1_LO_WIDTH 8 +#define M98090_REC_B2_HI_MASK (255<<0) +#define M98090_REC_B2_HI_SHIFT 0 +#define M98090_REC_B2_HI_WIDTH 8 +#define M98090_REC_B2_MID_MASK (255<<0) +#define M98090_REC_B2_MID_SHIFT 0 +#define M98090_REC_B2_MID_WIDTH 8 +#define M98090_REC_B2_LO_MASK (255<<0) +#define M98090_REC_B2_LO_SHIFT 0 +#define M98090_REC_B2_LO_WIDTH 8 +#define M98090_REC_A1_HI_MASK (255<<0) +#define M98090_REC_A1_HI_SHIFT 0 +#define M98090_REC_A1_HI_WIDTH 8 +#define M98090_REC_A1_MID_MASK (255<<0) +#define M98090_REC_A1_MID_SHIFT 0 +#define M98090_REC_A1_MID_WIDTH 8 +#define M98090_REC_A1_LO_MASK (255<<0) +#define M98090_REC_A1_LO_SHIFT 0 +#define M98090_REC_A1_LO_WIDTH 8 +#define M98090_REC_A2_HI_MASK (255<<0) +#define M98090_REC_A2_HI_SHIFT 0 +#define M98090_REC_A2_HI_WIDTH 8 +#define M98090_REC_A2_MID_MASK (255<<0) +#define M98090_REC_A2_MID_SHIFT 0 +#define M98090_REC_A2_MID_WIDTH 8 +#define M98090_REC_A2_LO_MASK (255<<0) +#define M98090_REC_A2_LO_SHIFT 0 +#define M98090_REC_A2_LO_WIDTH 8 + +/* + * M98090_REG_DMIC3_VOLUME + */ +#define M98090_DMIC_AV3G_MASK (7<<4) +#define M98090_DMIC_AV3G_SHIFT 4 +#define M98090_DMIC_AV3G_WIDTH 3 +#define M98090_DMIC_AV3G_NUM (1<<M98090_DMIC_AV3G_WIDTH) +#define M98090_DMIC_AV3_MASK (15<<0) +#define M98090_DMIC_AV3_SHIFT 0 +#define M98090_DMIC_AV3_WIDTH 4 +#define M98090_DMIC_AV3_NUM (1<<M98090_DMIC_AV3_WIDTH) + +/* + * M98090_REG_DMIC4_VOLUME + */ +#define M98090_DMIC_AV4G_MASK (7<<4) +#define M98090_DMIC_AV4G_SHIFT 4 +#define M98090_DMIC_AV4G_WIDTH 3 +#define M98090_DMIC_AV4G_NUM (1<<M98090_DMIC_AV4G_WIDTH) +#define M98090_DMIC_AV4_MASK (15<<0) +#define M98090_DMIC_AV4_SHIFT 0 +#define M98090_DMIC_AV4_WIDTH 4 +#define M98090_DMIC_AV4_NUM (1<<M98090_DMIC_AV4_WIDTH) + +/* + * M98090_REG_DMIC34_BQ_PREATTEN + */ +#define M98090_AV34BQ_MASK (15<<0) +#define M98090_AV34BQ_SHIFT 0 +#define M98090_AV34BQ_WIDTH 4 +#define M98090_AV34BQ_NUM (1<<M98090_AV34BQ_WIDTH) + +/* + * M98090_REG_RECORD_TDM_SLOT + */ +#define M98090_TDM_SLOTADCL_MASK (3<<6) +#define M98090_TDM_SLOTADCL_SHIFT 6 +#define M98090_TDM_SLOTADCL_WIDTH 2 +#define M98090_TDM_SLOTADCL_NUM (1<<M98090_TDM_SLOTADCL_WIDTH) +#define M98090_TDM_SLOTADCR_MASK (3<<4) +#define M98090_TDM_SLOTADCR_SHIFT 4 +#define M98090_TDM_SLOTADCR_WIDTH 2 +#define M98090_TDM_SLOTADCR_NUM (1<<M98090_TDM_SLOTADCR_WIDTH) +#define M98090_TDM_SLOTDMIC3_MASK (3<<2) +#define M98090_TDM_SLOTDMIC3_SHIFT 2 +#define M98090_TDM_SLOTDMIC3_WIDTH 2 +#define M98090_TDM_SLOTDMIC3_NUM (1<<M98090_TDM_SLOTDMIC3_WIDTH) +#define M98090_TDM_SLOTDMIC4_MASK (3<<0) +#define M98090_TDM_SLOTDMIC4_SHIFT 0 +#define M98090_TDM_SLOTDMIC4_WIDTH 2 +#define M98090_TDM_SLOTDMIC4_NUM (1<<M98090_TDM_SLOTDMIC4_WIDTH) + +/* + * M98090_REG_SAMPLE_RATE + */ +#define M98090_DMIC34_ZEROPAD_MASK (1<<4) +#define M98090_DMIC34_ZEROPAD_SHIFT 4 +#define M98090_DMIC34_ZEROPAD_WIDTH 1 +#define M98090_DMIC34_ZEROPAD_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DMIC34_SRDIV_MASK (7<<0) +#define M98090_DMIC34_SRDIV_SHIFT 0 +#define M98090_DMIC34_SRDIV_WIDTH 3 + +/* + * M98090_REG_DMIC34_BIQUAD_BASE + */ +#define M98090_DMIC34_B0_HI_MASK (255<<0) +#define M98090_DMIC34_B0_HI_SHIFT 0 +#define M98090_DMIC34_B0_HI_WIDTH 8 +#define M98090_DMIC34_B0_MID_MASK (255<<0) +#define M98090_DMIC34_B0_MID_SHIFT 0 +#define M98090_DMIC34_B0_MID_WIDTH 8 +#define M98090_DMIC34_B0_LO_MASK (255<<0) +#define M98090_DMIC34_B0_LO_SHIFT 0 +#define M98090_DMIC34_B0_LO_WIDTH 8 +#define M98090_DMIC34_B1_HI_MASK (255<<0) +#define M98090_DMIC34_B1_HI_SHIFT 0 +#define M98090_DMIC34_B1_HI_WIDTH 8 +#define M98090_DMIC34_B1_MID_MASK (255<<0) +#define M98090_DMIC34_B1_MID_SHIFT 0 +#define M98090_DMIC34_B1_MID_WIDTH 8 +#define M98090_DMIC34_B1_LO_MASK (255<<0) +#define M98090_DMIC34_B1_LO_SHIFT 0 +#define M98090_DMIC34_B1_LO_WIDTH 8 +#define M98090_DMIC34_B2_HI_MASK (255<<0) +#define M98090_DMIC34_B2_HI_SHIFT 0 +#define M98090_DMIC34_B2_HI_WIDTH 8 +#define M98090_DMIC34_B2_MID_MASK (255<<0) +#define M98090_DMIC34_B2_MID_SHIFT 0 +#define M98090_DMIC34_B2_MID_WIDTH 8 +#define M98090_DMIC34_B2_LO_MASK (255<<0) +#define M98090_DMIC34_B2_LO_SHIFT 0 +#define M98090_DMIC34_B2_LO_WIDTH 8 +#define M98090_DMIC34_A1_HI_MASK (255<<0) +#define M98090_DMIC34_A1_HI_SHIFT 0 +#define M98090_DMIC34_A1_HI_WIDTH 8 +#define M98090_DMIC34_A1_MID_MASK (255<<0) +#define M98090_DMIC34_A1_MID_SHIFT 0 +#define M98090_DMIC34_A1_MID_WIDTH 8 +#define M98090_DMIC34_A1_LO_MASK (255<<0) +#define M98090_DMIC34_A1_LO_SHIFT 0 +#define M98090_DMIC34_A1_LO_WIDTH 8 +#define M98090_DMIC34_A2_HI_MASK (255<<0) +#define M98090_DMIC34_A2_HI_SHIFT 0 +#define M98090_DMIC34_A2_HI_WIDTH 8 +#define M98090_DMIC34_A2_MID_MASK (255<<0) +#define M98090_DMIC34_A2_MID_SHIFT 0 +#define M98090_DMIC34_A2_MID_WIDTH 8 +#define M98090_DMIC34_A2_LO_MASK (255<<0) +#define M98090_DMIC34_A2_LO_SHIFT 0 +#define M98090_DMIC34_A2_LO_WIDTH 8 + +#define M98090_JACK_STATE_NO_HEADSET 0 +#define M98090_JACK_STATE_NO_HEADSET_2 1 +#define M98090_JACK_STATE_HEADPHONE 2 +#define M98090_JACK_STATE_HEADSET 3 + +/* + * M98090_REG_REVISION_ID + */ +#define M98090_REVID_MASK (255<<0) +#define M98090_REVID_SHIFT 0 +#define M98090_REVID_WIDTH 8 +#define M98090_REVID_NUM (1<<M98090_REVID_WIDTH) + +#define M98090_BYTE1(w) ((w >> 8) & 0xff) +#define M98090_BYTE0(w) (w & 0xff) + +/* Silicon revision number */ +#define M98090_REVA 0x40 +#define M98091_REVA 0x50 + +enum max98090_type { + MAX98090, + MAX98091, +}; + +struct max98090_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98090_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + enum max98090_type devtype; + void *control_data; + struct max98090_pdata *pdata; + unsigned int sysclk; + unsigned int bclk; + unsigned int lrclk; + struct max98090_cdata dai[1]; + int irq; + int jack_state; + struct delayed_work jack_work; + struct snd_soc_jack *jack; + unsigned int dai_fmt; + int tdm_slots; + int tdm_width; + u8 lin_state; + unsigned int pa1en; + unsigned int pa2en; + unsigned int extmic_mux; + unsigned int sidetone; +}; + +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 1.7.0.4

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[alsa-devel] crashes in namehint
by Harald Sitter 31 Jan '13

31 Jan '13

aloha, I am working on kde multimedia and we are getting a considerable amount of crash reports from x86 ubuntu and fedora users. https://bugs.kde.org/show_bug.cgi?id=268185 http://notes.kde.org/alsa-bugs these crashes appear to be somewhat limited to x86 with only about 2 reports seeing this happen on x86_64. most of the reports suggest that it happens at startup and in particular for applications that tend to query ALSA devices shortly after startup (in fact, out of main() via a direct call chain). there are 3 different reports of which a sigsev in strcpy and sprintf seem most concerning. both appear in try_config() which was called with a malformed `name` string. now at least the sprintf and the strcpy crashes I can somewhat reliably reproduce on kubuntu 12.10 x86 in a virtualbox vm using the command 'kcmshell4 phonon' with phonon-backend-vlc installed and selected as default. and it appears that those are somehow related to how ubuntu does the pulseaudio integration... in /usr/share/alsa/alsa.conf all of alsa.conf.d/* is included, then there is alsa.conf.d/pulse.conf which contains > hook_func.pulse_load_if_running { > lib "libasound_module_conf_pulse.so" > func "conf_pulse_hook_load_if_running" > } > > @hooks [ > { > func pulse_load_if_running > files [ > "/usr/share/alsa/pulse-alsa.conf" > ] > errors false > } > ] so they load /usr/share/alsa/pulse-alsa.conf after checking PA is running. that file in turn contains the default pulse setup as seen in alsa-plugins [1] now with that lineup I can crash the kcmshell binary about 9/10 times with the strcpy sigsev where name in try_config is 0. if I replace the conditional load in pulse.conf with an explicit one (func load) it crashes about half as often but with the sprintf sigsev in try_config (name now being out-of-bounds rather than 0). if the content of pulse-alsa.conf (i.e. the actual config) is copied into pulse.conf the binary does not crash *at all*. quite the mysterious problem. however, since the crash disappears entirely when removing the second load (i.e. instead of alsa.conf->alsa.conf.d/pulse.conf->pulse-alsa.conf one were to use alsa.conf->alsa.conf.d/pulse.conf), I am lead to believe that this is either a problem caused by the second level of loading in general or a timing problem at large. most likely it is a combination of both where the additional load() calls introduce sufficient overhead to trigger the timing problem. ideas/thoughts/fixes welcome :S HS [1] http://git.alsa-project.org/?p=alsa-plugins.git;a=blob;f=pulse/99-pulseaudi…

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[alsa-devel] [PATCH] ASoC: wm2200: Add ANC source select for outputs
by Chris Rattray 31 Jan '13

31 Jan '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 0181444..a850d88 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1162,6 +1162,14 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("OUT1L ANC Switch", WM2200_DAC_VOLUME_LIMIT_1L, + WM2200_OUT1L_ANC_SRC_SHIFT, 1, 0), +SOC_SINGLE("OUT1R ANC Switch", WM2200_DAC_VOLUME_LIMIT_1R, + WM2200_OUT1R_ANC_SRC_SHIFT, 1, 0), +SOC_SINGLE("OUT2L ANC Switch", WM2200_DAC_VOLUME_LIMIT_2L, + WM2200_OUT2L_ANC_SRC_SHIFT, 1, 0), +SOC_SINGLE("OUT2R ANC Switch", WM2200_DAC_VOLUME_LIMIT_2R, + WM2200_OUT2R_ANC_SRC_SHIFT, 1, 0), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- 1.7.2.5

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[alsa-devel] Previous Post on PandaBoard
by Brent Weatherall 31 Jan '13

31 Jan '13

Hello, My previous post about PandaBoardES audio had one mistake. The code regarding the min/max values was a result of stepping through the 'snd_pcm_hw_params' call and NOT the 'snd_pcm_set_rate_near' call. My apologies for the bad information. Thank you. Regards, Brent

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[alsa-devel] [PATCH 0/3] Make snd_device_name_hint() reentrant
by Jerome Forissier 31 Jan '13

31 Jan '13

The following 3 patches ensure that snd_device_name_hint() does not mdify any global variable. They fix spurious error messages that may occur when the function is called from multiple threads simultaneously. Jerome Forissier (3): Add snd_lib_error_set_local() to install a thread-local error handler. snd_device_name_hint(): do not change the global error handler. snd_device_name_hint(): do not use global snd_config. configure.in | 10 ++++++++ include/error.h | 6 +++++ src/control/namehint.c | 67 ++++++++++++++++++++++++++++---------------------- src/error.c | 24 +++++++++++++++++- 4 files changed, 76 insertions(+), 31 deletions(-) -- 1.8.1.2

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[alsa-devel] [PATCH v3 0/4] ASoC davinci/tlv320aic3x updates, fixes & DT support
by Hebbar Gururaja 31 Jan '13

31 Jan '13

1. Convert tlv320aic3x mic bias to a supply widget and related machine driver 2. Add DT support for Davinci machine platform 3. remove __dev* attributes This patch-set is tested on Davinci platform (DA850 EVM). This series applies on top of tag next-20130128 git tree https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git These patches cannot be tested directly on mainline kernel (DT based). This is because dma is not yet completely ported to Davinci ASoC. I had to add few hacks to get Audio working on Davinci DA850 platform. For reference, I have placed the entire working tree @ Tree : https://github.com/ghebbar/linux.git Branch : dev_next-20130128_davinci_asoc_dt_v3 changes in v3 - drop mcasp pinctrl handler patch from seried as it is now handled centrally. - Convert tlv320aic3x mic bias to a supply widget and related machine driver - remove __dev* attributes Changes in V2 - Remove reference to Linux & software details from DT binding Hebbar Gururaja (3): ASoC: tlv320aic3x: Convert mic bias to a supply widget ASoC: davinci: update machine driver dapm routes ASoC: davinci: remove __dev* attributes Hebbar, Gururaja (1): ASoC: davinci: machine: Add device tree binding .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 6 + include/sound/tlv320aic3x.h | 10 ++ sound/soc/codecs/tlv320aic3x.c | 83 ++++++++- sound/soc/codecs/tlv320aic3x.h | 4 + sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++--- 6 files changed, 312 insertions(+), 29 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] Fix thread safety issues
by Jerome Forissier 31 Jan '13

31 Jan '13

This is a set of two patches to fix thread safety issues I found when running a video application over ALSA. The application uses LibVLC to play several videos simultaneously. LibVLC calls snd_pcm_open() and snd_device_name_hint() from multiple threads, without locking--which is allowed according to http://www.alsa-project.org/main/index.php/SMP_Design. Without these patches, I would get random error messages like: ALSA lib confmisc.c:768:(parse_card) cannot find card '$CARD' ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:4663:(snd_config_expand) Evaluate error: No such device Jerome Forissier (2): Make snd_device_name_hint() thread-safe by locking a mutex snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() src/control/namehint.c | 31 +++++++++++++++++++++++++++++++ src/pcm/pcm_direct.c | 15 +++++++++++---- 2 files changed, 42 insertions(+), 4 deletions(-) -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Convert mic bias to a supply widget
by Hebbar Gururaja 31 Jan '13

31 Jan '13

Convert MicBias widgets to supply widget. On tlv320aic3x, Mic bias power on/off shares the same register bits with output mic bias voltage. So, when power on mic bias, we need reclaim it to voltage value. Provide a new platform data so that the micbias voltage can be sent according to board requirement. Now since tlc320aic3x codec driver is DT aware, update dt files and functions to handle this new "micbias-vg" platform data. Because of sharing of bits, when enabling the micbias, voltage also needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD macro to create an event to handle this. Signed-off-by: Hebbar Gururaja <gururaja.hebbar(a)ti.com> --- Mark, I have updated the codec driver as you suggested. Can you confirm if this is sufficient?. If yes, then I will send all the patch once again with new revision. :100644 100644 e7b98f4... f47c3f5... M Documentation/devicetree/bindings/sound/tlv320aic3x.txt :100644 100644 ffd9bc7... 9407fd0... M include/sound/tlv320aic3x.h :100644 100644 4989143... fe6dd27... M sound/soc/codecs/tlv320aic3x.c :100644 100644 6db3c41... e521ac3... M sound/soc/codecs/tlv320aic3x.h .../devicetree/bindings/sound/tlv320aic3x.txt | 6 ++ include/sound/tlv320aic3x.h | 10 +++ sound/soc/codecs/tlv320aic3x.c | 85 ++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.h | 4 + 4 files changed, 99 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index e7b98f4..f47c3f5 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -11,6 +11,12 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality +- ai3x-micbias-vg - MicBias Voltage required. + 1 - MICBIAS output is powered to 2.0V, + 2 - MICBIAS output is powered to 2.5V, + 3 - MICBIAS output is connected to AVDD, + If this node is not mentioned or if the value is incorrect, then MicBias + is powered down. Example: diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index ffd9bc7..9407fd0 100644 --- a/include/sound/tlv320aic3x.h +++ b/include/sound/tlv320aic3x.h @@ -46,6 +46,13 @@ enum { AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 }; +enum aic3x_micbias_voltage { + AIC3X_MICBIAS_OFF = 0, + AIC3X_MICBIAS_2_0V = 1, + AIC3X_MICBIAS_2_5V = 2, + AIC3X_MICBIAS_AVDDV = 3, +}; + struct aic3x_setup_data { unsigned int gpio_func[2]; }; @@ -53,6 +60,9 @@ struct aic3x_setup_data { struct aic3x_pdata { int gpio_reset; /* < 0 if not used */ struct aic3x_setup_data *setup; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; #endif diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4989143..fe6dd27 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -85,6 +85,9 @@ struct aic3x_priv { #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 u16 model; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; /* @@ -195,6 +198,37 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, return ret; } +/* + * mic bias power on/off share the same register bits with + * output voltage of mic bias. when power on mic bias, we + * need reclaim it to voltage value. + * 0x0 = Powered off + * 0x1 = MICBIAS output is powered to 2.0V, + * 0x2 = MICBIAS output is powered to 2.5V + * 0x3 = MICBIAS output is connected to AVDD + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + aic3x->micbias_vg << MICBIAS_LEVEL_SHIFT); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, 0); + break; + } + return 0; +} + static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; static const char *aic3x_left_hpcom_mux[] = @@ -596,12 +630,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), /* Mic Bias */ - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", - MICBIAS_CTRL, 6, 3, 1, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", - MICBIAS_CTRL, 6, 3, 2, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", - MICBIAS_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, @@ -1386,6 +1417,24 @@ static int aic3x_probe(struct snd_soc_codec *codec) if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + /* set mic bias voltage */ + switch (aic3x->micbias_vg) { + case AIC3X_MICBIAS_2_0V: + case AIC3X_MICBIAS_2_5V: + case AIC3X_MICBIAS_AVDDV: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + (aic3x->micbias_vg) << MICBIAS_LEVEL_SHIFT); + break; + case AIC3X_MICBIAS_OFF: + /* + * noting to do. target won't enter here. This is just to avoid + * compile time warning "warning: enumeration value + * 'AIC3X_MICBIAS_OFF' not handled in switch" + */ + break; + } + aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); @@ -1461,6 +1510,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; int ret; + u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1474,6 +1524,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (pdata) { aic3x->gpio_reset = pdata->gpio_reset; aic3x->setup = pdata->setup; + aic3x->micbias_vg = pdata->micbias_vg; } else if (np) { ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), GFP_KERNEL); @@ -1493,6 +1544,28 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->setup = ai3x_setup; } + if (!of_property_read_u32(np, "ai3x-micbias-vg", &value)) { + switch (value) { + case 1 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_0V; + break; + case 2 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_5V; + break; + case 3 : + aic3x->micbias_vg = AIC3X_MICBIAS_AVDDV; + break; + default : + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + dev_err(&i2c->dev, "Unsuitable MicBias voltage " + "found in DT\n"); + } + } else { + dev_warn(&i2c->dev, "No MicBias voltage found in DT - " + "using default\n"); + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + } + } else { aic3x->gpio_reset = -1; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6db3c41..e521ac3 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -238,6 +238,10 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* MICBIAS Control Register */ +#define MICBIAS_LEVEL_SHIFT (6) +#define MICBIAS_LEVEL_MASK (3 << 6) + /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) -- 1.7.9.5

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[alsa-devel] [PATCH] Asoc: Samsung: I2S: Add support for runtime S2R
by R. Chandrasekar 31 Jan '13

31 Jan '13

From: "R. Chandrasekar" <rcsekar(a)samsung.com> This patch adds runtime suspend to resume support for I2S. I2S clk is disabled at suspend and enabled at resume. Signed-off-by: R. Chandrasekar <rcsekar(a)samsung.com> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- sound/soc/samsung/i2s.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2fc42f9..d7231e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1071,6 +1071,26 @@ static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) return platform_get_device_id(pdev)->driver_data; } +#ifdef CONFIG_PM_RUNTIME +static int i2s_runtime_suspend(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + + return 0; +} + +static int i2s_runtime_resume(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_prepare_enable(i2s->clk); + + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; @@ -1288,6 +1308,11 @@ static const struct of_device_id exynos_i2s_match[] = { MODULE_DEVICE_TABLE(of, exynos_i2s_match); #endif +static const struct dev_pm_ops samsung_i2s_pm = { + SET_RUNTIME_PM_OPS(i2s_runtime_suspend, + i2s_runtime_resume, NULL) +}; + static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, .remove = samsung_i2s_remove, @@ -1296,6 +1321,7 @@ static struct platform_driver samsung_i2s_driver = { .name = "samsung-i2s", .owner = THIS_MODULE, .of_match_table = of_match_ptr(exynos_i2s_match), + .pm = &samsung_i2s_pm, }, }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: fsl: Fix multiple symbol definitions
by Thierry Reding 31 Jan '13

31 Jan '13

Commit 25b8d31 (ASoC: fsl: fix multiple definition of init_module) fixed a build error due to multiple symbol definitions when building as a module. However, it causes the build to break when the driver is builtin because the imx-pcm.o object is included multiple times. Solve the issue by adding imx-pcm.o to each of the modules that require it, but only add it to the kernel once when one or both drivers are builtin. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/soc/fsl/Makefile | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index ec14579..8446d49 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -30,21 +30,24 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o -snd-soc-imx-pcm-objs := imx-pcm.o -ifneq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),) - snd-soc-imx-pcm-objs += imx-pcm-fiq.o -endif -ifneq ($(CONFIG_SND_SOC_IMX_PCM_DMA),) - snd-soc-imx-pcm-objs += imx-pcm-dma.o -endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o -snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o +snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o +ifeq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),m) +snd-soc-imx-pcm-fiq-y += imx-pcm.o +else +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm.o +endif obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o -snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o +snd-soc-imx-pcm-dma-y := imx-pcm-dma.o +ifeq ($(CONFIG_SND_SOC_IMX_PCM_DMA),m) +snd-soc-imx-pcm-dma-y := imx-pcm.o +else +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm.o +endif # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o -- 1.8.1.2

3 3

[alsa-devel] volume adjust doesn't work after calling many times
by Keyon.Jey 31 Jan '13

31 Jan '13

hi, all, after I use UI interface with pavucontrol to tune output devices volume rapidly many times, the volume adjust seems doesn't work! error occur in dmesg: [ 568.807025] ALSA sound/pci/hda/hda_intel.c:716: azx_get_response timeout, switching to polling mode: last cmd=0x001f000a [ 569.808026] ALSA sound/pci/hda/hda_intel.c:755: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x001f000a realtek ALC888 codec. alsa version 1.0.23 attached the alsa-info output. any comment? BR, Keyon

1 0

Re: [alsa-devel] [PATCH 3/4] ASOC: imx: add machine driver using wm8962 codec
by Mark Brown 31 Jan '13

31 Jan '13

On Thu, Jan 31, 2013 at 03:02:21AM +0000, Zhang Quan-B13634 wrote: > > > + if (!codec_dai->active) > > > + clk_enable(data->codec_clk); > > Should be clk_prepare_enable() but it's not clear to me why you need > > this... Also the clock API does refcounting so there should be no need > > to check for active, you should get matching startups and shutdowns. > > > > This will also fail for analogue bypass paths, set_bias_level() would be a good > > place to cover those. > [Gary-b13634] for power saving consideration. For now remove this feature to > Implement a basic feature I get why you're doing this at all but doing it in startup seems odd - like I say set_bias_level() is the most idiomatic place to implement this usually. There's a few machine drivers doing that you should be able to refer to for examples.

1 0

Re: [alsa-devel] Sound/ARM/i.MX/SSI: fix SDMA starvation
by Shawn Guo 31 Jan '13

31 Jan '13

Thanks for the fix, Juergen. The patch subject could be written in a better form with a proper prefix added. "git log sound/soc/fsl/fsl_ssi.c" should give example. To have the patch get in sooner, you should copy Mark Brown on it. Also MAINTAINERS file gives the following, so you still do not have the right list copied. SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lrg(a)ti.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported F: sound/soc/ F: include/sound/soc* On Wed, Jan 30, 2013 at 02:44:25PM +0100, Juergen Beisert wrote: > Juergen Beisert wrote: > > In cases when capturing is already running and someone enables also > > playback, the SDMA unit of the i.MX SoC does not see an edge on its TX DMA > > request line. This is due to the TX FIFOs are empty, and this request line > > is active all the time, when the CCSR_SSI_SIER_TDMAE bit is set. > > In this case also the TX FIFO underrun interrupt will flood the system > > forever. > > > > To ensure the first edge happens after enabling the TX side of the SSI unit > > and to trigger the SDMA unit successfully enable the CCSR_SSI_SIER_TDMAE on > > demand only. To be sure, this patch does the same for the > > CCSR_SSI_SIER_RDMAE (RX side) bit. > > > > Signed-off-by: Juergen Beisert <jbe(a)pengutronix.de> I can see the patch does fix the problem. A small comment below, otherwise Acked-by: Shawn Guo <shawn.guo(a)linaro.org> > > > > --- > > sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++++++-------- > > 1 file changed, 16 insertions(+), 8 deletions(-) > > > > Index: linux.git/sound/soc/fsl/fsl_ssi.c > > =================================================================== > > --- linux.git.orig/sound/soc/fsl/fsl_ssi.c > > +++ linux.git/sound/soc/fsl/fsl_ssi.c > > @@ -86,10 +86,9 @@ static inline void write_ssi_mask(u32 __ > > #endif > > > > /* SIER bitflag of interrupts to enable */ > > -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ > > - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ > > - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ > > - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ > > +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TIE | \ > > + CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TUE1_EN | \ > > + CCSR_SSI_SIER_RFRC_EN | CCSR_SSI_SIER_RIE | \ It takes me sometime to see the real change here is to remove flags CCSR_SSI_SIER_TDMAE and CCSR_SSI_SIER_RDMAE. Then the changes below should look less dramatic and ease the reviewers a little bit, I think. Shawn --8<--- @@ -86,10 +86,9 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) #endif /* SIER bitflag of interrupts to enable */ -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_RIE | \ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) > > CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) > > > > /** > > @@ -489,6 +488,9 @@ static int fsl_ssi_hw_params(struct snd_ > > * > > * The DMA channel is in external master start and pause mode, which > > * means the SSI completely controls the flow of data. > > + * > > + * To use the i.MX's SDMA unit it is important to create an edge on the > > + * corresponding request line. Otherwise the request will be ignored! > > */ > > static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, > > struct snd_soc_dai *dai) > > @@ -500,20 +502,26 @@ static int fsl_ssi_trigger(struct snd_pc > > switch (cmd) { > > case SNDRV_PCM_TRIGGER_START: > > case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > > - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > > write_ssi_mask(&ssi->scr, 0, > > CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); > > - else > > + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TDMAE); > > + } else { > > write_ssi_mask(&ssi->scr, 0, > > CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); > > + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RDMAE); > > + } > > break; > > > > case SNDRV_PCM_TRIGGER_STOP: > > case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > > - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > > write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); > > - else > > + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TDMAE, 0); > > + } else { > > write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); > > + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RDMAE, 0); > > + } > > break; > > > > default: > > Any comments on this patch? > > jbe > > -- > Pengutronix e.K. | Juergen Beisert | > Linux Solutions for Science and Industry | Phone: +49-5121-206917-5128 | > Peiner Str. 6-8, 31137 Hildesheim, Germany | Fax: +49-5121-206917-5555 | > Amtsgericht Hildesheim, HRA 2686 | http://www.pengutronix.de/ | > > _______________________________________________ > linux-arm-kernel mailing list > linux-arm-kernel(a)lists.infradead.org > http://lists.infradead.org/mailman/listinfo/linux-arm-kernel -- 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] ASoC: wm2000: Expose ANC gain adjustment
by Mark Brown 30 Jan '13

30 Jan '13

No TLV information since it's not actually a direct gain control. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 4d809ab..f2ac38b 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -666,6 +666,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, } static const struct snd_kcontrol_new wm2000_controls[] = { + SOC_SINGLE("ANC Volume", WM2000_REG_ANC_GAIN_CTRL, 0, 255, 0), SOC_SINGLE_BOOL_EXT("WM2000 ANC Switch", 0, wm2000_anc_mode_get, wm2000_anc_mode_put), @@ -778,6 +779,8 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); + /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix mono
by Benoît Thébaudeau 30 Jan '13

30 Jan '13

The TLV320AIC3x DAI supports only stereo. This patch fixes mono support for this codec by allowing only stereo at codec level. Tested with imx-ssi and TLV320AIC3104. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5708a97..4989143 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1210,13 +1210,13 @@ static struct snd_soc_dai_driver aic3x_dai = { .name = "tlv320aic3x-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, -- 1.7.10.4

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[alsa-devel] Thread safety of snd_pcm_open?
by Jérôme Forissier 29 Jan '13

29 Jan '13

Hello, I'm having problems using ALSA with LibVLC. The project page http://www.alsa-project.org/main/index.php/SMP_Design claims that the snd_*_open() calls are thread safe, yet it seems that when LibVLC calls them from several threads it can lead to errors like: ALSA lib conf.c:4571:(parse_args) Unknown parameter CARD ALSA lib conf.c:4571:(parse_args) Unknown parameter DEV ALSA lib conf.c:4704:(snd_config_expand) Parse arguments error: No such file or directory ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=NVidia,DEV=0 ALSA lib conf.c:4571:(parse_args) Unknown parameter CARD ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=PCH,DEV=0 ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=NVidia,DEV=0 ALSA lib conf.c:4571:(parse_args) Unknown parameter DEV (Please see http://mailman.videolan.org/pipermail/vlc-devel/2013-January/091603.html and also http://mailman.videolan.org/pipermail/vlc-devel/2013-January/091633.html for a reproducer on top of LibVLC). I'm not familiar with the internals of ALSA lib, nor with the internals of LibVLC. The VLC devs claim that they respect the rules regarding ALSA access, but I can't tell for sure... I'm currently running my reproducer with valgrind, but before I dig too deeply, could you please give some advice? I have noticed that pcm_direct.c calls getgrnam() which is not reentrant (but replacing it with getgrnam_r is not enough to fix my problem). Also there does not seem to be any locking around snd_pcm_open[_noupdate]. Are those functions supposed to be reentrant? Thanks for your help, -- Jerome

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix non-snoop page handling
by Takashi Iwai 29 Jan '13

29 Jan '13

For non-snoop mode, we fiddle with the page attributes of CORB/RIRB and the position buffer, but also the ring buffers. The problem is that the current code blindly assumes that the buffer is contiguous. However, the ring buffers may be SG-buffers, thus a wrong vmapped address is passed there, leading to Oops. This patch fixes the handling for SG-buffers. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=800701 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++++++-------------- 1 file changed, 26 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bf0a004..c78286f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -656,29 +656,43 @@ static char *driver_short_names[] = { #define get_azx_dev(substream) (substream->runtime->private_data) #ifdef CONFIG_X86 -static void __mark_pages_wc(struct azx *chip, void *addr, size_t size, bool on) +static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on) { + int pages; + if (azx_snoop(chip)) return; - if (addr && size) { - int pages = (size + PAGE_SIZE - 1) >> PAGE_SHIFT; + if (!dmab || !dmab->area || !dmab->bytes) + return; + +#ifdef CONFIG_SND_DMA_SGBUF + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) { + struct snd_sg_buf *sgbuf = dmab->private_data; if (on) - set_memory_wc((unsigned long)addr, pages); + set_pages_array_wc(sgbuf->page_table, sgbuf->pages); else - set_memory_wb((unsigned long)addr, pages); + set_pages_array_wb(sgbuf->page_table, sgbuf->pages); + return; } +#endif + + pages = (dmab->bytes + PAGE_SIZE - 1) >> PAGE_SHIFT; + if (on) + set_memory_wc((unsigned long)dmab->area, pages); + else + set_memory_wb((unsigned long)dmab->area, pages); } static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, bool on) { - __mark_pages_wc(chip, buf->area, buf->bytes, on); + __mark_pages_wc(chip, buf, on); } static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, - struct snd_pcm_runtime *runtime, bool on) + struct snd_pcm_substream *substream, bool on) { if (azx_dev->wc_marked != on) { - __mark_pages_wc(chip, runtime->dma_area, runtime->dma_bytes, on); + __mark_pages_wc(chip, snd_pcm_get_dma_buf(substream), on); azx_dev->wc_marked = on; } } @@ -689,7 +703,7 @@ static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, { } static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, - struct snd_pcm_runtime *runtime, bool on) + struct snd_pcm_substream *substream, bool on) { } #endif @@ -1968,11 +1982,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; - struct snd_pcm_runtime *runtime = substream->runtime; struct azx_dev *azx_dev = get_azx_dev(substream); int ret; - mark_runtime_wc(chip, azx_dev, runtime, false); + mark_runtime_wc(chip, azx_dev, substream, false); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; azx_dev->format_val = 0; @@ -1980,7 +1993,7 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, params_buffer_bytes(hw_params)); if (ret < 0) return ret; - mark_runtime_wc(chip, azx_dev, runtime, true); + mark_runtime_wc(chip, azx_dev, substream, true); return ret; } @@ -1989,7 +2002,6 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx_dev *azx_dev = get_azx_dev(substream); struct azx *chip = apcm->chip; - struct snd_pcm_runtime *runtime = substream->runtime; struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; /* reset BDL address */ @@ -2002,7 +2014,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) snd_hda_codec_cleanup(apcm->codec, hinfo, substream); - mark_runtime_wc(chip, azx_dev, runtime, false); + mark_runtime_wc(chip, azx_dev, substream, false); return snd_pcm_lib_free_pages(substream); } -- 1.8.1.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable LPIB delay count for Poulsbo / Oaktrail
by Takashi Iwai 29 Jan '13

29 Jan '13

Currently we use LPIB forcibly for both playback and capture for Poulsbo and Oaktrail devices, and this seems rather problematic. The recent fix for LPIB delay count seems working well with these devices, so let's enable it instead. Reported-by: Martin Weishart <martin.weishart(a)telosalliance.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0b6aeba..bf0a004 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3613,13 +3613,12 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, - /* SCH */ + /* Poulsbo */ { PCI_DEVICE(0x8086, 0x811b), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Oaktrail */ + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- 1.8.1.1

1 0

[alsa-devel] [PATCH 0/6] Misc ALSA changes, documentation, style
by Antonio Ospite 29 Jan '13

29 Jan '13

From: Antonio Ospite <ao2(a)amarulasolutions.com> Hi, here are some fixes for things I noticed while studying ALSA. I'd like some feedback especially on patch 1. Patches 2 and 3 are documentation fixes. Patches 4, 5 and 6, are basically style fixes. Feel free to discard patch 6 if you find it too invasive, it's just cosmetics but some maintainers don't like big style changes even if they are in the "right" direction. I don't know alsa devs' position about that. I am sending the whole series to alsa-devel and CCing topic specific mailing lists or developers on a per-patch basis. Thanks, Antonio Antonio Ospite (6): Force a cast to silence a warning from "sparse" Documentation/DocBook/writing-an-alsa-driver.tmpl: fix some thinkos Documentation/DocBook/writing-an-alsa-driver.tmpl: fix some typos sound/usb/caiaq/device.c: fix use of MODULE_SUPPORTED_DEVICES() sound/usb/card.c: cosmetics, remove a leading space sound/core/pcm_native.c: cosmetics, fix use of spaces Documentation/DocBook/writing-an-alsa-driver.tmpl | 11 +- include/sound/memalloc.h | 2 +- sound/core/pcm_native.c | 120 ++++++++++----------- sound/usb/caiaq/device.c | 8 +- sound/usb/card.c | 2 +- 5 files changed, 71 insertions(+), 72 deletions(-) -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

3 11

Re: [alsa-devel] [PATCH 1/4] DMA: PL330: Add xlate function
by Arnd Bergmann 29 Jan '13

29 Jan '13

On Friday 18 January 2013, Padmavathi Venna wrote: > +struct dma_chan *of_dma_pl330_xlate(struct of_phandle_args *dma_spec, > + struct of_dma *ofdma) > +{ > + int count = dma_spec->args_count; > + struct of_dma_filter_info *info = ofdma->of_dma_data; > + > + if (!info || !info->filter_fn) > + return NULL; > + > + if (count != 1) > + return NULL; > + > + return dma_request_channel(info->dma_cap, info->filter_fn, dma_spec); > +} > +EXPORT_SYMBOL_GPL(of_dma_pl330_xlate); It seems a little sad that we still have to use dma_request_channel() to implement this, when that function will go off searching all channels and pass them tino the filter, which then has to look for the device node and match it with each channel. We already know the controller and should just be able to get a channel for it, although I don't exactly know how that is done. Further, your function is almost identical to the of_dma_simple_xlate function. Can't you use that one instead? Arnd

2 2

Re: [alsa-devel] [PATCH 4/4] ASOC: wm8962: add some operation in wm8962_runtime_resume()
by Mark Brown 29 Jan '13

29 Jan '13

On Tue, Jan 29, 2013 at 03:50:10PM +0800, Gary Zhang wrote: > 1. in this function, after wm8962_reset() is called, WM8962_SYSCLK_ENA bit > is reset to 1. make sure it is off so we can safely write to registers if > the device is declocked. > 2. add bias level management Split separate changes into separate patches; obviously there's bias level management in the driver since otherwise it'd be totally non-functional. > @@ -3721,6 +3721,24 @@ static int wm8962_runtime_resume(struct device *dev) > > regcache_sync(wm8962->regmap); > > + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, > + WM8962_SYSCLK_ENA, 0); > + This seems like a bodge to work around some other problem in your system. The device should never be runtime suspended with an active clock, and this seems like it isn't going to work as most of the register cache restore will be done with SYSCLK enabled. We need to figure out what went wrong to allow the device to enter runtime suspend with SYSCLK on. > + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, > + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, > + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); > + > + /* Bias enable at 2*50k for ramp */ > + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, > + WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, > + WM8962_BIAS_ENA | 0x180); > + > + msleep(5); > + > + /* VMID back to 2x250k for standby */ > + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, > + WM8962_VMID_SEL_MASK, 0x100); We already have code to ramp VMID when exiting _OFF...

1 0

Re: [alsa-devel] [PATCH V6 07/10] ASoC: SMDK: WM8994: Add device tree support for machine file
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:06PM +0530, Padmavathi Venna wrote: > +Samsung SMDK audio complex This is just for SMDKs with WM8994. I'll apply but please send a followup patch to clarify this - it'll be a different binding for others like the older boards using WM8580 and WM9713. > + samsung,i2s-controller = <&i2s0>; > + samsung,audio-codec = <&wm8994>; I was thinking this should really point at the DAI but since it's specific to the SMDKs I don't think it particularly matters - they all use AIF1 anyway.

2 1

Re: [alsa-devel] [PATCH V6 04/10] spi: s3c64xx: Modify SPI driver to use generic DMA DT support
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:03PM +0530, Padmavathi Venna wrote: > This patch modifies the SPI driver to use generic dma dt bindings > support. This passes all the required arguments to dma dev request > functon which in turn calls the dma_request_slave_channel or dma__ > request_channel based on DT or non-DT respectively. This loooks OK and I'm actually applying SPI patches so I could apply it but I'm not sure I see the relevance of this patch to the rest of the series (which is about ASoC). Is there some reason why it's included in this patch series or can it be applied to the SPI tree? The SMDKs use I2C for their CODEC control rather than SPI. Also is there a binding document update or device tree file update to go with this? > + if (!sdd->pdev->dev.of_node) { > + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); > + if (!res) { > + dev_err(&pdev->dev, "Unable to get SPI tx dma " > + "resource\n"); I appreciate that this is cut'n'paste from the code you're refactoring but please don't split error messages over lines, it makes it hard to grep for them in the kernel source.

2 1

Re: [alsa-devel] [PATCH V6 03/10] ARM: SAMSUNG: Make dma request compatible to generic dma bindings.
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:02PM +0530, Padmavathi Venna wrote: > This patch make the dma dev request operation compatible for both > DT and non-DT cases. It takes the all the arguments required for > dma_request_slave_channel and dma_request_channel. If the driver > is initiated via DT or non-DT the corresponding call will be made. OK, so this is where the SPI change came from - the API has been changed incompatibly and SPI needs updating. For bisection this API update ought to be done in a single commit, otherwise there will be steps in the bisection where nothing will work. Anyway, I'll go ahead and apply this, the SPI and the ASoC ones but this is something to bear in mind for the future.

2 1

Re: [alsa-devel] [PATCH 3/4] ASOC: imx: add machine driver using wm8962 codec
by Mark Brown 29 Jan '13

29 Jan '13

On Tue, Jan 29, 2013 at 03:49:50PM +0800, Gary Zhang wrote: > +static struct imx_priv card_priv; Global data bad... > +static int imx_wm8962_startup(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_dai *codec_dai = rtd->codec_dai; > + struct imx_priv *priv = &card_priv; > + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); > + > + if (!codec_dai->active) > + clk_enable(data->codec_clk); Should be clk_prepare_enable() but it's not clear to me why you need this... Also the clock API does refcounting so there should be no need to check for active, you should get matching startups and shutdowns. This will also fail for analogue bypass paths, set_bias_level() would be a good place to cover those. > + int_port--; > + ext_port--; > + ret = imx_audmux_v2_configure_port(int_port, > + IMX_AUDMUX_V2_PTCR_SYN | > + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | > + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | > + IMX_AUDMUX_V2_PTCR_TFSDIR | > + IMX_AUDMUX_V2_PTCR_TCLKDIR, > + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); > + if (ret) { > + dev_err(&pdev->dev, "audmux internal port setup failed\n"); > + return ret; > + } > + imx_audmux_v2_configure_port(ext_port, > + IMX_AUDMUX_V2_PTCR_SYN, > + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); > + if (ret) { > + dev_err(&pdev->dev, "audmux external port setup failed\n"); > + return ret; > + } This stuff is pretty common for i.MX drivers - can we factor it out into a helper library? > + ssi_pdev = of_find_device_by_node(ssi_np); > + if (!ssi_pdev) { > + dev_err(&pdev->dev, "failed to find SSI platform device\n"); > + ret = -EINVAL; > + goto fail; > + } Please remember to include binding documentation for all new device tree bindings.

1 0

Re: [alsa-devel] [PATCH 3/4] ARM: dts: Add #dma-cells for generic dma binding support
by Arnd Bergmann 29 Jan '13

29 Jan '13

On Friday 18 January 2013, Padmavathi Venna wrote: > diff --git a/Documentation/devicetree/bindings/dma/arm-pl330.txt b/Documentation/devicetree/bindings/dma/arm-pl330.txt > index 36e27d5..457a233 100644 > --- a/Documentation/devicetree/bindings/dma/arm-pl330.txt > +++ b/Documentation/devicetree/bindings/dma/arm-pl330.txt > @@ -8,6 +8,8 @@ Required properties: > - reg: physical base address of the controller and length of memory mapped > region. > - interrupts: interrupt number to the cpu. > + - #dma-cells: must be at least 1. used to represent the number of integer > + cells in the dmas property of client device. The wording 'at least' seems wrong here: that is what we use in the generic DMA binding, but in the part that is specific to one driver, I would expect to see - #dma-cells: must be <1> Since that is what this particular driver requires. > Client drivers (device nodes requiring dma transfers from dev-to-mem or > @@ -27,7 +30,7 @@ as shown below. > [property name] = <[phandle of the dma controller] [dma request id]>; > > where 'dma request id' is the dma request number which is connected > - to the client controller. The 'property name' is recommended to be > - of the form <name>-dma-channel. > + to the client controller. The 'property name' is 'dmas' as recommended > + by the generic dma device tree binding helpers. s/recommended/required/ Also, the "dma-names" property is required as well. Arnd

2 1

Re: [alsa-devel] [PATCH V6 10/10] dmaengine: Fix compilation error in non-DT case
by Vinod Koul 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:09PM +0530, Padmavathi Venna wrote: > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > include/linux/dmaengine.h | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h > index 8cd0e25..c88f302 100644 > --- a/include/linux/dmaengine.h > +++ b/include/linux/dmaengine.h > @@ -992,7 +992,7 @@ static inline struct dma_chan *__dma_request_channel(dma_cap_mask_t *mask, > static inline struct dma_chan *dma_request_slave_channel(struct device *dev, > char *name) > { > - return NULL > + return NULL; what tree was this generated against? this was fixed by 678bd8 -- ~Vinod

2 1

Re: [alsa-devel] [PATCH V6 09/10] ASoC: Samsung: Update Kconfig for I2S, SPDIF and PCM audio
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:08PM +0530, Padmavathi Venna wrote: > Update Kconfig file to enable I2S,PCM audio for wm8994 and spdif on all > samsung platforms. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH V6 08/10] ARM: dts: Modify SPI nodes according generic DMA DT bindings
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:07PM +0530, Padmavathi Venna wrote: > This patch removes custom way of adding spi dma channels and > adds according to new generic DMA DT bindings. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/4] DMA: PL330: Register the DMA controller with the generic DMA helpers
by Arnd Bergmann 28 Jan '13

28 Jan '13

On Monday 28 January 2013, Jon Hunter wrote: > On 01/28/2013 08:51 AM, Arnd Bergmann wrote: > > On Friday 18 January 2013, Padmavathi Venna wrote: > >> + pfi = devm_kzalloc(&adev->dev, sizeof(*pfi), GFP_KERNEL); > >> + if (!pfi) { > >> + dev_err(&adev->dev, "unable to allocate mem\n"); > >> + return -ENOMEM; > >> + } > >> + > >> + pfi->dma_cap = pd->cap_mask; > >> + pfi->filter_fn = pl330_filter; > >> + > >> + ret = of_dma_controller_register(adev->dev.of_node, > >> + of_dma_pl330_xlate, pfi); > > > > Why do you pass a 'struct of_dma_filter_info' here? I would > > expect that you pass the pdmac object as the third argument. > > I believe it is because that is the data that the xlate function is > using. Are you suggesting the data should be stored in the pdmac object > and extracted from there? That could be done too given that this > controller has its own xlate. It just seems weird that we are passing a constant cap_mask in a data structure, and that we need our own filter function still, but don't pass the pointer to the data structure that we actually need in the filter function (the dma_device *). Arnd

1 0

Re: [alsa-devel] [PATCH 0/4] Add generic DMA DT binding support
by Vinod Koul 28 Jan '13

28 Jan '13

On Fri, Jan 18, 2013 at 05:03:40PM +0530, Padmavathi Venna wrote: > This patch set adds support for generic dma device tree bindings for > Samsung platforms and is dependent on the following patches from > Vinod Koul next branch > 1)of: Add generic device tree DMA helpers > 2)dmaengine: add helper function to request a slave DMA channel Changes look fairly decent. I need somone with better knowldge of DT to akc this before this is applied. Arnd...? > > This patch set is made based Kukjin Kim for-next branch > > Padmavathi Venna (4): > DMA: PL330: Add xlate function > DMA: PL330: Register the DMA controller with the generic DMA helpers > ARM: dts: Add #dma-cells for generic dma binding support > DMA: PL330: Modify pl330 filter based on new generic dma dt bindings. > > .../devicetree/bindings/dma/arm-pl330.txt | 9 +++- > arch/arm/boot/dts/exynos5250.dtsi | 4 ++ > drivers/dma/pl330.c | 48 ++++++++++++++++---- > 3 files changed, 49 insertions(+), 12 deletions(-) > > -- > 1.7.4.4 >

2 1

Re: [alsa-devel] [PATCH 2/4] DMA: PL330: Register the DMA controller with the generic DMA helpers
by Arnd Bergmann 28 Jan '13

28 Jan '13

On Friday 18 January 2013, Padmavathi Venna wrote: > + pfi = devm_kzalloc(&adev->dev, sizeof(*pfi), GFP_KERNEL); > + if (!pfi) { > + dev_err(&adev->dev, "unable to allocate mem\n"); > + return -ENOMEM; > + } > + > + pfi->dma_cap = pd->cap_mask; > + pfi->filter_fn = pl330_filter; > + > + ret = of_dma_controller_register(adev->dev.of_node, > + of_dma_pl330_xlate, pfi); Why do you pass a 'struct of_dma_filter_info' here? I would expect that you pass the pdmac object as the third argument. Arnd

1 0

[alsa-devel] Some information about UsbPre2 UAC2 support
by shal＠free.fr 28 Jan '13

28 Jan '13

Hi all, The Sound Devices USBPre2 is an UAC2 device but recognized as a UAC1 device by Linux. I have spend some time in order to found a solution. I have successfully setup my sound card as a UAC2 mode and I can play/record a 24/192khz without problem. But my solution is very ugly and cannot be considered as a real solution. Perhaps alsa developers have an idea to integrated UAC2 support for the USBPRe2 in the standard stack.... There is two distinct problems: - The sound card recognizes Linux as a Windows OS and send a incorrect device descriptor. - The USB stack choose the bad configuration descriptor. More details on first problem: For some compatibility raison, Sound device developer want only send a UAC1 configuration when the Operating system is Windows. The used method is based on the length of the setup packet during the device get_descriptor function. If the length is 64, the OS is considered as Windows. Linux have choosen also this value. Extract of kernel source code: " NOTE: If USE_NEW_SCHEME() is true we will start by issuing a 64-byte GET_DESCRIPTOR request. This is what Windows does, so it may help with some non-standards-compliant devices." So I have patched my kernel. In driver/usb/core/hub.c: #define GET_DESCRIPTOR_BUFSIZE 65 (instead 64) This solve the first problem. More details on the second problem: Now, the USB device descriptor has two configuration descriptors :the first is a UAC1 and the second is a UAC2. My "lsusb -v -d 0926:" can be view at: http://pastebin.com/nFTN2E04 As UAC1 is a valid configuration descriptor, the usb stack choose this configuration. I have patched the usb_choose_configuration() function in file driver/usb/core/generic.c for overwrite the value with the good one. That all, no other problem was detected. The Linux UAC2 stack seems to be OK with this sound card. I joined (ugly) patches , built on linux 3.7.1. If you have an idea about how have a standard support of UAC2 for the Sound Devices USBPRe2, I (and a lot of people) will be happy. Thanks Olivier PS: A thread exists on LAU : http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-January/089199.…

2 5

[alsa-devel] [PATCH] ALSA: hda - Fix powermap for external mics on IDT codecs
by David Henningsson 28 Jan '13

28 Jan '13

This patch fixes a regression of the external mic not working on HP Probook 4520s. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 8 ++++++++ 1 file changed, 8 insertions(+) This is why my external mic stopped working; I now had some time to track it down. Alsa-info at http://paste.ubuntu.com/1579993/ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1c49861..951fc0d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -502,6 +502,13 @@ static void stac_line_automute(struct hda_codec *codec, jack_update_power(codec, jack); } +static void stac_mic_autoswitch(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + snd_hda_gen_mic_autoswitch(codec, jack); + jack_update_power(codec, jack); +} + static void stac_vref_event(struct hda_codec *codec, struct hda_jack_tbl *event) { unsigned int data; @@ -3509,6 +3516,7 @@ static int stac_parse_auto_config(struct hda_codec *codec) spec->gen.automute_hook = stac_update_outputs; spec->gen.hp_automute_hook = stac_hp_automute; spec->gen.line_automute_hook = stac_line_automute; + spec->gen.mic_autoswitch_hook = stac_mic_autoswitch; err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ucm: Set uc_mgr->ctl to NULL after closing it.
by Tanu Kaskinen 28 Jan '13

28 Jan '13

Fixes a double-free bug. Signed-off-by: Tanu Kaskinen <tanuk(a)iki.fi> --- src/ucm/main.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/ucm/main.c b/src/ucm/main.c index 0d7d841..d8b8ada 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -145,6 +145,7 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, free(uc_mgr->ctl_dev); uc_mgr->ctl_dev = NULL; snd_ctl_close(uc_mgr->ctl); + uc_mgr->ctl = NULL; } err = snd_ctl_open(ctl, ctl_dev, 0); -- 1.7.10.4

3 2

Re: [alsa-devel] 3.7 HDMI channel map regression
by Takashi Iwai 28 Jan '13

28 Jan '13

At Sun, 27 Jan 2013 19:18:27 -0600, Shawn Bohrer wrote: > > Hi Takashi, > > I recently updated my HTPC from 3.6.11 to 3.7.2 and this caused my RL > and FC channels to swap, and my RR and LFE channels to swap for PCM > audio. Doing a git bisect identified > d45e6889ee69456a4d5b1bbb32252f460cd48fa9 "ALSA: hda - Provide the > proper channel mapping for generic HDMI driver" as the commit that > caused my channels to swap. The commit doesn't revert cleanly on > 3.7.4, and I haven't really looked to see what the correct fix might > be. > > Some info that may be relevant, the sound card is a: > > 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset > Family High Definition Audio Controller (rev 04) > > The machine is running Fedora 18 and audio goes over HDMI to a 5.1 > receiver. I'm not really sure what other info you might need, but > let me know if you need something else or have any patches you would > like me to test. OK, it's the first time to get a bug report about this. Could you tell me how did you test it (i.e. which application, which sound backend)? Can you confirm that it's reproduced via speaker-test program in alsa-utils package? For further debugging, please give the following: - alsa-info.sh output while playing 5.1 sound - /proc/asound/card*/eld* contents while playing 5.1 sound Also, try to update alsa-lib and alsa-utils from alsa-project.org git tree, and build alsa-lib/test/chmap program. % cd alsa-lib/test % make chmap and run like ./chmap -Dhdmi:1 query (-Dhdmi:1 might be different depending on the system setup) thanks, Takashi

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[alsa-devel] [PATCH] ALSA: hda - fix inverted internal mic on Acer AOA150/ZG5
by David Henningsson 28 Jan '13

28 Jan '13

This patch enables internal mic input on the machine. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1107477 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc75607..de512c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2363,6 +2363,7 @@ static const struct hda_model_fixup alc268_fixup_models[] = { }; static const struct snd_pci_quirk alc268_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x015b, "Acer AOA 150 (ZG5)", ALC268_FIXUP_INV_DMIC), /* below is codec SSID since multiple Toshiba laptops have the * same PCI SSID 1179:ff00 */ -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 3/3] ALSA: intel8x0: Check return value before assigning
by Takashi Iwai 28 Jan '13

28 Jan '13

At Sun, 27 Jan 2013 17:40:02 +0100, Emil Goode wrote: > > The first patch in this series changes the return type of function > kvm_para_available to bool. In the function snd_intel8x0_inside_vm > we now need to check it's return value before assigning a value to > the result variable. > > Signed-off-by: Emil Goode <emilgoode(a)gmail.com> > --- > sound/pci/intel8x0.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c > index 3b9be75..fa9337e 100644 > --- a/sound/pci/intel8x0.c > +++ b/sound/pci/intel8x0.c > @@ -2982,7 +2982,7 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) > } > > /* detect KVM and Parallels virtual environments */ > - result = kvm_para_available(); > + result = kvm_para_available() ? 1 : 0; This is superfluous. The conversion from bool to int works implicitly like that. Takashi

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[alsa-devel] [PATCH 1/2] Revert "ASoC: fsl: fix multiple definition of init_module"
by Shawn Guo 28 Jan '13

28 Jan '13

This reverts commit 25b8d31488a3fb3611651991969526b2ea475764. While the commit fixes multiple init_module definition error with module build, it breaks build when both imx-pcm-fiq and imx-pcm-dma are built in as below. LD sound/soc/fsl/snd-soc-fsl-ssi.o LD sound/soc/fsl/snd-soc-fsl-utils.o LD sound/soc/fsl/snd-soc-imx-ssi.o LD sound/soc/fsl/snd-soc-imx-audmux.o LD sound/soc/fsl/snd-soc-imx-pcm-fiq.o LD sound/soc/fsl/snd-soc-imx-pcm-dma.o LD sound/soc/fsl/snd-soc-eukrea-tlv320.o LD sound/soc/fsl/snd-soc-imx-sgtl5000.o LD sound/soc/fsl/snd-soc-imx-mc13783.o LD sound/soc/fsl/built-in.o sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_free': imx-pcm.c:(.text+0x464): multiple definition of `imx_pcm_free' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x1a8): first defined here sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `snd_imx_pcm_mmap': imx-pcm.c:(.text+0x35c): multiple definition of `snd_imx_pcm_mmap' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0xa0): first defined here sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_new': imx-pcm.c:(.text+0x3dc): multiple definition of `imx_pcm_new' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x120): first defined here make[4]: *** [sound/soc/fsl/built-in.o] Error 1 Let's revert the commit and find a proper fix for multiple init_module definition error later. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/Kconfig | 9 +++++++-- sound/soc/fsl/Makefile | 5 +---- sound/soc/fsl/imx-pcm.c | 3 +++ 3 files changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a210c8d..3b98159 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,13 +108,18 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM_FIQ +config SND_SOC_IMX_PCM tristate + +config SND_SOC_IMX_PCM_FIQ + bool select FIQ + select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - tristate + bool select SND_SOC_DMAENGINE_PCM + select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index ec14579..afd3479 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -41,10 +41,7 @@ endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o -snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o -obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o -snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index 0c9f188..d5cd9ef 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -31,6 +31,7 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); return ret; } +EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -79,6 +80,7 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } +EXPORT_SYMBOL_GPL(imx_pcm_new); void imx_pcm_free(struct snd_pcm *pcm) { @@ -100,6 +102,7 @@ void imx_pcm_free(struct snd_pcm *pcm) buf->area = NULL; } } +EXPORT_SYMBOL_GPL(imx_pcm_free); MODULE_DESCRIPTION("Freescale i.MX PCM driver"); MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); -- 1.7.9.5

2 2

[alsa-devel] [PATCH - TINYCOMPRESS 1/4] compress: Extend limit on error messages
by Charles Keepax 28 Jan '13

28 Jan '13

32 characters is a very short size for the error message, extend this to 128 characters to prevent clipping on the majority of messages. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> diff --git a/compress.c b/compress.c index faef982..05e54ae 100644 --- a/compress.c +++ b/compress.c @@ -75,7 +75,7 @@ #include "sound/compress_offload.h" #include "tinycompress/tinycompress.h" -#define COMPR_ERR_MAX 32 +#define COMPR_ERR_MAX 128 struct compress { int fd; -- 1.7.2.5

2 4

Re: [alsa-devel] [PATCH 3/3] ALSA: intel8x0: Check return value before assigning
by H. Peter Anvin 27 Jan '13

27 Jan '13

Wtf? This patch seems semibogus at best and the description is even more ridiculous... Emil Goode <emilgoode(a)gmail.com> wrote: >The first patch in this series changes the return type of function >kvm_para_available to bool. In the function snd_intel8x0_inside_vm >we now need to check it's return value before assigning a value to >the result variable. > >Signed-off-by: Emil Goode <emilgoode(a)gmail.com> >--- > sound/pci/intel8x0.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > >diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c >index 3b9be75..fa9337e 100644 >--- a/sound/pci/intel8x0.c >+++ b/sound/pci/intel8x0.c >@@ -2982,7 +2982,7 @@ static int snd_intel8x0_inside_vm(struct pci_dev >*pci) > } > > /* detect KVM and Parallels virtual environments */ >- result = kvm_para_available(); >+ result = kvm_para_available() ? 1 : 0; > #ifdef X86_FEATURE_HYPERVISOR > result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); > #endif -- Sent from my mobile phone. Please excuse brevity and lack of formatting.

1 0

[alsa-devel] ASoC updates for v3.8
by Mark Brown 27 Jan '13

27 Jan '13

The following changes since commit 7d1f9aeff1ee4a20b1aeb377dd0f579fe9647619: Linux 3.8-rc4 (2013-01-17 19:25:45 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.8-rc4 for you to fetch changes up to a8c136d2eba7b0b0c872e59eb404b1fe36504112: Merge remote-tracking branch 'asoc/fix/wm2200' into tmp (2013-01-22 16:26:33 +0800) ---------------------------------------------------------------- ASoC: Updates for v3.8-rc4 The usual set of driver updates, nothing too thrilling in here - one core change for the regulator bypass mode which was just not doing the right thing at all and a bunch of driver specifics. ---------------------------------------------------------------- Charles Keepax (2): ASoC: arizona: Disable free-running mode on FLL1 ASoC: wm_adsp: Release firmware on error Chris Rattray (1): ASoC: wm2200: correct mixer values and text Liam Girdwood (1): ASoC: MAINTAINERS: Update email address. Mark Brown (10): ASoC: dapm: Fix sense of regulator bypass mode ASoC: wm5102: Correct AEC loopback mask ASoC: wm5110: Correct AEC loopback mask ASoC: arizona: Use actual rather than desired BCLK when calculating LRCLK ASoC: wm_adsp: Use GFP_DMA for things that may be DMAed Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/fsl' into tmp Merge remote-tracking branch 'asoc/fix/lrg' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Shawn Guo (1): ASoC: fsl: fix multiple definition of init_module MAINTAINERS | 2 +- sound/soc/codecs/arizona.c | 5 ++++- sound/soc/codecs/wm2200.c | 3 --- sound/soc/codecs/wm5102.c | 3 +-- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm_adsp.c | 6 +++--- sound/soc/fsl/Kconfig | 9 ++------- sound/soc/fsl/Makefile | 5 ++++- sound/soc/fsl/imx-pcm.c | 3 --- sound/soc/soc-dapm.c | 12 ++++++++++-- 10 files changed, 26 insertions(+), 25 deletions(-)

2 1

[alsa-devel] [git pull] usb-audio: RME Babyface/Fireface fix
by Clemens Ladisch 27 Jan '13

27 Jan '13

The following change since commit 29594404d7fe73cd80eaa4ee8c43dcc53970c60e: Linux 3.7 (2012-12-10 19:30:57 -0800) is available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git usb-audio-fix for you to fetch changes up to 5fcbad428a81efa318f15bfcfbcd8f0a98ec0b11: ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices (2013-01-26 17:49:35 +0100) ---------------------------------------------------------------- Clemens Ladisch (1): ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices sound/usb/mixer.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices Commit 23caaf19b11e (ALSA: usb-mixer: Add support for Audio Class v2.0) forgot to adjust the length check for UAC 2.0 feature unit descriptors. This would make the code abort on encountering a feature unit without per-channel controls, and thus prevented the driver to work with any device having such a unit, such as the RME Babyface or Fireface UCX. Reported-by: Florian Hanisch <fhanisch(a)uni-potsdam.de> Tested-by: Matthew Robbetts <wingfeathera(a)gmail.com> Tested-by: Michael Beer <beerml(a)sigma6audio.de> Cc: Daniel Mack <daniel(a)caiaq.de> Cc: 2.6.35+ <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 298070e..41e8bfb 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1259,16 +1259,23 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } channels = (hdr->bLength - 7) / csize - 1; bmaControls = hdr->bmaControls; + if (hdr->bLength < 7 + csize) { + snd_printk(KERN_ERR "usbaudio: unit %u: " + "invalid UAC_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } } else { struct uac2_feature_unit_descriptor *ftr = _ftr; csize = 4; channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; - } - - if (hdr->bLength < 7 || !csize || hdr->bLength < 7 + csize) { - snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); - return -EINVAL; + if (hdr->bLength < 6 + csize) { + snd_printk(KERN_ERR "usbaudio: unit %u: " + "invalid UAC_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } } /* parse the source unit */

2 1

Re: [alsa-devel] [PATCH v2] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Mark Brown 27 Jan '13

27 Jan '13

On Fri, Jan 18, 2013 at 10:17:00AM +0100, Michal Bachraty wrote: > u32 rotate = (32 - word_length) / 4; > This implementation is wrong, but it works only for 16, or 32 bit audio data. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH V6 10/10] dmaengine: Fix compilation error in non-DT case
by Mark Brown 27 Jan '13

27 Jan '13

On Fri, Jan 18, 2013 at 05:17:09PM +0530, Padmavathi Venna wrote: > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> This should be sent separately to Vinod for merge into v3.8, it's not obviously related to the rest of the series and is a bug fix.

1 0

[alsa-devel] upsampling
by Ran Shalit 26 Jan '13

26 Jan '13

Hello, I would like to ask if I can use alsa-lib to resample (downsample or sample) input audio and playback audio. I searched for appropriate API but found none. Regards, Ran

2 1

[alsa-devel] [patch] ASoC: dwc: fix support for more than two channels
by Dan Carpenter 26 Jan '13

26 Jan '13

There were missing break statements so everything used TWO_CHANNEL_SUPPORT. Also I added a return statement to silence a GCC warning: sound/soc/dwc/designware_i2s.c: In function ‘dw_i2s_hw_params’: sound/soc/dwc/designware_i2s.c:236:32: warning: ‘ch_reg’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Static analysis. Untested. diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 1aa5130..deb30d5 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -210,15 +210,19 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, switch (config->chan_nr) { case EIGHT_CHANNEL_SUPPORT: ch_reg = 3; + break; case SIX_CHANNEL_SUPPORT: ch_reg = 2; + break; case FOUR_CHANNEL_SUPPORT: ch_reg = 1; + break; case TWO_CHANNEL_SUPPORT: ch_reg = 0; break; default: dev_err(dev->dev, "channel not supported\n"); + return -EINVAL; } i2s_disable_channels(dev, substream->stream);

3 2

Re: [alsa-devel] [PATCH] Asoc: Samsung: Add support for I2S runtime S2R
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 06:05:30PM +0530, R. Chandrasekar wrote: > From: "R. Chandrasekar" <rcsekar(a)samsung.com> > > This patch adds runtime suspend to resume support for I2S. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > Signed-off-by: R. Chandrasekar <rcsekar(a)samsung.com> This is good but doesn't apply against topic/samsung, please check and resend. Note that the signoffs should be in author order - the author should be first, then people forwarding on the patch. Other people should have other tags like Acked-by or Reviewed-by.

1 0

Re: [alsa-devel] [PATCH] Asoc: Samsung: Add I2S S/W RST in startup function
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 06:05:31PM +0530, R. Chandrasekar wrote: > From: Padmavathi Venna <padma.v(a)samsung.com> > > I2S module need to be reset after S2R. Keeping the S/W rst > control part in resume didn't help in playing audio after resume. > So this patch adds S/W RST control part in startup function which > gets triggered for every new audio stream playback. Applied, thanks. Please use subject lines matching the subsystem.

1 0

Re: [alsa-devel] [PATCH 002/002] ASoC: Replace max98090 Device Driver
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 03:28:54PM -0800, Jerry Wong wrote: > This is an odd control that is a combination of an enum and a power > management widget. The 0 value of the enum turns power off. > Not sure what is another choice. The enum value needs to be > saved so it can be restored when DAPM turns the control on. That doesn't appear to be what your control actually does - what happens if someone writes ot the control while the widget is powered down? > > +static const struct snd_kcontrol_new max98090_snd_controls[] = { > > + SOC_SINGLE("MIC Bias VCM Bandgap", M98090_REG_BIAS_CNTL, > > + M98090_VCM_MODE_SHIFT, M98090_VCM_MODE_NUM - 1, 0), > > Why is this a user visible control? > It is not controlled by DAPM. The low power function is taken care of by > /SHDN\, so this retains the user choice of bandgap or divider reference. Why would the user want this choice? > > +static int max98090_runtime_resume(struct device *dev) > > +{ > > + struct max98090_priv *max98090 = dev_get_drvdata(dev); > > + > > + regcache_cache_only(max98090->regmap, false); > > + > > + max98090_reset(max98090); > > + > > + regcache_sync(max98090->regmap); > > Do you really need to reset the device here? If anything I'd expect it > > on suspend since the reset state is probably the lowest power state. > Yes, for a soft reset where the device is not power cycled, DAPM has > the device go in and out of the low power /SHDN\ state, and the old > register values are retained. Without this, we would need to specify > that the PMIC power cycle the chip. This doesn't really answer my questions. Why would it matter if the PMIC has power cycled the chip, and if you do need to do this why is resume the time to do it?

2 4

[alsa-devel] [PATCH 04/19] oss/dmabuf: use dma_map_single
by Arnd Bergmann 25 Jan '13

25 Jan '13

The virt_to_bus/bus_to_virt functions have been deprecated for as long as I can remember, and they are used in very few remaining instances, usually in obscure ISA device drivers. The OSS sound drivers are the only ones that are still used on the ARM architecture, and only on some of the earliest StrongARM machines. The problem for converting the OSS subsystem to use dma_map_single instead is that the caller of virt_to_bus does not have a device pointer, since the subsystem has never been ported to use the common device infrastructure. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmabuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index bcc3e8e..a59c888 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap) } } dmap->raw_buf = start_addr; - dmap->raw_buf_phys = virt_to_bus(start_addr); + dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL); for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) SetPageReserved(page); @@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap) for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) ClearPageReserved(page); + dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL); free_pages((unsigned long) dmap->raw_buf, sz); dmap->raw_buf = NULL; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: wm_adsp: Accept 0 as a parameter block address
by Mark Brown 25 Jan '13

25 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 85 ++++++++++++++++++-------------------------- 1 file changed, 35 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bc5e383..3ba4237 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -589,27 +589,21 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].dm), be32_to_cpu(adsp1_alg[i].zm)); - if (adsp1_alg[i].dm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_DM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].dm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].dm); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp1_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_ZM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].zm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].zm); + list_add_tail(&region->list, &dsp->alg_regions); break; case WMFW_ADSP2: @@ -623,38 +617,29 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].ym), be32_to_cpu(adsp2_alg[i].zm)); - if (adsp2_alg[i].xm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_XM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].xm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].xm); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp2_alg[i].ym) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_YM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].ym); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].ym); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp2_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_ZM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].zm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].zm); + list_add_tail(&region->list, &dsp->alg_regions); break; } } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 04/19] oss/dmabuf: use dma_map_single
by Arnd Bergmann 25 Jan '13

25 Jan '13

The virt_to_bus/bus_to_virt functions have been deprecated for as long as I can remember, and they are used in very few remaining instances, usually in obscure ISA device drivers. The OSS sound drivers are the only ones that are still used on the ARM architecture, and only on some of the earliest StrongARM machines. The problem for converting the OSS subsystem to use dma_map_single instead is that the caller of virt_to_bus does not have a device pointer, since the subsystem has never been ported to use the common device infrastructure. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmabuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index bcc3e8e..a59c888 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap) } } dmap->raw_buf = start_addr; - dmap->raw_buf_phys = virt_to_bus(start_addr); + dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL); for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) SetPageReserved(page); @@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap) for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) ClearPageReserved(page); + dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL); free_pages((unsigned long) dmap->raw_buf, sz); dmap->raw_buf = NULL; } -- 1.8.0

1 0

[alsa-devel] [PATCH] ASoC: arizona: Always keep SYSCLK link for DAIs
by Mark Brown 25 Jan '13

25 Jan '13

Don't disconnect SYSCLK when a DAI moves to the ASYNCCLK domain, we always need SYSCLK for operation even if it's not the reference clock for the domain. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 27b6efc..f76e982 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -978,13 +978,27 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, routes[0].sink = dai->driver->capture.stream_name; routes[1].sink = dai->driver->playback.stream_name; - routes[0].source = arizona_dai_clk_str(dai_priv->clk); - routes[1].source = arizona_dai_clk_str(dai_priv->clk); - snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + routes[0].source = arizona_dai_clk_str(dai_priv->clk); + routes[1].source = arizona_dai_clk_str(dai_priv->clk); + snd_soc_dapm_del_routes(&codec->dapm, routes, + ARRAY_SIZE(routes)); + break; + default: + break; + } - routes[0].source = arizona_dai_clk_str(clk_id); - routes[1].source = arizona_dai_clk_str(clk_id); - snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + switch (clk_id) { + case ARIZONA_CLK_ASYNCCLK: + routes[0].source = arizona_dai_clk_str(clk_id); + routes[1].source = arizona_dai_clk_str(clk_id); + snd_soc_dapm_add_routes(&codec->dapm, routes, + ARRAY_SIZE(routes)); + break; + default: + break; + } dai_priv->clk = clk_id; -- 1.7.10.4

1 0

Re: [alsa-devel] [PATCH next-20130124] Sound: pci: Fix unused variable warning in patch_sigmatel.c
by Takashi Iwai 25 Jan '13

25 Jan '13

At Fri, 25 Jan 2013 00:38:06 +0200, Stratos Karafotis wrote: > > Fix the following build warnings > > sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’: > sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable] Thanks for catching. But such temporal cast like your patch makes the code readability much worse. The code readability (thus its maintainability) is a more important factor. I'm going to fix the definition of snd_printd() itself for fixing this kind of warnings. We can use inline functions instead of empty macros to achieve the same but without triggering compiler warnings. Takashi > > Signed-off-by: Stratos Karafotis <stratosk(a)semaphore.gr> > --- > sound/pci/hda/patch_sigmatel.c | 14 +++++--------- > 1 file changed, 5 insertions(+), 9 deletions(-) > > diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c > index 0aa0ceb..f269d1f 100644 > --- a/sound/pci/hda/patch_sigmatel.c > +++ b/sound/pci/hda/patch_sigmatel.c > @@ -2365,7 +2365,6 @@ static void stac92hd71bxx_fixup_ref(struct hda_codec *codec, > static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > struct hda_jack_tbl *jack; > > if (action != HDA_FIXUP_ACT_PRE_PROBE) > @@ -2381,7 +2380,7 @@ static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, > if (jack) > jack->private_data = 0x02; > > - spec->gpio_mask |= 0x02; > + ((struct sigmatel_spec *)codec->spec)->gpio_mask |= 0x02; > > /* enable internal microphone */ > snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); > @@ -2420,19 +2419,15 @@ static void stac92hd71bxx_fixup_hp_dv5(struct hda_codec *codec, > static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > - > if (action != HDA_FIXUP_ACT_PRE_PROBE) > return; > - spec->gpio_led = 0x08; > + ((struct sigmatel_spec *)codec->spec)->gpio_led = 0x08; > } > > > static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > - > if (action != HDA_FIXUP_ACT_PRE_PROBE) > return; > > @@ -2456,8 +2451,9 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, > > if (find_mute_led_cfg(codec, 1)) > snd_printd("mute LED gpio %d polarity %d\n", > - spec->gpio_led, > - spec->gpio_led_polarity); > + ((struct sigmatel_spec *)codec->spec)->gpio_led, > + ((struct sigmatel_spec *)codec->spec)-> > + gpio_led_polarity); > > } > > -- > 1.8.1 >

2 2

[alsa-devel] ASoC: codec jz4740: Failed to build for use of "‘devm_ioremap_resource" on branch "for-next"
by Lin, Mengdong 25 Jan '13

25 Jan '13

Hi, I failed to build the branch “for-next”. Here is error information: sound/soc/codecs/jz4740.c: In function ‘jz4740_codec_probe’: sound/soc/codecs/jz4740.c:364:2: error: implicit declaration of function ‘devm_ioremap_resource’ [-Werror=implicit-function-declaration] sound/soc/codecs/jz4740.c:364:7: warning: assignment makes pointer from integer without a cast [enabled by default] It seems the newly introduced “devm_ioremap_resource” is to replace devm_request_and_ioremap(). But its definition are absent from the kernel. I cannot find where it’s defined, neither in device.h nor in io.h . Thanks Mengdong

1 0

[alsa-devel] hdsp rpm support
by Roman Haefeli 25 Jan '13

25 Jan '13

Hi everyone I would like to know current state of the RME HDSP RPM support. There was a thread [1] in 2008 on this list about the issue and things looked promising back then. However, I'm still unable to upload the firmware with alsa-driver from git on my Ubuntu 11.04 (32bit) machine with a 2.6.38 kernel. I posted a detailed description of the issue on the alsa-user list here [2]. Is it supposed to work with Cardbus? If not, is there something I can do to help support it? Any hints appreciated. Roman [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-May/008128.html [2] http://article.gmane.org/gmane.linux.alsa.user/36022

5 12

[alsa-devel] [PATCH 00/13] A few more fixes / enhancements for HD-audio
by Takashi Iwai 24 Jan '13

24 Jan '13

Not surprisingly, there are yet a few more patches for HD-audio. All patches are found in for-next branch of sound git tree as usual. Takashi

1 13

Re: [alsa-devel] spdif input, source synchronous?
by Rafal Powierski 24 Jan '13

24 Jan '13

Dave, it took me some time, but I finally got a proper equipment to test ALC886 SPDIF input. (hw:1,1,0) The test results show that the input is not "source synchronous" experiment: Analog->digital (spdif,toslink) Audio converter box, where internal crystal is removed and replaced with precision adjustable PLL-based gen. 2 simple independent sinus audio generators, providing audio source for ADC L,R inputs software: stream consequent samples (L,R) are compared. frequency of identical consequent samples (32bits) is calculated (ppm) results: crystal marking is: 12288000 = 256*48kHz gen. Hz | gen. ppm | duplicate probability (ppm) 12288000 0 0 12287856 -11.72 2 12287840 -13.02 4 12287824 -14.32 5 12287808 -15.62 7 12287792 -16.93 9 12287776 -18.22 10 12287760 -19.53 13 12287744 -20.83 14 12287728 -22.13 16 12287712 -23.44 17 12287696 -24.74 18 12287680 -26.04 20 12287664 -27.34 21 It seams that my HDA clock is a bit slower then definition ( about -10ppm) Therefore resampling starts below -10ppm It probably means that when spdif signal is 0ppm, about 10ppm of samples are missing. Is this a bug? or it is planed like this? Is this a software problem or hardware problem (codec or DMA sequencer) or configuration problem? since it is not a real resampling (no signal reconstruction takes place) samples are just repeated or skipped, it seams not right to me. Who could comment on this? Regards, Rafal -----Original Message----- From: Dave Platt Sent: Monday, November 12, 2012 11:40 PM To: powierski(a)mediascore.de Subject: re: [alsa-devel] spdif input, source synchronous? > Does anyone know how the SPDIF input is working on HDA? > Is it using a source-synchronous data transfer over HDA? > In my arrangement (ALC886) I can set up any rate and I am receiving a > rectangular “resampling”. > – is this a bug? > will this stay? so I can use higher ratio and some kind of software PLL to > recover input rate? I tested out S/PDIF input on my ALC262 setup some time ago. What I found, at the time, was that the data capture did appear to be driven by the rate of the incoming signal. My outboard ADC provides a 44100 sample/second rate, and that seemed to be what I was getting. I tried forcing the capture to 48000 samples per second, and saw no difference in the captured file size... the number of samples per second actually captured seemed to be the same (within experimental error) and I didn't see any evidence of resampling going on. Now, this was a capture with "hw:Intel,1" as the device. I expect that I would have seen resampling, and a different type of delivery if I had specified "plughw:Intel,1" or some other input style which had gone through the plug interface. Unfortunately I don't have any way of generating a CD-rate data stream with a specific set of known values... I'd need a CD player with an S/PDIF output and I don't have one of those handy. If I can figure out a way of doing this I'll try generating a set of CDs with a known-value pattern, play it, capture it, and see if it does or does not exactly correlate.

2 1

[alsa-devel] Soc MC13783: Improve driver to handle LineIn
by Gaëtan Carlier 24 Jan '13

24 Jan '13

Hello, I would like to improve MC13783 driver to be able to control bypass of LineIn (RXINR/L) and feed Analog LineIn stream into ouput Mixer via ADDRXIN flag. This first step one is for test purpose but I also need to select RXINR/L as capture source (I saw in RM that CODEC must be enabled and only RXINR flag must be selected as input) but nothing seems to be captured this is why I need bypass working to test input level, and so on. I try to create a Mixer based on other driver but it doesn't work. I try to finc information about widget macro SND_SOC_DAPM_* but it is hard to find something relevant. * Which widgets can be link together ? * Is there functional difference between VIRT and non-VIRT widget ? I will post my code in some minutes after clean-up. Thanks a lot. Gaëtan Carlier.

1 1

[alsa-devel] [PATCH Resend 0/4] ASoC: tegra: Some fixes in codec files
by Sachin Kamat 24 Jan '13

24 Jan '13

Resending this series with Stephen Warren's 'Reviewed-by' tag and dropping a patch as suggested by him. Compile tested using tegra_defconfig on the for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Sachin Kamat (4): ASoC: tegra_wm9712: Remove __devinitconst attribute ASoC: tegra20_ac97: Remove __devinitconst attribute ASoC: tegra: Use NULL instead of 0 for pointers ASoC: tegra: Staticize some functions in tegra30_i2s.c sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) -- 1.7.4.1

2 5

[alsa-devel] [PATCH 1/3] ASoC: soc-compress: Add missing brackets around else
by Charles Keepax 24 Jan '13

24 Jan '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 5fbfb06..80040f0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -112,10 +112,11 @@ static int soc_compr_free(struct snd_compr_stream *cstream) snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); - } else + } else { rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); + } } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, -- 1.7.2.5

3 4

[alsa-devel] [PATCH V2 0/2] ARM/ASoC: Davinci: Device Tree Update
by Hebbar Gururaja 24 Jan '13

24 Jan '13

1. Add pinctrl for McASP Module 2. Add DT support for Davinci machine platform This patch-set is tested on Davinci platform (DA850 EVM). This series applies on top of tag next-20130103 git tree https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git These patches cannot be tested directly on mainline kernel (DT based). This is because dma is not yet completely ported to Davinci ASoC. I had to add few hacks to get Audio working on Davinci DA850 platform. For reference, I have placed the entire working tree @ Tree : https://gitorious.org/davinci-asoc-dt/davinci-asoc-dt Branch : dev_next-20130103_davinci_asoc_dt Changes in V2 - Remove reference to Linux & software details from DT binding Hebbar, Gururaja (2): ASoC: davinci-mcasp: Add pinctrl support ASoC: Davinci: machine: Add device tree binding .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ sound/soc/davinci/davinci-evm.c | 179 +++++++++++++++++--- sound/soc/davinci/davinci-mcasp.c | 7 + 3 files changed, 219 insertions(+), 20 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

3 13

Re: [alsa-devel] [PATCH V6 01/10] ASoC: Samsung: Rename samsung i2s secondary device name
by Mark Brown 24 Jan '13

24 Jan '13

On Fri, Jan 18, 2013 at 05:17:00PM +0530, Padmavathi Venna wrote: > All Samsung SoCs has max 3 i2s controllers. So the i2s secondary fifo > interface device id was named as samsung-i2s.4. Renaming this to > "samsung-i2s-sec" to support device tree in i2s driver. Applied, thanks.

1 0

[alsa-devel] [PATCH 002/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 23 Jan '13

23 Jan '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch completes the replacement of the existing max98090 driver, by installing a more complete driver. Signed-off-by: Jerry Wong <jerry.wong(a)maximintegrated.com> Tested-by: Matthew Mowdy <matthew.mowdy(a)maximintegrated.com> Reviewed-by: Ralph Birt <ralph.birt(a)maximintegrated.com> --- I have updated most per the review comments, except... "External MIC Mux" is a mux with 0=power down, and exists in DAPM as well "Digital Sidetone Source", with a master SHDN has no power impact, so was left out of DAPM "Quick Setup System Clock" was just a question what they do... they override the clock settings for common use cases. "VCM bandgap reference", with a master SHDN has no power impact, so was left out of DAPM include/sound/max98090.h | 29 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98090.c | 2603 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98090.h | 1537 +++++++++++++++++++++++++ 5 files changed, 4175 insertions(+), 0 deletions(-) create mode 100755 include/sound/max98090.h create mode 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h diff --git a/include/sound/max98090.h b/include/sound/max98090.h new file mode 100755 index 0000000..95efb13 --- /dev/null +++ b/include/sound/max98090.h @@ -0,0 +1,29 @@ +/* + * Platform data for MAX98090 + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SOUND_MAX98090_PDATA_H__ +#define __SOUND_MAX98090_PDATA_H__ + +/* codec platform data */ +struct max98090_pdata { + + /* Analog/digital microphone configuration: + * 0 = analog microphone input (normal setting) + * 1 = digital microphone input + */ + unsigned int digmic_left_mode:1; + unsigned int digmic_right_mode:1; + unsigned int digmic_3_mode:1; + unsigned int digmic_4_mode:1; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0e368d4..3a84782 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -267,6 +268,9 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate +config SND_SOC_MAX98090 + tristate + config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aa56312..f6e8e36 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o +snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -157,6 +158,7 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c new file mode 100755 index 0000000..b796c55 --- /dev/null +++ b/sound/soc/codecs/max98090.c @@ -0,0 +1,2603 @@ +/* + * max98090.c -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/max98090.h> +#include "max98090.h" + +#include <linux/version.h> + +#define DEBUG +#define EXTMIC_METHOD +#define EXTMIC_METHOD_TEST + +/* Allows for sparsely populated register maps */ +static struct reg_default max98090_reg[] = { + { 0x00, 0x00 }, /* 00 Software Reset */ + { 0x03, 0x04 }, /* 03 Interrupt Masks */ + { 0x04, 0x00 }, /* 04 System Clock Quick */ + { 0x05, 0x00 }, /* 05 Sample Rate Quick */ + { 0x06, 0x00 }, /* 06 DAI Interface Quick */ + { 0x07, 0x00 }, /* 07 DAC Path Quick */ + { 0x08, 0x00 }, /* 08 Mic/Direct to ADC Quick */ + { 0x09, 0x00 }, /* 09 Line to ADC Quick */ + { 0x0A, 0x00 }, /* 0A Analog Mic Loop Quick */ + { 0x0B, 0x00 }, /* 0B Analog Line Loop Quick */ + { 0x0C, 0x00 }, /* 0C Reserved */ + { 0x0D, 0x00 }, /* 0D Input Config */ + { 0x0E, 0x1B }, /* 0E Line Input Level */ + { 0x0F, 0x00 }, /* 0F Line Config */ + + { 0x10, 0x14 }, /* 10 Mic1 Input Level */ + { 0x11, 0x14 }, /* 11 Mic2 Input Level */ + { 0x12, 0x00 }, /* 12 Mic Bias Voltage */ + { 0x13, 0x00 }, /* 13 Digital Mic Config */ + { 0x14, 0x00 }, /* 14 Digital Mic Mode */ + { 0x15, 0x00 }, /* 15 Left ADC Mixer */ + { 0x16, 0x00 }, /* 16 Right ADC Mixer */ + { 0x17, 0x03 }, /* 17 Left ADC Level */ + { 0x18, 0x03 }, /* 18 Right ADC Level */ + { 0x19, 0x00 }, /* 19 ADC Biquad Level */ + { 0x1A, 0x00 }, /* 1A ADC Sidetone */ + { 0x1B, 0x00 }, /* 1B System Clock */ + { 0x1C, 0x00 }, /* 1C Clock Mode */ + { 0x1D, 0x00 }, /* 1D Any Clock 1 */ + { 0x1E, 0x00 }, /* 1E Any Clock 2 */ + { 0x1F, 0x00 }, /* 1F Any Clock 3 */ + + { 0x20, 0x00 }, /* 20 Any Clock 4 */ + { 0x21, 0x00 }, /* 21 Master Mode */ + { 0x22, 0x00 }, /* 22 Interface Format */ + { 0x23, 0x00 }, /* 23 TDM Format 1*/ + { 0x24, 0x00 }, /* 24 TDM Format 2*/ + { 0x25, 0x00 }, /* 25 I/O Configuration */ + { 0x26, 0x80 }, /* 26 Filter Config */ + { 0x27, 0x00 }, /* 27 DAI Playback Level */ + { 0x28, 0x00 }, /* 28 EQ Playback Level */ + { 0x29, 0x00 }, /* 29 Left HP Mixer */ + { 0x2A, 0x00 }, /* 2A Right HP Mixer */ + { 0x2B, 0x00 }, /* 2B HP Control */ + { 0x2C, 0x1A }, /* 2C Left HP Volume */ + { 0x2D, 0x1A }, /* 2D Right HP Volume */ + { 0x2E, 0x00 }, /* 2E Left Spk Mixer */ + { 0x2F, 0x00 }, /* 2F Right Spk Mixer */ + + { 0x30, 0x00 }, /* 30 Spk Control */ + { 0x31, 0x2C }, /* 31 Left Spk Volume */ + { 0x32, 0x2C }, /* 32 Right Spk Volume */ + { 0x33, 0x00 }, /* 33 ALC Timing */ + { 0x34, 0x00 }, /* 34 ALC Compressor */ + { 0x35, 0x00 }, /* 35 ALC Expander */ + { 0x36, 0x00 }, /* 36 ALC Gain */ + { 0x37, 0x00 }, /* 37 Rcv/Line OutL Mixer */ + { 0x38, 0x00 }, /* 38 Rcv/Line OutL Control */ + { 0x39, 0x15 }, /* 39 Rcv/Line OutL Volume */ + { 0x3A, 0x00 }, /* 3A Line OutR Mixer */ + { 0x3B, 0x00 }, /* 3B Line OutR Control */ + { 0x3C, 0x15 }, /* 3C Line OutR Volume */ + { 0x3D, 0x00 }, /* 3D Jack Detect */ + { 0x3E, 0x00 }, /* 3E Input Enable */ + { 0x3F, 0x00 }, /* 3F Output Enable */ + + { 0x40, 0x00 }, /* 40 Level Control */ + { 0x41, 0x00 }, /* 41 DSP Filter Enable */ + { 0x42, 0x00 }, /* 42 Bias Control */ + { 0x43, 0x00 }, /* 43 DAC Control */ + { 0x44, 0x06 }, /* 44 ADC Control */ + { 0x45, 0x00 }, /* 45 Device Shutdown */ + { 0x46, 0x00 }, /* 46 Equalizer Band 1 Coefficient B0 */ + { 0x47, 0x00 }, /* 47 Equalizer Band 1 Coefficient B0 */ + { 0x48, 0x00 }, /* 48 Equalizer Band 1 Coefficient B0 */ + { 0x49, 0x00 }, /* 49 Equalizer Band 1 Coefficient B1 */ + { 0x4A, 0x00 }, /* 4A Equalizer Band 1 Coefficient B1 */ + { 0x4B, 0x00 }, /* 4B Equalizer Band 1 Coefficient B1 */ + { 0x4C, 0x00 }, /* 4C Equalizer Band 1 Coefficient B2 */ + { 0x4D, 0x00 }, /* 4D Equalizer Band 1 Coefficient B2 */ + { 0x4E, 0x00 }, /* 4E Equalizer Band 1 Coefficient B2 */ + { 0x4F, 0x00 }, /* 4F Equalizer Band 1 Coefficient A1 */ + + { 0x50, 0x00 }, /* 50 Equalizer Band 1 Coefficient A1 */ + { 0x51, 0x00 }, /* 51 Equalizer Band 1 Coefficient A1 */ + { 0x52, 0x00 }, /* 52 Equalizer Band 1 Coefficient A2 */ + { 0x53, 0x00 }, /* 53 Equalizer Band 1 Coefficient A2 */ + { 0x54, 0x00 }, /* 54 Equalizer Band 1 Coefficient A2 */ + { 0x55, 0x00 }, /* 55 Equalizer Band 2 Coefficient B0 */ + { 0x56, 0x00 }, /* 56 Equalizer Band 2 Coefficient B0 */ + { 0x57, 0x00 }, /* 57 Equalizer Band 2 Coefficient B0 */ + { 0x58, 0x00 }, /* 58 Equalizer Band 2 Coefficient B1 */ + { 0x59, 0x00 }, /* 59 Equalizer Band 2 Coefficient B1 */ + { 0x5A, 0x00 }, /* 5A Equalizer Band 2 Coefficient B1 */ + { 0x5B, 0x00 }, /* 5B Equalizer Band 2 Coefficient B2 */ + { 0x5C, 0x00 }, /* 5C Equalizer Band 2 Coefficient B2 */ + { 0x5D, 0x00 }, /* 5D Equalizer Band 2 Coefficient B2 */ + { 0x5E, 0x00 }, /* 5E Equalizer Band 2 Coefficient A1 */ + { 0x5F, 0x00 }, /* 5F Equalizer Band 2 Coefficient A1 */ + + { 0x60, 0x00 }, /* 60 Equalizer Band 2 Coefficient A1 */ + { 0x61, 0x00 }, /* 61 Equalizer Band 2 Coefficient A2 */ + { 0x62, 0x00 }, /* 62 Equalizer Band 2 Coefficient A2 */ + { 0x63, 0x00 }, /* 63 Equalizer Band 2 Coefficient A2 */ + { 0x64, 0x00 }, /* 64 Equalizer Band 3 Coefficient B0 */ + { 0x65, 0x00 }, /* 65 Equalizer Band 3 Coefficient B0 */ + { 0x66, 0x00 }, /* 66 Equalizer Band 3 Coefficient B0 */ + { 0x67, 0x00 }, /* 67 Equalizer Band 3 Coefficient B1 */ + { 0x68, 0x00 }, /* 68 Equalizer Band 3 Coefficient B1 */ + { 0x69, 0x00 }, /* 69 Equalizer Band 3 Coefficient B1 */ + { 0x6A, 0x00 }, /* 6A Equalizer Band 3 Coefficient B2 */ + { 0x6B, 0x00 }, /* 6B Equalizer Band 3 Coefficient B2 */ + { 0x6C, 0x00 }, /* 6C Equalizer Band 3 Coefficient B2 */ + { 0x6D, 0x00 }, /* 6D Equalizer Band 3 Coefficient A1 */ + { 0x6E, 0x00 }, /* 6E Equalizer Band 3 Coefficient A1 */ + { 0x6F, 0x00 }, /* 6F Equalizer Band 3 Coefficient A1 */ + + { 0x70, 0x00 }, /* 70 Equalizer Band 3 Coefficient A2 */ + { 0x71, 0x00 }, /* 71 Equalizer Band 3 Coefficient A2 */ + { 0x72, 0x00 }, /* 72 Equalizer Band 3 Coefficient A2 */ + { 0x73, 0x00 }, /* 73 Equalizer Band 4 Coefficient B0 */ + { 0x74, 0x00 }, /* 74 Equalizer Band 4 Coefficient B0 */ + { 0x75, 0x00 }, /* 75 Equalizer Band 4 Coefficient B0 */ + { 0x76, 0x00 }, /* 76 Equalizer Band 4 Coefficient B1 */ + { 0x77, 0x00 }, /* 77 Equalizer Band 4 Coefficient B1 */ + { 0x78, 0x00 }, /* 78 Equalizer Band 4 Coefficient B1 */ + { 0x79, 0x00 }, /* 79 Equalizer Band 4 Coefficient B2 */ + { 0x7A, 0x00 }, /* 7A Equalizer Band 4 Coefficient B2 */ + { 0x7B, 0x00 }, /* 7B Equalizer Band 4 Coefficient B2 */ + { 0x7C, 0x00 }, /* 7C Equalizer Band 4 Coefficient A1 */ + { 0x7D, 0x00 }, /* 7D Equalizer Band 4 Coefficient A1 */ + { 0x7E, 0x00 }, /* 7E Equalizer Band 4 Coefficient A1 */ + { 0x7F, 0x00 }, /* 7F Equalizer Band 4 Coefficient A2 */ + + { 0x80, 0x00 }, /* 80 Equalizer Band 4 Coefficient A2 */ + { 0x81, 0x00 }, /* 81 Equalizer Band 4 Coefficient A2 */ + { 0x82, 0x00 }, /* 82 Equalizer Band 5 Coefficient B0 */ + { 0x83, 0x00 }, /* 83 Equalizer Band 5 Coefficient B0 */ + { 0x84, 0x00 }, /* 84 Equalizer Band 5 Coefficient B0 */ + { 0x85, 0x00 }, /* 85 Equalizer Band 5 Coefficient B1 */ + { 0x86, 0x00 }, /* 86 Equalizer Band 5 Coefficient B1 */ + { 0x87, 0x00 }, /* 87 Equalizer Band 5 Coefficient B1 */ + { 0x88, 0x00 }, /* 88 Equalizer Band 5 Coefficient B2 */ + { 0x89, 0x00 }, /* 89 Equalizer Band 5 Coefficient B2 */ + { 0x8A, 0x00 }, /* 8A Equalizer Band 5 Coefficient B2 */ + { 0x8B, 0x00 }, /* 8B Equalizer Band 5 Coefficient A1 */ + { 0x8C, 0x00 }, /* 8C Equalizer Band 5 Coefficient A1 */ + { 0x8D, 0x00 }, /* 8D Equalizer Band 5 Coefficient A1 */ + { 0x8E, 0x00 }, /* 8E Equalizer Band 5 Coefficient A2 */ + { 0x8F, 0x00 }, /* 8F Equalizer Band 5 Coefficient A2 */ + + { 0x90, 0x00 }, /* 90 Equalizer Band 5 Coefficient A2 */ + { 0x91, 0x00 }, /* 91 Equalizer Band 6 Coefficient B0 */ + { 0x92, 0x00 }, /* 92 Equalizer Band 6 Coefficient B0 */ + { 0x93, 0x00 }, /* 93 Equalizer Band 6 Coefficient B0 */ + { 0x94, 0x00 }, /* 94 Equalizer Band 6 Coefficient B1 */ + { 0x95, 0x00 }, /* 95 Equalizer Band 6 Coefficient B1 */ + { 0x96, 0x00 }, /* 96 Equalizer Band 6 Coefficient B1 */ + { 0x97, 0x00 }, /* 97 Equalizer Band 6 Coefficient B2 */ + { 0x98, 0x00 }, /* 98 Equalizer Band 6 Coefficient B2 */ + { 0x99, 0x00 }, /* 99 Equalizer Band 6 Coefficient B2 */ + { 0x9A, 0x00 }, /* 9A Equalizer Band 6 Coefficient A1 */ + { 0x9B, 0x00 }, /* 9B Equalizer Band 6 Coefficient A1 */ + { 0x9C, 0x00 }, /* 9C Equalizer Band 6 Coefficient A1 */ + { 0x9D, 0x00 }, /* 9D Equalizer Band 6 Coefficient A2 */ + { 0x9E, 0x00 }, /* 9E Equalizer Band 6 Coefficient A2 */ + { 0x9F, 0x00 }, /* 9F Equalizer Band 6 Coefficient A2 */ + + { 0xA0, 0x00 }, /* A0 Equalizer Band 7 Coefficient B0 */ + { 0xA1, 0x00 }, /* A1 Equalizer Band 7 Coefficient B0 */ + { 0xA2, 0x00 }, /* A2 Equalizer Band 7 Coefficient B0 */ + { 0xA3, 0x00 }, /* A3 Equalizer Band 7 Coefficient B1 */ + { 0xA4, 0x00 }, /* A4 Equalizer Band 7 Coefficient B1 */ + { 0xA5, 0x00 }, /* A5 Equalizer Band 7 Coefficient B1 */ + { 0xA6, 0x00 }, /* A6 Equalizer Band 7 Coefficient B2 */ + { 0xA7, 0x00 }, /* A7 Equalizer Band 7 Coefficient B2 */ + { 0xA8, 0x00 }, /* A8 Equalizer Band 7 Coefficient B2 */ + { 0xA9, 0x00 }, /* A9 Equalizer Band 7 Coefficient A1 */ + { 0xAA, 0x00 }, /* AA Equalizer Band 7 Coefficient A1 */ + { 0xAB, 0x00 }, /* AB Equalizer Band 7 Coefficient A1 */ + { 0xAC, 0x00 }, /* AC Equalizer Band 7 Coefficient A2 */ + { 0xAD, 0x00 }, /* AD Equalizer Band 7 Coefficient A2 */ + { 0xAE, 0x00 }, /* AE Equalizer Band 7 Coefficient A2 */ + { 0xAF, 0x00 }, /* AF ADC Biquad Coefficient B0 */ + + { 0xB0, 0x00 }, /* B0 ADC Biquad Coefficient B0 */ + { 0xB1, 0x00 }, /* B1 ADC Biquad Coefficient B0 */ + { 0xB2, 0x00 }, /* B2 ADC Biquad Coefficient B1 */ + { 0xB3, 0x00 }, /* B3 ADC Biquad Coefficient B1 */ + { 0xB4, 0x00 }, /* B4 ADC Biquad Coefficient B1 */ + { 0xB5, 0x00 }, /* B5 ADC Biquad Coefficient B2 */ + { 0xB6, 0x00 }, /* B6 ADC Biquad Coefficient B2 */ + { 0xB7, 0x00 }, /* B7 ADC Biquad Coefficient B2 */ + { 0xB8, 0x00 }, /* B8 ADC Biquad Coefficient A1 */ + { 0xB9, 0x00 }, /* B9 ADC Biquad Coefficient A1 */ + { 0xBA, 0x00 }, /* BA ADC Biquad Coefficient A1 */ + { 0xBB, 0x00 }, /* BB ADC Biquad Coefficient A2 */ + { 0xBC, 0x00 }, /* BC ADC Biquad Coefficient A2 */ + { 0xBD, 0x00 }, /* BD ADC Biquad Coefficient A2 */ + { 0xBE, 0x00 }, /* BE Digital Mic 3 Volume */ + { 0xBF, 0x00 }, /* BF Digital Mic 4 Volume */ + + { 0xC0, 0x00 }, /* C0 Digital Mic 34 Biquad Pre Atten */ + { 0xC1, 0x00 }, /* C1 Record TDM Slot */ + { 0xC2, 0x00 }, /* C2 Sample Rate */ + { 0xC3, 0x00 }, /* C3 Digital Mic 34 Biquad Coefficient C3 */ + { 0xC4, 0x00 }, /* C4 Digital Mic 34 Biquad Coefficient C4 */ + { 0xC5, 0x00 }, /* C5 Digital Mic 34 Biquad Coefficient C5 */ + { 0xC6, 0x00 }, /* C6 Digital Mic 34 Biquad Coefficient C6 */ + { 0xC7, 0x00 }, /* C7 Digital Mic 34 Biquad Coefficient C7 */ + { 0xC8, 0x00 }, /* C8 Digital Mic 34 Biquad Coefficient C8 */ + { 0xC9, 0x00 }, /* C9 Digital Mic 34 Biquad Coefficient C9 */ + { 0xCA, 0x00 }, /* CA Digital Mic 34 Biquad Coefficient CA */ + { 0xCB, 0x00 }, /* CB Digital Mic 34 Biquad Coefficient CB */ + { 0xCC, 0x00 }, /* CC Digital Mic 34 Biquad Coefficient CC */ + { 0xCD, 0x00 }, /* CD Digital Mic 34 Biquad Coefficient CD */ + { 0xCE, 0x00 }, /* CE Digital Mic 34 Biquad Coefficient CE */ + { 0xCF, 0x00 }, /* CF Digital Mic 34 Biquad Coefficient CF */ + + { 0xD0, 0x00 }, /* D0 Digital Mic 34 Biquad Coefficient D0 */ + { 0xD1, 0x00 }, /* D1 Digital Mic 34 Biquad Coefficient D1 */ +}; + +static bool max98090_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_IRQ_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_REV_ID: + return true; + default: + return false; + } +} + +static bool max98090_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_IRQ_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_IRQ_ENABLE: + case M98090_REG_RESERVED: + case M98090_REG_CFG_INPUT: + case M98090_REG_LVL_LINE: + case M98090_REG_CFG_LINE: + case M98090_REG_LVL_MIC1: + case M98090_REG_LVL_MIC2: + case M98090_REG_MIC_BIAS: + case M98090_REG_MIC_CFG1: + case M98090_REG_MIC_CFG2: + case M98090_REG_MIX_ADC_L: + case M98090_REG_MIX_ADC_R: + case M98090_REG_LVL_ADC_L: + case M98090_REG_LVL_ADC_R: + case M98090_REG_LVL_BIQUAD: + case M98090_REG_LVL_SIDETONE: + case M98090_REG_SYS_CLOCK: + case M98090_REG_CLOCK_MODE: + case M98090_REG_CLOCK_DAI1_NI_HI: + case M98090_REG_CLOCK_DAI2_NI_LO: + case M98090_REG_CLOCK_DAI3_MI_HI: + case M98090_REG_CLOCK_DAI4_MI_LO: + case M98090_REG_CLOCK_MAS_MODE: + case M98090_REG_DAI_INTERFACE_FORMAT: + case M98090_REG_DAI_TDM_CONTROL: + case M98090_REG_DAI_TDM_FORMAT: + case M98090_REG_DAI_IOCFG: + case M98090_REG_DAI_FILTERS: + case M98090_REG_DAI_LVL: + case M98090_REG_DAI_LVL_EQ: + case M98090_REG_MIX_HP_LEFT: + case M98090_REG_MIX_HP_RIGHT: + case M98090_REG_MIX_HP_CNTL: + case M98090_REG_LVL_HP_LEFT: + case M98090_REG_LVL_HP_RIGHT: + case M98090_REG_MIX_SPK_LEFT: + case M98090_REG_MIX_SPK_RIGHT: + case M98090_REG_MIX_SPK_CNTL: + case M98090_REG_LVL_SPK_LEFT: + case M98090_REG_LVL_SPK_RIGHT: + case M98090_REG_ALC_TIMING: + case M98090_REG_ALC_CMPR: + case M98090_REG_ALC_EXP: + case M98090_REG_LVL_ALC: + case M98090_REG_MIX_RCV_LEFT: + case M98090_REG_MIX_RCV_CTRL_LEFT: + case M98090_REG_LVL_RCV_LEFT: + case M98090_REG_MIX_RCV_RIGHT: + case M98090_REG_MIX_RCV_CNTL_RIGHT: + case M98090_REG_LVL_RCV_RIGHT: + case M98090_REG_CFG_JACK: + case M98090_REG_PWR_EN_IN: + case M98090_REG_PWR_EN_OUT: + case M98090_REG_CFG_LVL: + case M98090_REG_DSP_EQ_EN: + case M98090_REG_BIAS_CNTL: + case M98090_REG_DAC_CFG: + case M98090_REG_ADC_CFG: + case M98090_REG_PWR_SYS: + case M98090_REG_EQ_BASE ... M98090_REG_EQ_BASE + 0x68: + case M98090_REG_BIQUAD_BASE ... M98090_REG_BIQUAD_BASE + 0x0E: + case M98090_REG_DMIC3_VOLUME: + case M98090_REG_DMIC4_VOLUME: + case M98090_REG_DMIC34_BQ_PREATTEN: + case M98090_REG_RECORD_TDM_SLOT: + case M98090_REG_SAMPLE_RATE: + case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + return true; + default: + return false; + } +} + +static int max98090_reset(struct max98090_priv *max98090) +{ + int ret; + + /* Reset the codec by writing to this write-only reset register */ + ret = regmap_write(max98090->regmap, M98090_REG_SW_RESET, + M98090_SWRESET_MASK); + if (ret < 0) { + dev_err(max98090->codec->dev, + "Failed to reset codec: %d\n", ret); + return ret; + } + + msleep(20); + return ret; +} + +static const unsigned int max98090_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_mic_tlv, 0, 100, 0); + +static const unsigned int max98090_lin_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-600, 300, 0), + 4, 5, TLV_DB_SCALE_ITEM(1400, 600, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_adcboost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_adc_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_dv_tlv, -1500, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_sidetone_tlv, -6050, 200, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_alc_tlv, -1500, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcmakeup_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alccomp_tlv, -3100, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcexp_tlv, -6600, 100, 0); + +static const unsigned int max98090_mixout_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(-1200, 250, 0), + 2, 3, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +static const unsigned int max98090_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98090_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 4, TLV_DB_SCALE_ITEM(-4800, 400, 0), + 5, 10, TLV_DB_SCALE_ITEM(-2900, 300, 0), + 11, 14, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 15, 29, TLV_DB_SCALE_ITEM(-500, 100, 0), + 30, 39, TLV_DB_SCALE_ITEM(950, 50, 0), +}; + +static const unsigned int max98090_rcv_lout_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static int max98090_extmic_mux_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, M98090_REG_CFG_LINE); + + val = (val & M98090_EXTMIC_MASK) >> M98090_EXTMIC_SHIFT; + + max98090->extmic_mux = sel; + + /* Setting a channel is only valid if it is already On */ + if ((val == 1) || (val == 2)) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, M98090_REG_CFG_LINE, + M98090_EXTMIC_MASK, + sel << M98090_EXTMIC_SHIFT); + + return 0; +} + +static int max98090_extmic_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned int val = snd_soc_read(codec, M98090_REG_CFG_LINE); + + val = (val & M98090_EXTMIC_MASK) >> M98090_EXTMIC_SHIFT; + + if ((val == 1) || (val == 2)) { + /* If on, return the channel */ + val = val - 1; + max98090->extmic_mux = val; + } else { + /* If off, return last stored value */ + val = max98090->extmic_mux; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_LVL_MIC1: + select = &(max98090->mic1pre); + break; + case M98090_REG_LVL_MIC2: + select = &(max98090->mic2pre); + break; + case M98090_REG_LVL_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + if (val >= 1) { + /* If on, return the volume */ + val = val - 1; + *select = val; + } else { + /* If off, return last stored value */ + val = *select; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_LVL_MIC1: + select = &(max98090->mic1pre); + break; + case M98090_REG_LVL_MIC2: + select = &(max98090->mic2pre); + break; + case M98090_REG_LVL_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + *select = sel; + + /* Setting a volume is only valid if it is already On */ + if (val >= 1) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, mc->reg, + mask << mc->shift, + sel << mc->shift); + + return 0; +} + +static const char * max98090_extmic_mux_text[] = { "MIC1", "MIC2" }; + +static const struct soc_enum max98090_extmic_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC_SHIFT, + ARRAY_SIZE(max98090_extmic_mux_text), + max98090_extmic_mux_text); + +static const char * max98090_mixg_text[] = { "Normal", "6dB" }; + +static const struct soc_enum max98090_mixg135_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_LINE, M98090_MIXG135_SHIFT, + ARRAY_SIZE(max98090_mixg_text), max98090_mixg_text); + +static const struct soc_enum max98090_mixg246_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_LINE, M98090_MIXT246_SHIFT, + ARRAY_SIZE(max98090_mixg_text), max98090_mixg_text); + +static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; + +static const struct soc_enum max98090_osr128_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CFG, M98090_ADC_OSR128_SHIFT, + ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); + +static const char * max98090_enableddisabled_text[] = + { "Disabled", "Enabled" }; +static const char * max98090_enableddisabled_inv_text[] = + { "Enabled", "Disabled" }; +/* Note: Inverted Logic (0 = Enabled) */ + +static const struct soc_enum max98090_eqclpn_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_LVL_EQ, M98090_DAI_LVL_EQCLPN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct soc_enum max98090_sdien_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_SDIEN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_text), + max98090_enableddisabled_text); + +static const struct soc_enum max98090_sdoen_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_SDOEN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_text), + max98090_enableddisabled_text); + +static const struct soc_enum max98090_hizoff_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_HIZOFF_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const char *max98090_fltr_mode_text[] = { "Voice", "Music" }; + +static const struct soc_enum max98090_filter_mode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_FILTERS, M98090_DAI_FLT_MODE_SHIFT, + ARRAY_SIZE(max98090_fltr_mode_text), max98090_fltr_mode_text); + +static const struct soc_enum max98090_filter_dmic34mode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DMIC34MODE_SHIFT, + ARRAY_SIZE(max98090_fltr_mode_text), max98090_fltr_mode_text); + +static const char * max98090_dsts_text[] = + { "None", "Left", "Right", "Left and Right" }; + +static const struct soc_enum max98090_dsts_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_SIDETONE, M98090_DSTS_SHIFT, + ARRAY_SIZE(max98090_dsts_text), max98090_dsts_text); + +static const char * max98090_alcatk_text[] = + { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; + +static const struct soc_enum max98090_alcatk_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_TIMING, M98090_ALCATK_SHIFT, + ARRAY_SIZE(max98090_alcatk_text), max98090_alcatk_text); + +static const char * max98090_alcrls_text[] = + { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; + +static const struct soc_enum max98090_alcrls_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_TIMING, M98090_ALCRLS_SHIFT, + ARRAY_SIZE(max98090_alcrls_text), max98090_alcrls_text); + +static const char * max98090_alccmp_text[] = + { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; + +static const struct soc_enum max98090_alccmp_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_CMPR, M98090_ALCCMP_SHIFT, + ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); + +static const char * max98090_alcexp_text[] = { "1:1", "2:1", "3:1" }; + +static const struct soc_enum max98090_alcexp_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_EXP, M98090_ALCEXP_SHIFT, + ARRAY_SIZE(max98090_alcexp_text), max98090_alcexp_text); + +static const char * max98090_perf_pwr_text[] = { "High Performance", "Low Power" }; +static const char * max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; + +static const struct soc_enum max98090_dac_perfmode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CFG, M98090_DAC_PERFMODE_SHIFT, + ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); + +static const struct soc_enum max98090_dachp_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CFG, M98090_DACHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_adchp_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CFG, M98090_ADCHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_zdenn_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LVL, M98090_ZDENN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct soc_enum max98090_vs2enn_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LVL, M98090_VS2ENN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct soc_enum max98090_vsenn_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LVL, M98090_VSENN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct snd_kcontrol_new max98090_snd_controls[] = { + SOC_SINGLE("MIC Bias VCM Bandgap", M98090_REG_BIAS_CNTL, + M98090_VCM_MODE_SHIFT, M98090_VCM_MODE_NUM - 1, 0), + + SOC_SINGLE("DMIC MIC Comp Filter Config", M98090_REG_MIC_CFG2, + M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0), + + SOC_ENUM_EXT("External MIC Mux", max98090_extmic_mux_enum, + max98090_extmic_mux_get, max98090_extmic_mux_set), + + SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", + M98090_REG_LVL_MIC1, M98090_MIC_PA1EN_SHIFT, + M98090_MIC_PA1EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_EXT_TLV("MIC2 Boost Volume", + M98090_REG_LVL_MIC2, M98090_MIC_PA2EN_SHIFT, + M98090_MIC_PA2EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_TLV("MIC1 Volume", M98090_REG_LVL_MIC1, + M98090_MIC_PGAM1_SHIFT, M98090_MIC_PGAM1_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_TLV("MIC2 Volume", M98090_REG_LVL_MIC2, + M98090_MIC_PGAM2_SHIFT, M98090_MIC_PGAM2_NUM - 1, 1, + max98090_mic_tlv), + + SOC_ENUM("LINEA Single Ended Reduction", max98090_mixg135_enum), + SOC_ENUM("LINEB Single Ended Reduction", max98090_mixg246_enum), + + SOC_SINGLE_RANGE_TLV("LINEA Volume", M98090_REG_LVL_LINE, + M98090_LINAPGA_SHIFT, 0, M98090_LINAPGA_NUM - 1, 1, + max98090_lin_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Volume", M98090_REG_LVL_LINE, + M98090_LINBPGA_SHIFT, 0, M98090_LINBPGA_NUM - 1, 1, + max98090_lin_tlv), + + SOC_SINGLE("LINEA Ext Resistor Gain Mode", M98090_REG_CFG_LINE, + M98090_EXTBUFA_SHIFT, M98090_EXTBUFA_NUM - 1, 0), + SOC_SINGLE("LINEB Ext Resistor Gain Mode", M98090_REG_CFG_LINE, + M98090_EXTBUFB_SHIFT, M98090_EXTBUFB_NUM - 1, 0), + + SOC_SINGLE_TLV("ADCL Boost Volume", M98090_REG_LVL_ADC_L, + M98090_ADC_AVLG_SHIFT, M98090_ADC_AVLG_NUM - 1, 0, + max98090_adcboost_tlv), + SOC_SINGLE_TLV("ADCR Boost Volume", M98090_REG_LVL_ADC_R, + M98090_ADC_AVRG_SHIFT, M98090_ADC_AVLG_NUM - 1, 0, + max98090_adcboost_tlv), + + SOC_SINGLE_TLV("ADCL Volume", M98090_REG_LVL_ADC_L, + M98090_ADC_AVL_SHIFT, M98090_ADC_AVL_NUM - 1, 1, + max98090_adc_tlv), + SOC_SINGLE_TLV("ADCR Volume", M98090_REG_LVL_ADC_R, + M98090_ADC_AVR_SHIFT, M98090_ADC_AVR_NUM - 1, 1, + max98090_adc_tlv), + + SOC_ENUM("ADC Oversampling Rate", max98090_osr128_enum), + SOC_SINGLE("ADC Quantizer Dither", M98090_REG_ADC_CFG, + M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0), + SOC_ENUM("ADC High Performance Mode", max98090_adchp_enum), + + SOC_SINGLE("DAC Mono Mode", M98090_REG_DAI_IOCFG, + M98090_DAI_DMONO_SHIFT, M98090_DAI_DMONO_NUM - 1, 0), + SOC_ENUM("SDIN Mode", max98090_sdien_enum), + SOC_ENUM("SDOUT Mode", max98090_sdoen_enum), + SOC_ENUM("SDOUT Hi-Z Mode", max98090_hizoff_enum), + SOC_ENUM("Filter Mode", max98090_filter_mode_enum), + SOC_SINGLE("Record Path DC Blocking", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_AHPF_SHIFT, M98090_DAI_FLT_AHPF_NUM - 1, 0), + SOC_SINGLE("Playback Path DC Blocking", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DHPF_SHIFT, M98090_DAI_FLT_DHPF_NUM - 1, 0), + SOC_SINGLE_TLV("Digital BQ Volume", M98090_REG_LVL_BIQUAD, + M98090_AVBQ_SHIFT, M98090_AVBQ_NUM - 1, 1, max98090_dv_tlv), + SOC_ENUM("Digital Sidetone Source", max98090_dsts_enum), + SOC_SINGLE_EXT_TLV("Digital Sidetone Volume", + M98090_REG_LVL_SIDETONE, M98090_DVST_SHIFT, + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + SOC_SINGLE_TLV("Digital Coarse Volume", M98090_REG_DAI_LVL, + M98090_DAI_LVL_DVG_SHIFT, M98090_DAI_LVL_DVG_NUM - 1, 0, + max98090_dvg_tlv), + SOC_SINGLE_TLV("Digital Volume", M98090_REG_DAI_LVL, + M98090_DAI_LVL_DV_SHIFT, M98090_DAI_LVL_DV_NUM - 1, 1, + max98090_dv_tlv), + SND_SOC_BYTES("EQ Coefficients", M98090_REG_EQ_BASE, 105), + SOC_SINGLE("Digital EQ 3 Band Switch", M98090_REG_DSP_EQ_EN, + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 5 Band Switch", M98090_REG_DSP_EQ_EN, + M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 7 Band Switch", M98090_REG_DSP_EQ_EN, + M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0), + SOC_ENUM("Digital EQ Clipping Detection", max98090_eqclpn_enum), + SOC_SINGLE_TLV("Digital EQ Volume", M98090_REG_DAI_LVL_EQ, + M98090_DAI_LVL_DVEQ_SHIFT, M98090_DAI_LVL_DVEQ_NUM - 1, 1, + max98090_dv_tlv), + + SOC_SINGLE("ALC Enable", M98090_REG_ALC_TIMING, + M98090_ALCEN_SHIFT, M98090_ALCEN_NUM - 1, 0), + SOC_ENUM("ALC Attack Time", max98090_alcatk_enum), + SOC_ENUM("ALC Release Time", max98090_alcrls_enum), + SOC_SINGLE_TLV("ALC Make Up Volume", M98090_REG_LVL_ALC, + M98090_ALCG_SHIFT, M98090_ALCG_NUM - 1, 0, + max98090_alcmakeup_tlv), + SOC_ENUM("ALC Compression Ratio", max98090_alccmp_enum), + SOC_ENUM("ALC Expansion Ratio", max98090_alcexp_enum), + SOC_SINGLE_TLV("ALC Compression Threshold Volume", + M98090_REG_ALC_CMPR, M98090_ALCTHC_SHIFT, + M98090_ALCTHC_NUM - 1, 1, max98090_alccomp_tlv), + SOC_SINGLE_TLV("ALC Expansion Threshold Volume", + M98090_REG_ALC_EXP, M98090_ALCTHE_SHIFT, + M98090_ALCTHE_NUM - 1, 1, max98090_alcexp_tlv), + + SOC_ENUM("DAC HP Playback Performance Mode", + max98090_dac_perfmode_enum), + SOC_ENUM("DAC High Performance Mode", max98090_dachp_enum), + + SOC_SINGLE_TLV("Headphone Left Mixer Volume", + M98090_REG_MIX_HP_CNTL, M98090_MIXHPLG_SHIFT, + M98090_MIXHPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Headphone Right Mixer Volume", + M98090_REG_MIX_HP_CNTL, M98090_MIXHPRG_SHIFT, + M98090_MIXHPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Speaker Left Mixer Volume", + M98090_REG_MIX_SPK_CNTL, M98090_MIXSPLG_SHIFT, + M98090_MIXSPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Speaker Right Mixer Volume", + M98090_REG_MIX_SPK_CNTL, M98090_MIXSPRG_SHIFT, + M98090_MIXSPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Receiver Left Mixer Volume", + M98090_REG_MIX_RCV_CTRL_LEFT, M98090_MIXRCVLG_SHIFT, + M98090_MIXRCVLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Receiver Right Mixer Volume", + M98090_REG_MIX_RCV_CNTL_RIGHT, M98090_MIXRCVRG_SHIFT, + M98090_MIXRCVRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_DOUBLE_R_TLV("Headphone Volume", M98090_REG_LVL_HP_LEFT, + M98090_REG_LVL_HP_RIGHT, M98090_HPVOLL_SHIFT, + M98090_HPVOLL_NUM - 1, 0, max98090_hp_tlv), + + SOC_DOUBLE_R_RANGE_TLV("Speaker Volume", + M98090_REG_LVL_SPK_LEFT, M98090_REG_LVL_SPK_RIGHT, + M98090_SPVOLL_SHIFT, 24, M98090_SPVOLL_NUM - 1 + 24, + 0, max98090_spk_tlv), + + SOC_DOUBLE_R_TLV("Receiver Volume", M98090_REG_LVL_RCV_LEFT, + M98090_REG_LVL_RCV_RIGHT, M98090_RCVLVOL_SHIFT, + M98090_RCVLVOL_NUM - 1, 0, max98090_rcv_lout_tlv), + + SOC_SINGLE("Headphone Left Switch", M98090_REG_LVL_HP_LEFT, + M98090_HPLM_SHIFT, 1, 1), + SOC_SINGLE("Headphone Right Switch", M98090_REG_LVL_HP_RIGHT, + M98090_HPRM_SHIFT, 1, 1), + + SOC_SINGLE("Speaker Left Switch", M98090_REG_LVL_SPK_LEFT, + M98090_SPLM_SHIFT, 1, 1), + SOC_SINGLE("Speaker Right Switch", M98090_REG_LVL_SPK_RIGHT, + M98090_SPRM_SHIFT, 1, 1), + + SOC_SINGLE("Receiver Left Switch", M98090_REG_LVL_RCV_LEFT, + M98090_RCVLM_SHIFT, 1, 1), + SOC_SINGLE("Receiver Right Switch", M98090_REG_LVL_RCV_RIGHT, + M98090_RCVRM_SHIFT, 1, 1), + + SOC_ENUM("Zero-Crossing Detection", max98090_zdenn_enum), + SOC_ENUM("Enhanced Volume Smoothing", max98090_vs2enn_enum), + SOC_ENUM("Volume Adjustment Smoothing", max98090_vsenn_enum), + + SOC_SINGLE("Quick Setup System Clock", M98090_REG_QCFG_SYS_CLK, + M98090_CLK_ALL_SHIFT, M98090_CLK_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Sample Rate", M98090_REG_QCFG_RATE, + M98090_SR_ALL_SHIFT, M98090_SR_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup DAI Interface", M98090_REG_QCFG_DAI, + M98090_DAI_ALL_SHIFT, M98090_DAI_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup DAC Path", M98090_REG_QCFG_DAC, + M98090_DIG2_ALL_SHIFT, M98090_DIG2_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup MIC/Direct ADC", M98090_REG_QCFG_MIC_PATH, + M98090_MIC_ALL_SHIFT, M98090_MIC_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Line ADC", M98090_REG_QCFG_LINE_PATH, + M98090_LINE_ALL_SHIFT, M98090_LINE_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Analog MIC Loop", M98090_REG_QCFG_MIC_LOOP, + M98090_AMIC_ALL_SHIFT, M98090_AMIC_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Analog Line Loop", + M98090_REG_QCFG_LINE_LOOP, M98090_ALIN_ALL_SHIFT, + M98090_ALIN_ALL_NUM - 1, 0), + + SND_SOC_BYTES("Biquad Coefficients", M98090_REG_BIQUAD_BASE, 15), + SOC_SINGLE("Biquad Switch", M98090_REG_DSP_EQ_EN, + M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0), +}; + +static const struct snd_kcontrol_new max98091_snd_controls[] = { + + SOC_SINGLE("DMIC34 Zeropad", M98090_REG_SAMPLE_RATE, + M98090_DMIC34_ZEROPAD_SHIFT, + M98090_DMIC34_ZEROPAD_NUM - 1, 0), + + SOC_ENUM("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum), + SOC_SINGLE("DMIC34 DC Blocking", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DMIC34HPF_SHIFT, + M98090_DAI_FLT_DMIC34HPF_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC3 Boost Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3G_SHIFT, M98090_DMIC_AV3G_NUM - 1, 0, + max98090_adcboost_tlv), + SOC_SINGLE_TLV("DMIC4 Boost Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4G_SHIFT, M98090_DMIC_AV4G_NUM - 1, 0, + max98090_adcboost_tlv), + + SOC_SINGLE_TLV("DMIC3 Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3_SHIFT, M98090_DMIC_AV3_NUM - 1, 1, + max98090_adc_tlv), + SOC_SINGLE_TLV("DMIC4 Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4_SHIFT, M98090_DMIC_AV4_NUM - 1, 1, + max98090_adc_tlv), + + SND_SOC_BYTES("DMIC34 Biquad Coefficients", + M98090_REG_DMIC34_BIQUAD_BASE, 15), + SOC_SINGLE("DMIC34 Biquad Switch", M98090_REG_DSP_EQ_EN, + M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC34 BQ PreAttenuation Volume", + M98090_REG_DMIC34_BQ_PREATTEN, M98090_AV34BQ_SHIFT, + M98090_AV34BQ_NUM - 1, 1, max98090_dv_tlv), + + SOC_SINGLE("Rev ID", M98090_REG_REV_ID, + M98090_REVID_SHIFT, M98090_REVID_NUM - 1, 0), +}; + +static int max98090_mic_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event, + int channel) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, M98090_REG_CFG_LINE); + + val = (val & M98090_EXTMIC_MASK) >> M98090_EXTMIC_SHIFT; + + if ((val == (M98090_EXTMIC_MIC1 >> M98090_EXTMIC_SHIFT)) || + (val == (M98090_EXTMIC_MIC2 >> M98090_EXTMIC_SHIFT))) + { + max98090->extmic_mux = val - 1; /* Update for volatile */ + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to selected channel */ + max98090->extmic_mux = channel; + val = max98090->extmic_mux + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = M98090_EXTMIC_NONE >> M98090_EXTMIC_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98090_REG_CFG_LINE, M98090_EXTMIC_MASK, + val << M98090_EXTMIC_SHIFT); + + return 0; +} + +static int max98090_mic1_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return max98090_mic_mux_event(w, kcontrol, event, 0); +} + +static int max98090_mic2_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return max98090_mic_mux_event(w, kcontrol, event, 1); +} + +static int max98090_micinput_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, w->reg); + + if (w->reg == M98090_REG_LVL_MIC1) + val = (val & M98090_MIC_PA1EN_MASK) >> M98090_MIC_PA1EN_SHIFT; + else + val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; + + + if (val >= 1) { + if (w->reg == M98090_REG_LVL_MIC1) { + max98090->mic1pre = val - 1; /* Update for volatile */ + } else { + max98090->mic2pre = val - 1; /* Update for volatile */ + } + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to most recently selected volume */ + if (w->reg == M98090_REG_LVL_MIC1) + val = max98090->mic1pre + 1; + else + val = max98090->mic2pre + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = 0; + break; + default: + return -EINVAL; + } + + if (w->reg == M98090_REG_LVL_MIC1) + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA1EN_MASK, + val << M98090_MIC_PA1EN_SHIFT); + else + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA2EN_MASK, + val << M98090_MIC_PA2EN_SHIFT); + + return 0; +} + +static int max98090_sidetone_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, w->reg, M98090_DVST_MASK, + (1 + max98090->sidetone) << M98090_DVST_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, M98090_DVST_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char *mic1_mux_text[] = { "IN12", "IN56" }; + +static const struct soc_enum mic1_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC1_SHIFT, + ARRAY_SIZE(mic1_mux_text), mic1_mux_text); + +static const struct snd_kcontrol_new max98090_mic1_mux = + SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); + +static const char *mic2_mux_text[] = { "IN34", "IN56" }; + +static const struct soc_enum mic2_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC2_SHIFT, + ARRAY_SIZE(mic2_mux_text), mic2_mux_text); + +static const struct snd_kcontrol_new max98090_mic2_mux = + SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); + +static const char * max98090_micpre_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_mic1pre_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_MIC1, M98090_MIC_PA1EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const struct soc_enum max98090_mic2pre_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_MIC2, M98090_MIC_PA2EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const char * max98090_extmic_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_extmic_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC_SHIFT, + ARRAY_SIZE(max98090_extmic_text), max98090_extmic_text); + +static const struct snd_kcontrol_new max98090_extmic = + SOC_DAPM_ENUM("External MIC", max98090_extmic_enum); + +/* LINEA mixer switch */ +static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1 Switch", M98090_REG_CFG_INPUT, + M98090_IN1SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN3 Switch", M98090_REG_CFG_INPUT, + M98090_IN3SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN5 Switch", M98090_REG_CFG_INPUT, + M98090_IN5SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_CFG_INPUT, + M98090_IN34DIFF_SHIFT, 1, 0), +}; + +/* LINEB mixer switch */ +static const struct snd_kcontrol_new max98090_lineb_mixer_controls[] = { + SOC_DAPM_SINGLE("IN2 Switch", M98090_REG_CFG_INPUT, + M98090_IN2SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN4 Switch", M98090_REG_CFG_INPUT, + M98090_IN4SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN6 Switch", M98090_REG_CFG_INPUT, + M98090_IN6SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_CFG_INPUT, + M98090_IN56DIFF_SHIFT, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98090_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_MIC2_SHIFT, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_MIC2_SHIFT, 1, 0), +}; + +static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; + +static const struct soc_enum ltenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct soc_enum ltenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct snd_kcontrol_new max98090_ltenl_mux = + SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); + +static const struct snd_kcontrol_new max98090_ltenr_mux = + SOC_DAPM_ENUM("LTENR Mux", ltenr_mux_enum); + +static const char *lben_mux_text[] = { "Normal", "Loopback" }; + +static const struct soc_enum lbenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct soc_enum lbenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct snd_kcontrol_new max98090_lbenl_mux = + SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); + +static const struct snd_kcontrol_new max98090_lbenr_mux = + SOC_DAPM_ENUM("LBENR Mux", lbenr_mux_enum); + +/* Left speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_MIC2_SHIFT, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_MIC2_SHIFT, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_MIC2_SHIFT, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_MIC2_SHIFT, 1, 0), +}; + +/* Left receiver mixer switch */ +static const struct snd_kcontrol_new max98090_left_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_MIC2_SHIFT, 1, 0), +}; + +/* Right receiver mixer switch */ +static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_MIC2_SHIFT, 1, 0), +}; + +static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; + +static const struct soc_enum linmod_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_MIX_RCV_RIGHT, M98090_LINMOD_SHIFT, + ARRAY_SIZE(linmod_mux_text), linmod_mux_text); + +static const struct snd_kcontrol_new max98090_linmod_mux = + SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); + +static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; + +/* + * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable + */ +static const struct soc_enum mixhplsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_MIX_HP_CNTL, M98090_MIXHPLSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhplsel_mux = + SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); + +static const struct soc_enum mixhprsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_MIX_HP_CNTL, M98090_MIXHPRSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhprsel_mux = + SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); + +static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMICL"), + SND_SOC_DAPM_INPUT("DMICR"), + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN12"), + SND_SOC_DAPM_INPUT("IN34"), + SND_SOC_DAPM_INPUT("IN56"), + + SND_SOC_DAPM_MICBIAS("MICBIAS", M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_SHIFT, 0), + SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_PWR_SYS, + M98090_PWR_SHDNN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VCM", M98090_REG_BIAS_CNTL, + M98090_VCM_MODE_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDIEN", M98090_REG_DAI_IOCFG, + M98090_DAI_SDIEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_DAI_IOCFG, + M98090_DAI_SDOEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMICL_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMICR_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_AHPF_SHIFT, 0, NULL, 0), + +/* + * Note: Sysclk and misc power supplies are taken care of by SHDN and VCM + */ + + SND_SOC_DAPM_MUX_E("MIC1 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic1_mux, + max98090_mic1_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX_E("MIC2 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic2_mux, + max98090_mic2_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_LVL_MIC1, + M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98090_REG_LVL_MIC2, + M98090_MIC_PA2EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("LINEA Mixer", SND_SOC_NOPM, 0, 0, + &max98090_linea_mixer_controls[0], + ARRAY_SIZE(max98090_linea_mixer_controls)), + + SND_SOC_DAPM_MIXER("LINEB Mixer", SND_SOC_NOPM, 0, 0, + &max98090_lineb_mixer_controls[0], + ARRAY_SIZE(max98090_lineb_mixer_controls)), + + SND_SOC_DAPM_PGA("LINEA Input", M98090_REG_PWR_EN_IN, + M98090_PWR_LINEAEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("LINEB Input", M98090_REG_PWR_EN_IN, + M98090_PWR_LINEBEN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_adc_mixer_controls[0], + ARRAY_SIZE(max98090_left_adc_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_adc_mixer_controls[0], + ARRAY_SIZE(max98090_right_adc_mixer_controls)), + + SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_PWR_EN_IN, + M98090_PWR_ADLEN_SHIFT, 0), + SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_PWR_EN_IN, + M98090_PWR_ADREN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", "HiFi Capture", 1, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("LBENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenl_mux), + + SND_SOC_DAPM_MUX("LBENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenr_mux), + + SND_SOC_DAPM_MUX("LTENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenl_mux), + + SND_SOC_DAPM_MUX("LTENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenr_mux), + + SND_SOC_DAPM_PGA_E("Sidetone", M98090_REG_LVL_SIDETONE, + M98090_DVST_SHIFT, 0, NULL, 0, max98090_sidetone_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_AIF_IN("AIFINL", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", "HiFi Playback", 1, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DACL", NULL, M98090_REG_PWR_EN_OUT, + M98090_PWR_DALEN_SHIFT, 0), + SND_SOC_DAPM_DAC("DACR", NULL, M98090_REG_PWR_EN_OUT, + M98090_PWR_DAREN_SHIFT, 0), + + SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_hp_mixer_controls[0], + ARRAY_SIZE(max98090_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_hp_mixer_controls[0], + ARRAY_SIZE(max98090_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_left_rcv_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_right_rcv_mixer_controls)), + + SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_MIX_RCV_RIGHT, + M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_MIX_HP_CNTL, + M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_MIX_HP_CNTL, + M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + + SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_HPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_HPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_SPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_SPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("RCV Left Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_RCVLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("RCV Right Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_RCVREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RCVL"), + SND_SOC_DAPM_OUTPUT("RCVR"), +}; + +static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + + SND_SOC_DAPM_SUPPLY("DMIC3_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMIC3_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMIC4_SHIFT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { + + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, + {"MIC1 Input", NULL, "MICBIAS"}, + {"MIC2 Input", NULL, "MICBIAS"}, + + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, + {"DMICL", NULL, "AHPF"}, + {"DMICR", NULL, "AHPF"}, + + /* MIC1 input mux */ + {"MIC1 Mux", "IN12", "IN12"}, + {"MIC1 Mux", "IN56", "IN56"}, + + /* MIC2 input mux */ + {"MIC2 Mux", "IN34", "IN34"}, + {"MIC2 Mux", "IN56", "IN56"}, + + {"MIC1 Input", NULL, "MIC1 Mux"}, + {"MIC2 Input", NULL, "MIC2 Mux"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "IN12 Switch", "IN12"}, + {"Left ADC Mixer", "IN34 Switch", "IN34"}, + {"Left ADC Mixer", "IN56 Switch", "IN56"}, + {"Left ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Left ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "IN12 Switch", "IN12"}, + {"Right ADC Mixer", "IN34 Switch", "IN34"}, + {"Right ADC Mixer", "IN56 Switch", "IN56"}, + {"Right ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Right ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Line A input mixer */ + {"LINEA Mixer", "IN1 Switch", "IN1"}, + {"LINEA Mixer", "IN3 Switch", "IN3"}, + {"LINEA Mixer", "IN5 Switch", "IN5"}, + {"LINEA Mixer", "IN34 Switch", "IN34"}, + + /* Line B input mixer */ + {"LINEB Mixer", "IN2 Switch", "IN2"}, + {"LINEB Mixer", "IN4 Switch", "IN4"}, + {"LINEB Mixer", "IN6 Switch", "IN6"}, + {"LINEB Mixer", "IN56 Switch", "IN56"}, + + {"LINEA Input", NULL, "LINEA Mixer"}, + {"LINEB Input", NULL, "LINEB Mixer"}, + + /* Inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + {"ADCL", NULL, "SHDN"}, + {"ADCR", NULL, "SHDN"}, + {"ADCL", NULL, "VCM"}, + {"ADCR", NULL, "VCM"}, + + {"LBENL Mux", "Normal", "ADCL"}, + {"LBENL Mux", "Normal", "DMICL"}, + {"LBENL Mux", "Loopback", "LTENL Mux"}, + {"LBENR Mux", "Normal", "ADCR"}, + {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Loopback", "LTENR Mux"}, + + {"AIFOUTL", NULL, "LBENL Mux"}, + {"AIFOUTR", NULL, "LBENR Mux"}, + {"AIFOUTL", NULL, "SHDN"}, + {"AIFOUTR", NULL, "SHDN"}, + {"AIFOUTL", NULL, "VCM"}, + {"AIFOUTR", NULL, "VCM"}, + {"AIFOUTL", NULL, "SDOEN"}, + {"AIFOUTR", NULL, "SDOEN"}, + + {"LTENL Mux", "Normal", "AIFINL"}, + {"LTENL Mux", "Loopthrough", "LBENL Mux"}, + {"LTENR Mux", "Normal", "AIFINR"}, + {"LTENR Mux", "Loopthrough", "LBENR Mux"}, + + {"DACL", NULL, "LTENL Mux"}, + {"DACR", NULL, "LTENR Mux"}, + + {"AIFINL", NULL, "SHDN"}, + {"AIFINR", NULL, "SHDN"}, + {"AIFINL", NULL, "VCM"}, + {"AIFINR", NULL, "VCM"}, + {"AIFINL", NULL, "SDIEN"}, + {"AIFINR", NULL, "SDIEN"}, + {"DACL", NULL, "SHDN"}, + {"DACR", NULL, "SHDN"}, + {"DACL", NULL, "VCM"}, + {"DACR", NULL, "VCM"}, + + /* Left headphone output mixer */ + {"Left Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Left Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Left Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right headphone output mixer */ + {"Right Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Right Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Right Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left speaker output mixer */ + {"Left Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Left Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Left Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right speaker output mixer */ + {"Right Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Right Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Right Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left Receiver output mixer */ + {"Left Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Left Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Left Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right Receiver output mixer */ + {"Right Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Right Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Right Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + {"MIXHPLSEL Mux", "HP Mixer", "Left Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Left Out", NULL, "DACL"}, + {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + + {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Right Out", NULL, "DACR"}, + {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + + {"SPK Left Out", NULL, "Left Speaker Mixer"}, + {"SPK Right Out", NULL, "Right Speaker Mixer"}, + {"RCV Left Out", NULL, "Left Receiver Mixer"}, + + {"LINMOD Mux", "Left and Right", "Right Receiver Mixer"}, + {"LINMOD Mux", "Left Only", "Left Receiver Mixer"}, + {"RCV Right Out", NULL, "LINMOD Mux"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RCVL", NULL, "RCV Left Out"}, + {"RCVR", NULL, "RCV Right Out"}, + +}; + +static const struct snd_soc_dapm_route max98091_dapm_routes[] = { + + /* DMIC inputs */ + {"DMIC3", NULL, "DMIC3_ENA"}, + {"DMIC4", NULL, "DMIC4_ENA"}, + {"DMIC3", NULL, "AHPF"}, + {"DMIC4", NULL, "AHPF"}, + +}; + +static int max98090_add_widgets(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_add_codec_controls(codec, max98090_snd_controls, + ARRAY_SIZE(max98090_snd_controls)); + + if (max98090->devtype == MAX98091) { + snd_soc_add_codec_controls(codec, max98091_snd_controls, + ARRAY_SIZE(max98091_snd_controls)); + } + + snd_soc_dapm_new_controls(dapm, max98090_dapm_widgets, + ARRAY_SIZE(max98090_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98090_dapm_routes, + ARRAY_SIZE(max98090_dapm_routes)); + + if (max98090->devtype == MAX98091) { + snd_soc_dapm_new_controls(dapm, max98091_dapm_widgets, + ARRAY_SIZE(max98091_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, + ARRAY_SIZE(max98091_dapm_routes)); + + } + + return 0; +} + +static const int pclk_rates[] = { + 12000000, 12000000, 13000000, 13000000, + 16000000, 16000000, 19200000, 19200000 +}; + +static const int lrclk_rates[] = { + 8000, 16000, 8000, 16000, + 8000, 16000, 8000, 16000 +}; + +static const int user_pclk_rates[] = { + 13000000, 13000000 +}; + +static const int user_lrclk_rates[] = { + 44100, 48000 +}; + +static const unsigned long long ni_value[] = { + 3528, 768 +}; + +static const unsigned long long mi_value[] = { + 8125, 1625 +}; + +static void max98090_configure_bclk(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned long long ni; + int i; + + if (!max98090->sysclk) { + dev_err(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!max98090->bclk || !max98090->lrclk) { + dev_err(codec->dev, "No audio clocks configured\n"); + return; + } + + /* Skip configuration when operating as slave */ + if (!(snd_soc_read(codec, M98090_REG_CLOCK_MAS_MODE) & + M98090_CLK_MAS_MASK)) { + return; + } + + /* Check for supported PCLK to LRCLK ratios */ + for (i = 0; i < ARRAY_SIZE(pclk_rates); i++) { + if ((pclk_rates[i] == max98090->sysclk) && + (lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found supported PCLK to LRCLK rates 0x%x\n", + i + 0x8); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_FREQ1_MASK, + (i + 0x8) << M98090_CLK_FREQ1_SHIFT); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, 0); + return; + } + } + + /* Check for user calculated MI and NI ratios */ + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { + if ((user_pclk_rates[i] == max98090->sysclk) && + (user_lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found user supported PCLK to LRCLK rates\n"); + dev_dbg(codec->dev, "i %d ni %lld mi %lld\n", + i, ni_value[i], mi_value[i]); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_FREQ1_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, + 1 << M98090_CLK_USE_M1_SHIFT); + + snd_soc_write(codec, M98090_REG_CLOCK_DAI1_NI_HI, + (ni_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_DAI2_NI_LO, + ni_value[i] & 0xFF); + snd_soc_write(codec, M98090_REG_CLOCK_DAI3_MI_HI, + (mi_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_DAI4_MI_LO, + mi_value[i] & 0xFF); + + return; + } + } + + /* + * Calculate based on MI = 65536 (not as good as either method above) + */ + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_FREQ1_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, 0); + + /* + * Configure NI when operating as master + * Note: There is a small, but significant audio quality improvement + * by calculating ni and mi. + */ + ni = 65536ULL * (max98090->lrclk < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)max98090->lrclk; + do_div(ni, (unsigned long long int)max98090->sysclk); + dev_info(codec->dev, "No better method found\n"); + dev_info(codec->dev, "Calculating ni %lld with mi 65536\n", ni); + snd_soc_write(codec, M98090_REG_CLOCK_DAI1_NI_HI, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_DAI2_NI_LO, ni & 0xFF); +} + +static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + u8 regval; + + max98090->dai_fmt = fmt; + cdata = &max98090->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Set to slave mode PLL - MAS mode off */ + snd_soc_write(codec, + M98090_REG_CLOCK_DAI1_NI_HI, 0x00); + snd_soc_write(codec, + M98090_REG_CLOCK_DAI2_NI_LO, 0x00); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, 0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + if (max98090->tdm_slots == 4) { + /* TDM */ + regval |= M98090_CLK_MAS_MASK | + M98090_CLK_BSEL_64; + } else if (max98090->tdm_slots == 3) { + /* TDM */ + regval |= M98090_CLK_MAS_MASK | + M98090_CLK_BSEL_48; + } else { + /* Few TDM slots, or No TDM */ + regval |= M98090_CLK_MAS_MASK | + M98090_CLK_BSEL_32; + } + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + snd_soc_write(codec, M98090_REG_CLOCK_MAS_MODE, regval); + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98090_DAI_DLY_MASK; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regval |= M98090_DAI_RJ_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Not supported mode */ + default: + dev_err(codec->dev, "DAI format unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98090_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98090_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98090_DAI_BCI_MASK|M98090_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + +/* + * This accommodates an inverted logic in the MAX98090 chip + * for Bit Clock Invert (BCI). The inverted logic is only seen + * for the case of TDM mode. The remaining cases have normal logic. + */ + if (max98090->tdm_slots > 1) { + regval ^= M98090_DAI_BCI_MASK; + } + + snd_soc_write(codec, + M98090_REG_DAI_INTERFACE_FORMAT, regval); + } + + return 0; +} + +static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + cdata = &max98090->dai[0]; + + if (slots < 0 || slots > 4) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_width = slot_width; + + if (max98090->tdm_slots > 1) { + /* SLOTL SLOTR SLOTDLY */ + snd_soc_write(codec, M98090_REG_DAI_TDM_FORMAT, + 0 << M98090_DAI_TDM_SLOTL_SHIFT | + 1 << M98090_DAI_TDM_SLOTR_SHIFT | + 0 << M98090_DAI_TDM_SLOTDLY_SHIFT); + + /* FSW TDM */ + snd_soc_update_bits(codec, M98090_REG_DAI_TDM_CONTROL, + M98090_DAI_TDM_MASK, + M98090_DAI_TDM_MASK); + } + + /* + * Normally advisable to set TDM first, but this permits either order + */ + cdata->fmt = 0; + max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + + return 0; +} + +static int max98090_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { + /* + * Set to normal bias level. + */ + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS, + M98090_MBVSEL_MASK, M98090_MBVSEL_2V4); + + snd_soc_update_bits(codec, M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_MASK, M98090_PWR_MBEN_MASK); + } + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_MASK, 0); + /* Set internal pull-up to lowest power mode */ + snd_soc_update_bits(codec, M98090_REG_CFG_JACK, + M98090_JDWK_MASK, M98090_JDWK_MASK); + regcache_mark_dirty(max98090->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const int comp_pclk_rates[] = { + 11289600, 12288000, 12000000, 13000000, 19200000 +}; + +static const int dmic_micclk[] = { + 2, 2, 2, 2, 4, 2 +}; + +static const int comp_lrclk_rates[] = { + 8000, 16000, 32000, 44100, 48000, 96000 +}; + +static const int dmic_comp[6][6] = { + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 1, 1, 1}, + {7, 8, 3, 1, 2, 2}, + {7, 8, 3, 3, 3, 3} +}; + +static int max98090_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int i, j; + + cdata = &max98090->dai[0]; + max98090->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + max98090->bclk *= 2; + + max98090->lrclk = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98090_REG_DAI_INTERFACE_FORMAT, + M98090_DAI_WS_MASK, 0); + break; + default: + return -EINVAL; + } + + max98090_configure_bclk(codec); + + cdata->rate = max98090->lrclk; + + /* Update filter mode */ + if (max98090->lrclk < 24000) + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_MODE_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_MODE_MASK, M98090_DAI_FLT_MODE_MASK); + + /* Update sample rate mode */ + if (max98090->lrclk < 50000) + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DHF_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DHF_MASK, M98090_DAI_FLT_DHF_MASK); + + /* Check for supported PCLK to LRCLK ratios */ + for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) { + if (comp_pclk_rates[j] == max98090->sysclk) { + break; + } + } + + for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { + if (max98090->lrclk <= (comp_lrclk_rates[i] + + comp_lrclk_rates[i + 1]) / 2) { + break; + } + } + + snd_soc_update_bits(codec, M98090_REG_MIC_CFG1, + M98090_MICCLK_MASK, + dmic_micclk[j] << M98090_MICCLK_SHIFT); + + snd_soc_update_bits(codec, M98090_REG_MIC_CFG2, + M98090_DMIC_COMP_MASK, + dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT); + + return 0; +} + +/* + * PLL / Sysclk + */ +static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + /* Requested clock frequency is already setup */ + if (freq == max98090->sysclk) + return 0; + + /* Setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 40MHz).. + * 0x03 (when master clk is 40MHz to 60MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98090_REG_SYS_CLOCK, + M98090_CLK_PSCLK_DIV1); + } else if ((freq >= 20000000) && (freq < 40000000)) { + snd_soc_write(codec, M98090_REG_SYS_CLOCK, + M98090_CLK_PSCLK_DIV2); + } else if ((freq >= 40000000) && (freq < 60000000)) { + snd_soc_write(codec, M98090_REG_SYS_CLOCK, + M98090_CLK_PSCLK_DIV4); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + max98090->sysclk = freq; + + max98090_configure_bclk(codec); + + return 0; +} + +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int regval; + + regval = mute ? M98090_DAI_LVL_DVM_MASK : 0; + snd_soc_update_bits(codec, M98090_REG_DAI_LVL, + M98090_DAI_LVL_DVM_MASK, regval); + + return 0; +} + +static void max98090_jack_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = container_of(work, + struct max98090_priv, + jack_work.work); + struct snd_soc_codec *codec = max98090->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int status = 0; + int reg; + + /* Read a second time */ + if (max98090->jack_state == M98090_JACK_STATE_NO_HEADSET) { + + snd_soc_update_bits(codec, M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_MASK, 0); + + /* Strong pull up allows mic detection */ + snd_soc_update_bits(codec, M98090_REG_CFG_JACK, + M98090_JDWK_MASK, 0); + + msleep(50); + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + /* Weak pull up allows only insertion detection */ + snd_soc_update_bits(codec, M98090_REG_CFG_JACK, + M98090_JDWK_MASK, M98090_JDWK_MASK); + } else { + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + } + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { + case M98090_LSNS_MASK | M98090_JKSNS_MASK: + { + dev_info(codec->dev, "No Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + max98090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + status |= 0; + + break; + } + + case 0: + { + if (max98090->jack_state == + M98090_JACK_STATE_HEADSET) { + + dev_info(codec->dev, + "Headset Button Down Detected\n"); + + /* + * max98090_headset_button_event(codec) + * could be defined, then called here. + */ + + status |= SND_JACK_HEADSET; + status |= SND_JACK_BTN_0; + + break; + } + + /* Line is reported as Headphone */ + /* Nokia Headset is reported as Headphone */ + /* Mono Headphone is reported as Headphone */ + dev_info(codec->dev, "Headphone Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADPHONE; + + max98090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + status |= SND_JACK_HEADPHONE; + + break; + } + + case M98090_JKSNS_MASK: + { + dev_info(codec->dev, "Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADSET; + + max98090_set_bias_level(codec, SND_SOC_BIAS_ON); + + status |= SND_JACK_HEADSET; + + break; + } + + default: + { + dev_info(codec->dev, "Unrecognized Jack Status\n"); + break; + } + } + + snd_soc_jack_report(max98090->jack, status, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + snd_soc_dapm_sync(dapm); +} + +static irqreturn_t max98090_interrupt(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int mask; + unsigned int active; + + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); + + ret = regmap_read(max98090->regmap, M98090_REG_IRQ_ENABLE, &mask); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_IRQ_ENABLE: %d\n", + ret); + return IRQ_NONE; + } + + ret = regmap_read(max98090->regmap, M98090_REG_IRQ_STATUS, &active); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_IRQ_STATUS: %d\n", + ret); + return IRQ_NONE; + } + + dev_dbg(codec->dev, "active=0x%02x mask=0x%02x -> active=0x%02x\n", + active, mask, active & mask); + + active &= mask; + + if (!active) + return IRQ_NONE; + + if (active & M98090_IRQ_CLD_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_CLD_MASK\n"); + } + + if (active & M98090_IRQ_SLD_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_SLD_MASK\n"); + } + + if (active & M98090_IRQ_ULK_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_ULK_MASK\n"); + } + + if (active & M98090_IRQ_JDET_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_JDET_MASK\n"); + + pm_wakeup_event(codec->dev, 100); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + } + + if (active & M98090_IRQ_ALCACT_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_ALCACT_MASK\n"); + } + + if (active & M98090_IRQ_ALCCLP_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_ALCCLP_MASK\n"); + } + + return IRQ_HANDLED; +} + +/** + * max98090_mic_detect - Enable microphone detection via the MAX98090 IRQ + * + * @codec: MAX98090 codec + * @jack: jack to report detection events on + * + * Enable microphone detection via IRQ on the MAX98090. If GPIOs are + * being used to bring out signals to the processor then only platform + * data configuration is needed for MAX98090 and processor GPIOs should + * be configured using snd_soc_jack_add_gpios() instead. + * + * If no jack is supplied detection will be disabled. + */ +int max98090_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + dev_info(codec->dev, "max98090_mic_detect\n"); + + max98090->jack = jack; + if (jack) { + snd_soc_update_bits(codec, M98090_REG_IRQ_ENABLE, + M98090_IRQ_IJDET_MASK, + 1 << M98090_IRQ_IJDET_SHIFT); + } else { + snd_soc_update_bits(codec, M98090_REG_IRQ_ENABLE, + M98090_IRQ_IJDET_MASK, + 0); + } + + /* Send an initial empty report */ + snd_soc_jack_report(max98090->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + return 0; +} +EXPORT_SYMBOL_GPL(max98090_mic_detect); + +#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98090_dai_ops = { + .set_sysclk = max98090_dai_set_sysclk, + .set_fmt = max98090_dai_set_fmt, + .set_tdm_slot = max98090_set_tdm_slot, + .hw_params = max98090_dai_hw_params, + .digital_mute = max98090_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98090_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .ops = &max98090_dai_ops, +} +}; + +static void max98090_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_pdata *pdata = max98090->pdata; + + if (!pdata) { + dev_err(codec->dev, "No platform data\n"); + return; + } + +} + +static int max98090_probe(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int ret = 0; + + dev_info(codec->dev, "max98090_probe\n"); + + max98090->codec = codec; + + codec->control_data = max98090->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Reset the codec, the DSP core, and disable all interrupts */ + max98090_reset(max98090); + + /* Initialize private data */ + + max98090->sysclk = (unsigned)-1; + + cdata = &max98090->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + max98090->lin_state = 0; + max98090->mic1pre = 0; + max98090->mic2pre = 0; + max98090->extmic_mux = 0; + + ret = snd_soc_read(codec, M98090_REG_REV_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + + if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { + max98090->devtype = MAX98090; + dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); + } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { + max98090->devtype = MAX98091; + dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); + } else { + max98090->devtype = MAX98090; + dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); + } + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + + /* Enable jack detection */ + snd_soc_write(codec, M98090_REG_CFG_JACK, + M98090_JDETEN_MASK | M98090_JDEB_25MS); + + /* Register for interrupts */ + dev_dbg(codec->dev, "irq = %d\n", max98090->irq); + + ret = request_threaded_irq(max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING, + "max98090_interrupt", codec); + if (ret < 0) { + dev_err(codec->dev, "request_irq failed: %d\n", + ret); + } + + /* + * Clear any old interrupts. + * An old interrupt ocurring prior to installing the ISR + * can keep a new interrupt from generating a trigger. + */ + snd_soc_read(codec, M98090_REG_IRQ_STATUS); + + /* High Performance is default */ + snd_soc_update_bits(codec, M98090_REG_DAC_CFG, + M98090_DACHP_MASK, + 1 << M98090_DACHP_SHIFT); + snd_soc_update_bits(codec, M98090_REG_DAC_CFG, + M98090_DAC_PERFMODE_MASK, + 0 << M98090_DAC_PERFMODE_SHIFT); + snd_soc_update_bits(codec, M98090_REG_ADC_CFG, + M98090_ADCHP_MASK, + 1 << M98090_ADCHP_SHIFT); + + /* Turn on VCM bandgap reference */ + snd_soc_write(codec, M98090_REG_BIAS_CNTL, + M98090_VCM_MODE_MASK); + + max98090_handle_pdata(codec); + + max98090_add_widgets(codec); + +err_access: + return ret; +} + +static int max98090_remove(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&max98090->jack_work); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98090 = { + .probe = max98090_probe, + .remove = max98090_remove, + .set_bias_level = max98090_set_bias_level, +}; + +static const struct regmap_config max98090_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX98090_MAX_REGISTER, + .reg_defaults = max98090_reg, + .num_reg_defaults = ARRAY_SIZE(max98090_reg), + .volatile_reg = max98090_volatile_register, + .readable_reg = max98090_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98090_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98090_priv *max98090; + int ret; + + pr_info("max98090_i2c_probe\n"); + + max98090 = kzalloc(sizeof(struct max98090_priv), GFP_KERNEL); + if (max98090 == NULL) + return -ENOMEM; + + max98090->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98090); + max98090->control_data = i2c; + max98090->pdata = i2c->dev.platform_data; + max98090->irq = i2c->irq; + + max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + if (IS_ERR(max98090->regmap)) { + ret = PTR_ERR(max98090->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98090, max98090_dai, + ARRAY_SIZE(max98090_dai)); + if (ret < 0) + regmap_exit(max98090->regmap); + +err_enable: + return ret; +} + +static int __devexit max98090_i2c_remove(struct i2c_client *client) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); + snd_soc_unregister_codec(&client->dev); + regmap_exit(max98090->regmap); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static int max98090_runtime_resume(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, false); + + max98090_reset(max98090); + + regcache_sync(max98090->regmap); + + return 0; +} + +static int max98090_runtime_suspend(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, true); + + return 0; +} + +static struct dev_pm_ops max98090_pm = { + SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + max98090_runtime_resume, NULL) +}; + +static const struct i2c_device_id max98090_i2c_id[] = { + { "max98090", MAX98090 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); + +static struct i2c_driver max98090_i2c_driver = { + .driver = { + .name = "max98090", + .owner = THIS_MODULE, + .pm = &max98090_pm, + }, + .probe = max98090_i2c_probe, + .remove = __devexit_p(max98090_i2c_remove), + .id_table = max98090_i2c_id, +}; + +module_i2c_driver(max98090_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); +MODULE_AUTHOR("Peter Hsiang, Jesse Marroqin, Jerry Wong"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h new file mode 100755 index 0000000..4d48381 --- /dev/null +++ b/sound/soc/codecs/max98090.h @@ -0,0 +1,1537 @@ +/* + * max98090.h -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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 _MAX98090_H +#define _MAX98090_H + +#include <linux/version.h> + +/* One can override the Linux version here with an explicit version number */ +#define M98090_LINUX_VERSION LINUX_VERSION_CODE + +/* + * MAX98090 Register Definitions + */ + +#define M98090_REG_SW_RESET 0x00 +#define M98090_REG_IRQ_STATUS 0x01 +#define M98090_REG_JACK_STATUS 0x02 +#define M98090_REG_IRQ_ENABLE 0x03 +#define M98090_REG_QCFG_SYS_CLK 0x04 +#define M98090_REG_QCFG_RATE 0x05 +#define M98090_REG_QCFG_DAI 0x06 +#define M98090_REG_QCFG_DAC 0x07 +#define M98090_REG_QCFG_MIC_PATH 0x08 +#define M98090_REG_QCFG_LINE_PATH 0x09 +#define M98090_REG_QCFG_MIC_LOOP 0x0A +#define M98090_REG_QCFG_LINE_LOOP 0x0B +#define M98090_REG_RESERVED 0x0C +#define M98090_REG_CFG_INPUT 0x0D +#define M98090_REG_LVL_LINE 0x0E +#define M98090_REG_CFG_LINE 0x0F +#define M98090_REG_LVL_MIC1 0x10 +#define M98090_REG_LVL_MIC2 0x11 +#define M98090_REG_MIC_BIAS 0x12 +#define M98090_REG_MIC_CFG1 0x13 +#define M98090_REG_MIC_CFG2 0x14 +#define M98090_REG_MIX_ADC_L 0x15 +#define M98090_REG_MIX_ADC_R 0x16 +#define M98090_REG_LVL_ADC_L 0x17 +#define M98090_REG_LVL_ADC_R 0x18 +#define M98090_REG_LVL_BIQUAD 0x19 +#define M98090_REG_LVL_SIDETONE 0x1A +#define M98090_REG_SYS_CLOCK 0x1B +#define M98090_REG_CLOCK_MODE 0x1C +#define M98090_REG_CLOCK_DAI1_NI_HI 0x1D +#define M98090_REG_CLOCK_DAI2_NI_LO 0x1E +#define M98090_REG_CLOCK_DAI3_MI_HI 0x1F +#define M98090_REG_CLOCK_DAI4_MI_LO 0x20 +#define M98090_REG_CLOCK_MAS_MODE 0x21 +#define M98090_REG_DAI_INTERFACE_FORMAT 0x22 +#define M98090_REG_DAI_TDM_CONTROL 0x23 +#define M98090_REG_DAI_TDM_FORMAT 0x24 +#define M98090_REG_DAI_IOCFG 0x25 +#define M98090_REG_DAI_FILTERS 0x26 +#define M98090_REG_DAI_LVL 0x27 +#define M98090_REG_DAI_LVL_EQ 0x28 +#define M98090_REG_MIX_HP_LEFT 0x29 +#define M98090_REG_MIX_HP_RIGHT 0x2A +#define M98090_REG_MIX_HP_CNTL 0x2B +#define M98090_REG_LVL_HP_LEFT 0x2C +#define M98090_REG_LVL_HP_RIGHT 0x2D +#define M98090_REG_MIX_SPK_LEFT 0x2E +#define M98090_REG_MIX_SPK_RIGHT 0x2F +#define M98090_REG_MIX_SPK_CNTL 0x30 +#define M98090_REG_LVL_SPK_LEFT 0x31 +#define M98090_REG_LVL_SPK_RIGHT 0x32 +#define M98090_REG_ALC_TIMING 0x33 +#define M98090_REG_ALC_CMPR 0x34 +#define M98090_REG_ALC_EXP 0x35 +#define M98090_REG_LVL_ALC 0x36 +#define M98090_REG_MIX_RCV_LEFT 0x37 +#define M98090_REG_MIX_RCV_CTRL_LEFT 0x38 +#define M98090_REG_LVL_RCV_LEFT 0x39 +#define M98090_REG_MIX_RCV_RIGHT 0x3A +#define M98090_REG_MIX_RCV_CNTL_RIGHT 0x3B +#define M98090_REG_LVL_RCV_RIGHT 0x3C +#define M98090_REG_CFG_JACK 0x3D +#define M98090_REG_PWR_EN_IN 0x3E +#define M98090_REG_PWR_EN_OUT 0x3F +#define M98090_REG_CFG_LVL 0x40 +#define M98090_REG_DSP_EQ_EN 0x41 +#define M98090_REG_BIAS_CNTL 0x42 +#define M98090_REG_DAC_CFG 0x43 +#define M98090_REG_ADC_CFG 0x44 +#define M98090_REG_PWR_SYS 0x45 +#define M98090_REG_EQ_BASE 0x46 +#define M98090_REG_BIQUAD_BASE 0xAF +#define M98090_REG_DMIC3_VOLUME 0xBE +#define M98090_REG_DMIC4_VOLUME 0xBF +#define M98090_REG_DMIC34_BQ_PREATTEN 0xC0 +#define M98090_REG_RECORD_TDM_SLOT 0xC1 +#define M98090_REG_SAMPLE_RATE 0xC2 +#define M98090_REG_DMIC34_BIQUAD_BASE 0xC3 +#define M98090_REG_REV_ID 0xFF + +#define M98090_REG_CNT (0xFF+1) +#define MAX98090_MAX_REGISTER 0xFF + +/* MAX98090 Register Bit Fields */ + +/* + * M98090_REG_SW_RESET + */ +#define M98090_SWRESET_MASK (1<<7) +#define M98090_SWRESET_SHIFT 7 +#define M98090_SWRESET_WIDTH 1 + +/* + * M98090_REG_IRQ_STATUS + */ +#define M98090_IRQ_CLD_MASK (1<<7) +#define M98090_IRQ_CLD_SHIFT 7 +#define M98090_IRQ_CLD_WIDTH 1 +#define M98090_IRQ_SLD_MASK (1<<6) +#define M98090_IRQ_SLD_SHIFT 6 +#define M98090_IRQ_SLD_WIDTH 1 +#define M98090_IRQ_ULK_MASK (1<<5) +#define M98090_IRQ_ULK_SHIFT 5 +#define M98090_IRQ_ULK_WIDTH 1 +#define M98090_IRQ_JDET_MASK (1<<2) +#define M98090_IRQ_JDET_SHIFT 2 +#define M98090_IRQ_JDET_WIDTH 1 +#define M98090_IRQ_ALCACT_MASK (1<<1) +#define M98090_IRQ_ALCACT_SHIFT 1 +#define M98090_IRQ_ALCACT_WIDTH 1 +#define M98090_IRQ_ALCCLP_MASK (1<<0) +#define M98090_IRQ_ALCCLP_SHIFT 0 +#define M98090_IRQ_ALCCLP_WIDTH 1 + +/* + * M98090_REG_JACK_STATUS + */ +#define M98090_LSNS_MASK (1<<2) +#define M98090_LSNS_SHIFT 2 +#define M98090_LSNS_WIDTH 1 +#define M98090_JKSNS_MASK (1<<1) +#define M98090_JKSNS_SHIFT 1 +#define M98090_JKSNS_WIDTH 1 + +/* + * M98090_REG_IRQ_ENABLE + */ +#define M98090_IRQ_ICLD_MASK (1<<7) +#define M98090_IRQ_ICLD_SHIFT 7 +#define M98090_IRQ_ICLD_WIDTH 1 +#define M98090_IRQ_ISLD_MASK (1<<6) +#define M98090_IRQ_ISLD_SHIFT 6 +#define M98090_IRQ_ISLD_WIDTH 1 +#define M98090_IRQ_IULK_MASK (1<<5) +#define M98090_IRQ_IULK_SHIFT 5 +#define M98090_IRQ_IULK_WIDTH 1 +#define M98090_IRQ_IJDET_MASK (1<<2) +#define M98090_IRQ_IJDET_SHIFT 2 +#define M98090_IRQ_IJDET_WIDTH 1 +#define M98090_IRQ_IALCACT_MASK (1<<1) +#define M98090_IRQ_IALCACT_SHIFT 1 +#define M98090_IRQ_IALCACT_WIDTH 1 +#define M98090_IRQ_IALCCLP_MASK (1<<0) +#define M98090_IRQ_IALCCLP_SHIFT 0 +#define M98090_IRQ_IALCCLP_WIDTH 1 + +/* + * M98090_REG_QCFG_SYS_CLK + */ +#define M98090_CLK_26M_MASK (1<<7) +#define M98090_CLK_26M_SHIFT 7 +#define M98090_CLK_26M_WIDTH 1 +#define M98090_CLK_19P2M_MASK (1<<6) +#define M98090_CLK_19P2M_SHIFT 6 +#define M98090_CLK_19P2M_WIDTH 1 +#define M98090_CLK_13M_MASK (1<<5) +#define M98090_CLK_13M_SHIFT 5 +#define M98090_CLK_13M_WIDTH 1 +#define M98090_CLK_12P288M_MASK (1<<4) +#define M98090_CLK_12P288M_SHIFT 4 +#define M98090_CLK_12P288M_WIDTH 1 +#define M98090_CLK_12M_MASK (1<<3) +#define M98090_CLK_12M_SHIFT 3 +#define M98090_CLK_12M_WIDTH 1 +#define M98090_CLK_11P2896M_MASK (1<<2) +#define M98090_CLK_11P2896M_SHIFT 2 +#define M98090_CLK_11P2896M_WIDTH 1 +#define M98090_CLK_256FS_MASK (1<<0) +#define M98090_CLK_256FS_SHIFT 0 +#define M98090_CLK_256FS_WIDTH 1 +#define M98090_CLK_ALL_SHIFT 0 +#define M98090_CLK_ALL_WIDTH 8 +#define M98090_CLK_ALL_NUM (1<<M98090_CLK_ALL_WIDTH) + +/* + * M98090_REG_QCFG_RATE + */ +#define M98090_SR_96K_MASK (1<<5) +#define M98090_SR_96K_SHIFT 5 +#define M98090_SR_96K_WIDTH 1 +#define M98090_SR_32K_MASK (1<<4) +#define M98090_SR_32K_SHIFT 4 +#define M98090_SR_32K_WIDTH 1 +#define M98090_SR_48K_MASK (1<<3) +#define M98090_SR_48K_SHIFT 3 +#define M98090_SR_48K_WIDTH 1 +#define M98090_SR_44K1_MASK (1<<2) +#define M98090_SR_44K1_SHIFT 2 +#define M98090_SR_44K1_WIDTH 1 +#define M98090_SR_16K_MASK (1<<1) +#define M98090_SR_16K_SHIFT 1 +#define M98090_SR_16K_WIDTH 1 +#define M98090_SR_8K_MASK (1<<0) +#define M98090_SR_8K_SHIFT 0 +#define M98090_SR_8K_WIDTH 1 +#define M98090_SR_MASK 0x3F +#define M98090_SR_ALL_SHIFT 0 +#define M98090_SR_ALL_WIDTH 8 +#define M98090_SR_ALL_NUM (1<<M98090_SR_ALL_WIDTH) + +/* + * M98090_REG_QCFG_DAI + */ +#define M98090_DAI_RJ_M_MASK (1<<5) +#define M98090_DAI_RJ_M_SHIFT 5 +#define M98090_DAI_RJ_M_WIDTH 1 +#define M98090_DAI_RJ_S_MASK (1<<4) +#define M98090_DAI_RJ_S_SHIFT 4 +#define M98090_DAI_RJ_S_WIDTH 1 +#define M98090_DAI_LJ_M_MASK (1<<3) +#define M98090_DAI_LJ_M_SHIFT 3 +#define M98090_DAI_LJ_M_WIDTH 1 +#define M98090_DAI_LJ_S_MASK (1<<2) +#define M98090_DAI_LJ_S_SHIFT 2 +#define M98090_DAI_LJ_S_WIDTH 1 +#define M98090_DAI_I2S_M_MASK (1<<1) +#define M98090_DAI_I2S_M_SHIFT 1 +#define M98090_DAI_I2S_M_WIDTH 1 +#define M98090_DAI_I2S_S_MASK (1<<0) +#define M98090_DAI_I2S_S_SHIFT 0 +#define M98090_DAI_I2S_S_WIDTH 1 +#define M98090_DAI_ALL_SHIFT 0 +#define M98090_DAI_ALL_WIDTH 8 +#define M98090_DAI_ALL_NUM (1<<M98090_DAI_ALL_WIDTH) + +/* + * M98090_REG_QCFG_DAC + */ +#define M98090_DIG2_HP_MASK (1<<7) +#define M98090_DIG2_HP_SHIFT 7 +#define M98090_DIG2_HP_WIDTH 1 +#define M98090_DIG2_EAR_MASK (1<<6) +#define M98090_DIG2_EAR_SHIFT 6 +#define M98090_DIG2_EAR_WIDTH 1 +#define M98090_DIG2_SPK_MASK (1<<5) +#define M98090_DIG2_SPK_SHIFT 5 +#define M98090_DIG2_SPK_WIDTH 1 +#define M98090_DIG2_LOUT_MASK (1<<4) +#define M98090_DIG2_LOUT_SHIFT 4 +#define M98090_DIG2_LOUT_WIDTH 1 +#define M98090_DIG2_ALL_SHIFT 0 +#define M98090_DIG2_ALL_WIDTH 8 +#define M98090_DIG2_ALL_NUM (1<<M98090_DIG2_ALL_WIDTH) + +/* + * M98090_REG_QCFG_MIC_PATH + */ +#define M98090_IN12_MIC1_MASK (1<<7) +#define M98090_IN12_MIC1_SHIFT 7 +#define M98090_IN12_MIC1_WIDTH 1 +#define M98090_IN34_MIC2_MASK (1<<6) +#define M98090_IN34_MIC2_SHIFT 6 +#define M98090_IN34_MIC2_WIDTH 1 +#define M98090_IN56_MIC1_MASK (1<<5) +#define M98090_IN56_MIC1_SHIFT 5 +#define M98090_IN56_MIC1_WIDTH 1 +#define M98090_IN56_MIC2_MASK (1<<4) +#define M98090_IN56_MIC2_SHIFT 4 +#define M98090_IN56_MIC2_WIDTH 1 +#define M98090_IN12_DADC_MASK (1<<3) +#define M98090_IN12_DADC_SHIFT 3 +#define M98090_IN12_DADC_WIDTH 1 +#define M98090_IN34_DADC_MASK (1<<2) +#define M98090_IN34_DADC_SHIFT 2 +#define M98090_IN34_DADC_WIDTH 1 +#define M98090_IN56_DADC_MASK (1<<1) +#define M98090_IN56_DADC_SHIFT 1 +#define M98090_IN56_DADC_WIDTH 1 +#define M98090_MIC_ALL_SHIFT 0 +#define M98090_MIC_ALL_WIDTH 8 +#define M98090_MIC_ALL_NUM (1<<M98090_MIC_ALL_WIDTH) + +/* + * M98090_REG_QCFG_LINE_PATH + */ +#define M98090_IN12S_AB_MASK (1<<7) +#define M98090_IN12S_AB_SHIFT 7 +#define M98090_IN12S_AB_WIDTH 1 +#define M98090_IN34S_AB_MASK (1<<6) +#define M98090_IN34S_AB_SHIFT 6 +#define M98090_IN34S_AB_WIDTH 1 +#define M98090_IN56S_AB_MASK (1<<5) +#define M98090_IN56S_AB_SHIFT 5 +#define M98090_IN56S_AB_WIDTH 1 +#define M98090_IN34D_A_MASK (1<<4) +#define M98090_IN34D_A_SHIFT 4 +#define M98090_IN34D_A_WIDTH 1 +#define M98090_IN56D_B_MASK (1<<3) +#define M98090_IN56D_B_SHIFT 3 +#define M98090_IN56D_B_WIDTH 1 +#define M98090_LINE_ALL_SHIFT 0 +#define M98090_LINE_ALL_WIDTH 8 +#define M98090_LINE_ALL_NUM (1<<M98090_LINE_ALL_WIDTH) + +/* + * M98090_REG_QCFG_MIC_LOOP + */ +#define M98090_IN12_M1HPL_MASK (1<<7) +#define M98090_IN12_M1HPL_SHIFT 7 +#define M98090_IN12_M1HPL_WIDTH 1 +#define M98090_IN12_M1SPKL_MASK (1<<6) +#define M98090_IN12_M1SPKL_SHIFT 6 +#define M98090_IN12_M1SPKL_WIDTH 1 +#define M98090_IN12_M1EAR_MASK (1<<5) +#define M98090_IN12_M1EAR_SHIFT 5 +#define M98090_IN12_M1EAR_WIDTH 1 +#define M98090_IN12_M1LOUTL_MASK (1<<4) +#define M98090_IN12_M1LOUTL_SHIFT 4 +#define M98090_IN12_M1LOUTL_WIDTH 1 +#define M98090_IN34_M2HPR_MASK (1<<3) +#define M98090_IN34_M2HPR_SHIFT 3 +#define M98090_IN34_M2HPR_WIDTH 1 +#define M98090_IN34_M2SPKR_MASK (1<<2) +#define M98090_IN34_M2SPKR_SHIFT 2 +#define M98090_IN34_M2SPKR_WIDTH 1 +#define M98090_IN34_M2EAR_MASK (1<<1) +#define M98090_IN34_M2EAR_SHIFT 1 +#define M98090_IN34_M2EAR_WIDTH 1 +#define M98090_IN34_M2LOUTR_MASK (1<<0) +#define M98090_IN34_M2LOUTR_SHIFT 0 +#define M98090_IN34_M2LOUTR_WIDTH 1 +#define M98090_AMIC_ALL_SHIFT 0 +#define M98090_AMIC_ALL_WIDTH 8 +#define M98090_AMIC_ALL_NUM (1<<M98090_AMIC_ALL_WIDTH) + +/* + * M98090_REG_QCFG_LINE_LOOP + */ +#define M98090_IN12S_ABHP_MASK (1<<7) +#define M98090_IN12S_ABHP_SHIFT 7 +#define M98090_IN12S_ABHP_WIDTH 1 +#define M98090_IN34D_ASPKL_MASK (1<<6) +#define M98090_IN34D_ASPKL_SHIFT 6 +#define M98090_IN34D_ASPKL_WIDTH 1 +#define M98090_IN34D_AEAR_MASK (1<<5) +#define M98090_IN34D_AEAR_SHIFT 5 +#define M98090_IN34D_AEAR_WIDTH 1 +#define M98090_IN12S_ABLOUT_MASK (1<<4) +#define M98090_IN12S_ABLOUT_SHIFT 4 +#define M98090_IN12S_ABLOUT_WIDTH 1 +#define M98090_IN34S_ABHP_MASK (1<<3) +#define M98090_IN34S_ABHP_SHIFT 3 +#define M98090_IN34S_ABHP_WIDTH 1 +#define M98090_IN56D_BSPKR_MASK (1<<2) +#define M98090_IN56D_BSPKR_SHIFT 2 +#define M98090_IN56D_BSPKR_WIDTH 1 +#define M98090_IN56D_BEAR_MASK (1<<1) +#define M98090_IN56D_BEAR_SHIFT 1 +#define M98090_IN56D_BEAR_WIDTH 1 +#define M98090_IN34S_ABLOUT_MASK (1<<0) +#define M98090_IN34S_ABLOUT_SHIFT 0 +#define M98090_IN34S_ABLOUT_WIDTH 1 +#define M98090_ALIN_ALL_SHIFT 0 +#define M98090_ALIN_ALL_WIDTH 8 +#define M98090_ALIN_ALL_NUM (1<<M98090_ALIN_ALL_WIDTH) + +/* + * M98090_REG_RESERVED + */ + +/* + * M98090_REG_CFG_INPUT + */ +#define M98090_IN34DIFF_MASK (1<<7) +#define M98090_IN34DIFF_SHIFT 7 +#define M98090_IN34DIFF_WIDTH 1 +#define M98090_IN56DIFF_MASK (1<<6) +#define M98090_IN56DIFF_SHIFT 6 +#define M98090_IN56DIFF_WIDTH 1 +#define M98090_IN1SEEN_MASK (1<<5) +#define M98090_IN1SEEN_SHIFT 5 +#define M98090_IN1SEEN_WIDTH 1 +#define M98090_IN2SEEN_MASK (1<<4) +#define M98090_IN2SEEN_SHIFT 4 +#define M98090_IN2SEEN_WIDTH 1 +#define M98090_IN3SEEN_MASK (1<<3) +#define M98090_IN3SEEN_SHIFT 3 +#define M98090_IN3SEEN_WIDTH 1 +#define M98090_IN4SEEN_MASK (1<<2) +#define M98090_IN4SEEN_SHIFT 2 +#define M98090_IN4SEEN_WIDTH 1 +#define M98090_IN5SEEN_MASK (1<<1) +#define M98090_IN5SEEN_SHIFT 1 +#define M98090_IN5SEEN_WIDTH 1 +#define M98090_IN6SEEN_MASK (1<<0) +#define M98090_IN6SEEN_SHIFT 0 +#define M98090_IN6SEEN_WIDTH 1 + +/* + * M98090_REG_LVL_LINE + */ +#define M98090_MIXG135_MASK (1<<7) +#define M98090_MIXG135_SHIFT 7 +#define M98090_MIXG135_WIDTH 1 +#define M98090_MIXT246_MASK (1<<6) +#define M98090_MIXT246_SHIFT 6 +#define M98090_MIXT246_WIDTH 1 +#define M98090_LINAPGA_MASK (7<<3) +#define M98090_LINAPGA_SHIFT 3 +#define M98090_LINAPGA_WIDTH 3 +#define M98090_LINAPGA_NUM 6 +#define M98090_LINBPGA_MASK (7<<0) +#define M98090_LINBPGA_SHIFT 0 +#define M98090_LINBPGA_WIDTH 3 +#define M98090_LINBPGA_NUM 6 + +/* + * M98090_REG_CFG_LINE + */ +#define M98090_EXTBUFA_MASK (1<<7) +#define M98090_EXTBUFA_SHIFT 7 +#define M98090_EXTBUFA_WIDTH 1 +#define M98090_EXTBUFA_NUM (1<<M98090_EXTBUFA_WIDTH) +#define M98090_EXTBUFB_MASK (1<<6) +#define M98090_EXTBUFB_SHIFT 6 +#define M98090_EXTBUFB_WIDTH 1 +#define M98090_EXTBUFB_NUM (1<<M98090_EXTBUFB_WIDTH) +#define M98090_EXTMIC_MASK (3<<0) +#define M98090_EXTMIC_SHIFT 0 +#define M98090_EXTMIC1_SHIFT 0 +#define M98090_EXTMIC2_SHIFT 1 +#define M98090_EXTMIC_WIDTH 2 +#define M98090_EXTMIC_NONE (0<<0) +#define M98090_EXTMIC_MIC1 (1<<0) +#define M98090_EXTMIC_MIC2 (2<<0) + +/* + * M98090_REG_LVL_MIC1 + */ +#define M98090_MIC_PA1EN_MASK (3<<5) +#define M98090_MIC_PA1EN_SHIFT 5 +#define M98090_MIC_PA1EN_WIDTH 2 +#define M98090_MIC_PA1EN_NUM 3 +#define M98090_MIC_PGAM1_MASK (31<<0) +#define M98090_MIC_PGAM1_SHIFT 0 +#define M98090_MIC_PGAM1_WIDTH 5 +#define M98090_MIC_PGAM1_NUM 21 + +/* + * M98090_REG_LVL_MIC2 + */ +#define M98090_MIC_PA2EN_MASK (3<<5) +#define M98090_MIC_PA2EN_SHIFT 5 +#define M98090_MIC_PA2EN_WIDTH 2 +#define M98090_MIC_PA2EN_NUM 3 +#define M98090_MIC_PGAM2_MASK (31<<0) +#define M98090_MIC_PGAM2_SHIFT 0 +#define M98090_MIC_PGAM2_WIDTH 5 +#define M98090_MIC_PGAM2_NUM 21 + +/* + * M98090_REG_MIC_BIAS + */ +#define M98090_MBVSEL_MASK (3<<0) +#define M98090_MBVSEL_SHIFT 0 +#define M98090_MBVSEL_WIDTH 2 +#define M98090_MBVSEL_2V8 (3<<0) +#define M98090_MBVSEL_2V55 (2<<0) +#define M98090_MBVSEL_2V4 (1<<0) +#define M98090_MBVSEL_2V2 (0<<0) + +/* + * M98090_REG_MIC_CFG1 + */ +#define M98090_MICCLK_MASK (7<<4) +#define M98090_MICCLK_SHIFT 4 +#define M98090_MICCLK_WIDTH 3 +#define M98090_DIGMIC4_MASK (1<<3) +#define M98090_DIGMIC4_SHIFT 3 +#define M98090_DIGMIC4_WIDTH 1 +#define M98090_DIGMIC4_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DIGMIC3_MASK (1<<2) +#define M98090_DIGMIC3_SHIFT 2 +#define M98090_DIGMIC3_WIDTH 1 +#define M98090_DIGMIC3_NUM (1<<M98090_DIGMIC3_WIDTH) +#define M98090_DIGMICR_MASK (1<<1) +#define M98090_DIGMICR_SHIFT 1 +#define M98090_DIGMICR_WIDTH 1 +#define M98090_DIGMICR_NUM (1<<M98090_DIGMICR_WIDTH) +#define M98090_DIGMICL_MASK (1<<0) +#define M98090_DIGMICL_SHIFT 0 +#define M98090_DIGMICL_WIDTH 1 +#define M98090_DIGMICL_NUM (1<<M98090_DIGMICL_WIDTH) + +/* + * M98090_REG_MIC_CFG2 + */ +#define M98090_DMIC_COMP_MASK (15<<4) +#define M98090_DMIC_COMP_SHIFT 4 +#define M98090_DMIC_COMP_WIDTH 4 +#define M98090_DMIC_COMP_NUM (1<<M98090_DMIC_COMP_WIDTH) +#define M98090_DMIC_FREQ_MASK (3<<0) +#define M98090_DMIC_FREQ_SHIFT 0 +#define M98090_DMIC_FREQ_WIDTH 2 + +/* + * M98090_REG_MIX_ADC_L + */ +#define M98090_MIXADL_MIC2_MASK (1<<6) +#define M98090_MIXADL_MIC2_SHIFT 6 +#define M98090_MIXADL_MIC2_WIDTH 1 +#define M98090_MIXADL_MIC1_MASK (1<<5) +#define M98090_MIXADL_MIC1_SHIFT 5 +#define M98090_MIXADL_MIC1_WIDTH 1 +#define M98090_MIXADL_LINEB_MASK (1<<4) +#define M98090_MIXADL_LINEB_SHIFT 4 +#define M98090_MIXADL_LINEB_WIDTH 1 +#define M98090_MIXADL_LINEA_MASK (1<<3) +#define M98090_MIXADL_LINEA_SHIFT 3 +#define M98090_MIXADL_LINEA_WIDTH 1 +#define M98090_MIXADL_IN65DIFF_MASK (1<<2) +#define M98090_MIXADL_IN65DIFF_SHIFT 2 +#define M98090_MIXADL_IN65DIFF_WIDTH 1 +#define M98090_MIXADL_IN34DIFF_MASK (1<<1) +#define M98090_MIXADL_IN34DIFF_SHIFT 1 +#define M98090_MIXADL_IN34DIFF_WIDTH 1 +#define M98090_MIXADL_IN12DIFF_MASK (1<<0) +#define M98090_MIXADL_IN12DIFF_SHIFT 0 +#define M98090_MIXADL_IN12DIFF_WIDTH 1 +#define M98090_MIXADL_MASK (255<<0) +#define M98090_MIXADL_SHIFT 0 +#define M98090_MIXADL_WIDTH 8 + +/* + * M98090_REG_MIX_ADC_R + */ +#define M98090_MIXADR_MIC2_MASK (1<<6) +#define M98090_MIXADR_MIC2_SHIFT 6 +#define M98090_MIXADR_MIC2_WIDTH 1 +#define M98090_MIXADR_MIC1_MASK (1<<5) +#define M98090_MIXADR_MIC1_SHIFT 5 +#define M98090_MIXADR_MIC1_WIDTH 1 +#define M98090_MIXADR_LINEB_MASK (1<<4) +#define M98090_MIXADR_LINEB_SHIFT 4 +#define M98090_MIXADR_LINEB_WIDTH 1 +#define M98090_MIXADR_LINEA_MASK (1<<3) +#define M98090_MIXADR_LINEA_SHIFT 3 +#define M98090_MIXADR_LINEA_WIDTH 1 +#define M98090_MIXADR_IN65DIFF_MASK (1<<2) +#define M98090_MIXADR_IN65DIFF_SHIFT 2 +#define M98090_MIXADR_IN65DIFF_WIDTH 1 +#define M98090_MIXADR_IN34DIFF_MASK (1<<1) +#define M98090_MIXADR_IN34DIFF_SHIFT 1 +#define M98090_MIXADR_IN34DIFF_WIDTH 1 +#define M98090_MIXADR_IN12DIFF_MASK (1<<0) +#define M98090_MIXADR_IN12DIFF_SHIFT 0 +#define M98090_MIXADR_IN12DIFF_WIDTH 1 +#define M98090_MIXADR_MASK (255<<0) +#define M98090_MIXADR_SHIFT 0 +#define M98090_MIXADR_WIDTH 8 + +/* + * M98090_REG_LVL_ADC_L + */ +#define M98090_ADC_AVLG_MASK (7<<4) +#define M98090_ADC_AVLG_SHIFT 4 +#define M98090_ADC_AVLG_WIDTH 3 +#define M98090_ADC_AVLG_NUM (1<<M98090_ADC_AVLG_WIDTH) +#define M98090_ADC_AVL_MASK (15<<0) +#define M98090_ADC_AVL_SHIFT 0 +#define M98090_ADC_AVL_WIDTH 4 +#define M98090_ADC_AVL_NUM (1<<M98090_ADC_AVL_WIDTH) + +/* + * M98090_REG_LVL_ADC_R + */ +#define M98090_ADC_AVRG_MASK (7<<4) +#define M98090_ADC_AVRG_SHIFT 4 +#define M98090_ADC_AVRG_WIDTH 3 +#define M98090_ADC_AVRG_NUM (1<<M98090_ADC_AVRG_WIDTH) +#define M98090_ADC_AVR_MASK (15<<0) +#define M98090_ADC_AVR_SHIFT 0 +#define M98090_ADC_AVR_WIDTH 4 +#define M98090_ADC_AVR_NUM (1<<M98090_ADC_AVR_WIDTH) + +/* + * M98090_REG_LVL_BIQUAD + */ +#define M98090_AVBQ_MASK (15<<0) +#define M98090_AVBQ_SHIFT 0 +#define M98090_AVBQ_WIDTH 4 +#define M98090_AVBQ_NUM (1<<M98090_AVBQ_WIDTH) + +/* + * M98090_REG_LVL_SIDETONE + */ +#define M98090_DSTS_MASK (3<<6) +#define M98090_DSTS_SHIFT 6 +#define M98090_DSTS_WIDTH 2 +#define M98090_DVST_MASK (31<<0) +#define M98090_DVST_SHIFT 0 +#define M98090_DVST_WIDTH 5 +#define M98090_DVST_NUM 31 + +/* + * M98090_REG_SYS_CLOCK + */ +#define M98090_CLK_PSCLK_MASK (3<<4) +#define M98090_CLK_PSCLK_SHIFT 4 +#define M98090_CLK_PSCLK_WIDTH 2 +#define M98090_CLK_PSCLK_DISABLED (0<<4) +#define M98090_CLK_PSCLK_DIV1 (1<<4) +#define M98090_CLK_PSCLK_DIV2 (2<<4) +#define M98090_CLK_PSCLK_DIV4 (3<<4) + +/* + * M98090_REG_CLOCK_MODE + */ +#define M98090_CLK_FREQ1_MASK (15<<4) +#define M98090_CLK_FREQ1_SHIFT 4 +#define M98090_CLK_FREQ1_WIDTH 4 +#define M98090_CLK_USE_M1_MASK (1<<0) +#define M98090_CLK_USE_M1_SHIFT 0 +#define M98090_CLK_USE_M1_WIDTH 1 +#define M98090_CLK_USE_M1_NUM (1<<M98090_CLK_USE_M1_WIDTH) + +/* + * M98090_REG_CLOCK_DAI1_NI_HI + */ +#define M98090_CLK_NI_HI_MASK (127<<0) +#define M98090_CLK_NI_HI_SHIFT 0 +#define M98090_CLK_NI_HI_WIDTH 7 +#define M98090_CLK_NI_HI_NUM (1<<M98090_CLK_NI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_DAI2_NI_LO + */ +#define M98090_CLK_NI_LO_MASK (255<<0) +#define M98090_CLK_NI_LO_SHIFT 0 +#define M98090_CLK_NI_LO_WIDTH 8 +#define M98090_CLK_NI_LO_NUM (1<<M98090_CLK_NI_LO_WIDTH) + +/* + * M98090_REG_CLOCK_DAI3_MI_HI + */ +#define M98090_CLK_MI_HI_MASK (255<<0) +#define M98090_CLK_MI_HI_SHIFT 0 +#define M98090_CLK_MI_HI_WIDTH 8 +#define M98090_CLK_MI_HI_NUM (1<<M98090_CLK_MI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_DAI4_MI_LO + */ +#define M98090_CLK_MI_LO_MASK (255<<0) +#define M98090_CLK_MI_LO_SHIFT 0 +#define M98090_CLK_MI_LO_WIDTH 8 +#define M98090_CLK_MI_LO_NUM (1<<M98090_CLK_MI_LO_WIDTH) + +/* + * M98090_REG_CLOCK_MAS_MODE + */ +#define M98090_CLK_MAS_MASK (1<<7) +#define M98090_CLK_MAS_SHIFT 7 +#define M98090_CLK_MAS_WIDTH 1 +#define M98090_CLK_BSEL_MASK (1<<0) +#define M98090_CLK_BSEL_SHIFT 0 +#define M98090_CLK_BSEL_WIDTH 1 +#define M98090_CLK_BSEL_32 (1<<0) +#define M98090_CLK_BSEL_48 (2<<0) +#define M98090_CLK_BSEL_64 (3<<0) + +/* + * M98090_REG_DAI_INTERFACE_FORMAT + */ +#define M98090_DAI_RJ_MASK (1<<5) +#define M98090_DAI_RJ_SHIFT 5 +#define M98090_DAI_RJ_WIDTH 1 +#define M98090_DAI_WCI_MASK (1<<4) +#define M98090_DAI_WCI_SHIFT 4 +#define M98090_DAI_WCI_WIDTH 1 +#define M98090_DAI_BCI_MASK (1<<3) +#define M98090_DAI_BCI_SHIFT 3 +#define M98090_DAI_BCI_WIDTH 1 +#define M98090_DAI_DLY_MASK (1<<2) +#define M98090_DAI_DLY_SHIFT 2 +#define M98090_DAI_DLY_WIDTH 1 +#define M98090_DAI_WS_MASK (3<<0) +#define M98090_DAI_WS_SHIFT 0 +#define M98090_DAI_WS_WIDTH 2 +#define M98090_DAI_WS_NUM (1<<M98090_DAI_WS_WIDTH) + +/* + * M98090_REG_DAI_TDM_CONTROL + */ +#define M98090_DAI_FSW_MASK (1<<1) +#define M98090_DAI_FSW_SHIFT 1 +#define M98090_DAI_FSW_WIDTH 1 +#define M98090_DAI_TDM_MASK (1<<0) +#define M98090_DAI_TDM_SHIFT 0 +#define M98090_DAI_TDM_WIDTH 1 +#define M98090_DAI_TDM_NUM (1<<M98090_DAI_TDM_WIDTH) + +/* + * M98090_REG_DAI_TDM_FORMAT + */ +#define M98090_DAI_TDM_SLOTL_MASK (3<<6) +#define M98090_DAI_TDM_SLOTL_SHIFT 6 +#define M98090_DAI_TDM_SLOTL_WIDTH 2 +#define M98090_DAI_TDM_SLOTL_NUM (1<<M98090_DAI_TDM_SLOTL_WIDTH) +#define M98090_DAI_TDM_SLOTR_MASK (3<<4) +#define M98090_DAI_TDM_SLOTR_SHIFT 4 +#define M98090_DAI_TDM_SLOTR_WIDTH 2 +#define M98090_DAI_TDM_SLOTR_NUM (1<<M98090_DAI_TDM_SLOTR_WIDTH) +#define M98090_DAI_TDM_SLOTDLY_MASK (15<<0) +#define M98090_DAI_TDM_SLOTDLY_SHIFT 0 +#define M98090_DAI_TDM_SLOTDLY_WIDTH 4 +#define M98090_DAI_TDM_SLOTDLY_NUM (1<<M98090_DAI_TDM_SLOTDLY_WIDTH) + +/* + * M98090_REG_DAI_IOCFG + */ +#define M98090_DAI_LTEN_MASK (1<<5) +#define M98090_DAI_LTEN_SHIFT 5 +#define M98090_DAI_LTEN_WIDTH 1 +#define M98090_DAI_LTEN_NUM (1<<M98090_DAI_LTEN_WIDTH) +#define M98090_DAI_LBEN_MASK (1<<4) +#define M98090_DAI_LBEN_SHIFT 4 +#define M98090_DAI_LBEN_WIDTH 1 +#define M98090_DAI_LBEN_NUM (1<<M98090_DAI_LBEN_WIDTH) +#define M98090_DAI_DMONO_MASK (1<<3) +#define M98090_DAI_DMONO_SHIFT 3 +#define M98090_DAI_DMONO_WIDTH 1 +#define M98090_DAI_DMONO_NUM (1<<M98090_DAI_DMONO_WIDTH) +#define M98090_DAI_HIZOFF_MASK (1<<2) +#define M98090_DAI_HIZOFF_SHIFT 2 +#define M98090_DAI_HIZOFF_WIDTH 1 +#define M98090_DAI_SDOEN_MASK (1<<1) +#define M98090_DAI_SDOEN_SHIFT 1 +#define M98090_DAI_SDOEN_WIDTH 1 +#define M98090_DAI_SDIEN_MASK (1<<0) +#define M98090_DAI_SDIEN_SHIFT 0 +#define M98090_DAI_SDIEN_WIDTH 1 + +/* + * M98090_REG_DAI_FILTERS + */ +#define M98090_DAI_FLT_MODE_MASK (1<<7) +#define M98090_DAI_FLT_MODE_SHIFT 7 +#define M98090_DAI_FLT_MODE_WIDTH 1 +#define M98090_DAI_FLT_AHPF_MASK (1<<6) +#define M98090_DAI_FLT_AHPF_SHIFT 6 +#define M98090_DAI_FLT_AHPF_WIDTH 1 +#define M98090_DAI_FLT_AHPF_NUM (1<<M98090_DAI_FLT_AHPF_WIDTH) +#define M98090_DAI_FLT_DHPF_MASK (1<<5) +#define M98090_DAI_FLT_DHPF_SHIFT 5 +#define M98090_DAI_FLT_DHPF_WIDTH 1 +#define M98090_DAI_FLT_DHPF_NUM (1<<M98090_DAI_FLT_DHPF_WIDTH) +#define M98090_DAI_FLT_DHF_MASK (1<<4) +#define M98090_DAI_FLT_DHF_SHIFT 4 +#define M98090_DAI_FLT_DHF_WIDTH 1 +#define M98090_DAI_FLT_DMIC34MODE_MASK (1<<3) +#define M98090_DAI_FLT_DMIC34MODE_SHIFT 3 +#define M98090_DAI_FLT_DMIC34MODE_WIDTH 1 +#define M98090_DAI_FLT_DMIC34HPF_MASK (1<<2) +#define M98090_DAI_FLT_DMIC34HPF_SHIFT 2 +#define M98090_DAI_FLT_DMIC34HPF_WIDTH 1 +#define M98090_DAI_FLT_DMIC34HPF_NUM (1<<M98090_DAI_FLT_DMIC34HPF_WIDTH) + +/* + * M98090_REG_DAI_LVL + */ +#define M98090_DAI_LVL_DVM_MASK (1<<7) +#define M98090_DAI_LVL_DVM_SHIFT 7 +#define M98090_DAI_LVL_DVM_WIDTH 1 +#define M98090_DAI_LVL_DVG_MASK (3<<4) +#define M98090_DAI_LVL_DVG_SHIFT 4 +#define M98090_DAI_LVL_DVG_WIDTH 2 +#define M98090_DAI_LVL_DVG_NUM (1<<M98090_DAI_LVL_DVG_WIDTH) +#define M98090_DAI_LVL_DV_MASK (15<<0) +#define M98090_DAI_LVL_DV_SHIFT 0 +#define M98090_DAI_LVL_DV_WIDTH 4 +#define M98090_DAI_LVL_DV_NUM (1<<M98090_DAI_LVL_DV_WIDTH) + +/* + * M98090_REG_DAI_LVL_EQ + */ +#define M98090_DAI_LVL_EQCLPN_MASK (1<<4) +#define M98090_DAI_LVL_EQCLPN_SHIFT 4 +#define M98090_DAI_LVL_EQCLPN_WIDTH 1 +#define M98090_DAI_LVL_DVEQ_MASK (15<<0) +#define M98090_DAI_LVL_DVEQ_SHIFT 0 +#define M98090_DAI_LVL_DVEQ_WIDTH 4 +#define M98090_DAI_LVL_DVEQ_NUM (1<<M98090_DAI_LVL_DVEQ_WIDTH) + +/* + * M98090_REG_MIX_HP_LEFT + */ +#define M98090_MIXHPL_MIC2_MASK (1<<5) +#define M98090_MIXHPL_MIC2_SHIFT 5 +#define M98090_MIXHPL_MIC2_WIDTH 1 +#define M98090_MIXHPL_MIC1_MASK (1<<4) +#define M98090_MIXHPL_MIC1_SHIFT 4 +#define M98090_MIXHPL_MIC1_WIDTH 1 +#define M98090_MIXHPL_LINEB_MASK (1<<3) +#define M98090_MIXHPL_LINEB_SHIFT 3 +#define M98090_MIXHPL_LINEB_WIDTH 1 +#define M98090_MIXHPL_LINEA_MASK (1<<2) +#define M98090_MIXHPL_LINEA_SHIFT 2 +#define M98090_MIXHPL_LINEA_WIDTH 1 +#define M98090_MIXHPL_DACR_MASK (1<<1) +#define M98090_MIXHPL_DACR_SHIFT 1 +#define M98090_MIXHPL_DACR_WIDTH 1 +#define M98090_MIXHPL_DACL_MASK (1<<0) +#define M98090_MIXHPL_DACL_SHIFT 0 +#define M98090_MIXHPL_DACL_WIDTH 1 +#define M98090_MIXHPL_MASK (63<<0) +#define M98090_MIXHPL_SHIFT 0 +#define M98090_MIXHPL_WIDTH 6 + +/* + * M98090_REG_MIX_HP_RIGHT + */ +#define M98090_MIXHPR_MIC2_MASK (1<<5) +#define M98090_MIXHPR_MIC2_SHIFT 5 +#define M98090_MIXHPR_MIC2_WIDTH 1 +#define M98090_MIXHPR_MIC1_MASK (1<<4) +#define M98090_MIXHPR_MIC1_SHIFT 4 +#define M98090_MIXHPR_MIC1_WIDTH 1 +#define M98090_MIXHPR_LINEB_MASK (1<<3) +#define M98090_MIXHPR_LINEB_SHIFT 3 +#define M98090_MIXHPR_LINEB_WIDTH 1 +#define M98090_MIXHPR_LINEA_MASK (1<<2) +#define M98090_MIXHPR_LINEA_SHIFT 2 +#define M98090_MIXHPR_LINEA_WIDTH 1 +#define M98090_MIXHPR_DACR_MASK (1<<1) +#define M98090_MIXHPR_DACR_SHIFT 1 +#define M98090_MIXHPR_DACR_WIDTH 1 +#define M98090_MIXHPR_DACL_MASK (1<<0) +#define M98090_MIXHPR_DACL_SHIFT 0 +#define M98090_MIXHPR_DACL_WIDTH 1 +#define M98090_MIXHPR_MASK (63<<0) +#define M98090_MIXHPR_SHIFT 0 +#define M98090_MIXHPR_WIDTH 6 + +/* + * M98090_REG_MIX_HP_CNTL + */ +#define M98090_MIXHPRSEL_MASK (1<<5) +#define M98090_MIXHPRSEL_SHIFT 5 +#define M98090_MIXHPRSEL_WIDTH 1 +#define M98090_MIXHPLSEL_MASK (1<<4) +#define M98090_MIXHPLSEL_SHIFT 4 +#define M98090_MIXHPLSEL_WIDTH 1 +#define M98090_MIXHPRG_MASK (3<<2) +#define M98090_MIXHPRG_SHIFT 2 +#define M98090_MIXHPRG_WIDTH 2 +#define M98090_MIXHPRG_NUM (1<<M98090_MIXHPRG_WIDTH) +#define M98090_MIXHPLG_MASK (3<<0) +#define M98090_MIXHPLG_SHIFT 0 +#define M98090_MIXHPLG_WIDTH 2 +#define M98090_MIXHPLG_NUM (1<<M98090_MIXHPLG_WIDTH) + +/* + * M98090_REG_LVL_HP_LEFT + */ +#define M98090_HPLM_MASK (1<<7) +#define M98090_HPLM_SHIFT 7 +#define M98090_HPLM_WIDTH 1 +#define M98090_HPVOLL_MASK (31<<0) +#define M98090_HPVOLL_SHIFT 0 +#define M98090_HPVOLL_WIDTH 5 +#define M98090_HPVOLL_NUM (1<<M98090_HPVOLL_WIDTH) + +/* + * M98090_REG_LVL_HP_RIGHT + */ +#define M98090_HPRM_MASK (1<<7) +#define M98090_HPRM_SHIFT 7 +#define M98090_HPRM_WIDTH 1 +#define M98090_HPVOLR_MASK (31<<0) +#define M98090_HPVOLR_SHIFT 0 +#define M98090_HPVOLR_WIDTH 5 +#define M98090_HPVOLR_NUM (1<<M98090_HPVOLR_WIDTH) + +/* + * M98090_REG_MIX_SPK_LEFT + */ +#define M98090_MIXSPL_MIC2_MASK (1<<5) +#define M98090_MIXSPL_MIC2_SHIFT 5 +#define M98090_MIXSPL_MIC2_WIDTH 1 +#define M98090_MIXSPL_MIC1_MASK (1<<4) +#define M98090_MIXSPL_MIC1_SHIFT 4 +#define M98090_MIXSPL_MIC1_WIDTH 1 +#define M98090_MIXSPL_LINEB_MASK (1<<3) +#define M98090_MIXSPL_LINEB_SHIFT 3 +#define M98090_MIXSPL_LINEB_WIDTH 1 +#define M98090_MIXSPL_LINEA_MASK (1<<2) +#define M98090_MIXSPL_LINEA_SHIFT 2 +#define M98090_MIXSPL_LINEA_WIDTH 1 +#define M98090_MIXSPL_DACR_MASK (1<<1) +#define M98090_MIXSPL_DACR_SHIFT 1 +#define M98090_MIXSPL_DACR_WIDTH 1 +#define M98090_MIXSPL_DACL_MASK (1<<0) +#define M98090_MIXSPL_DACL_SHIFT 0 +#define M98090_MIXSPL_DACL_WIDTH 1 +#define M98090_MIXSPL_MASK (63<<0) +#define M98090_MIXSPL_SHIFT 0 +#define M98090_MIXSPL_WIDTH 6 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 + + +/* + * M98090_REG_MIX_SPK_RIGHT + */ +#define M98090_SPK_SLAVE_MASK (1<<6) +#define M98090_SPK_SLAVE_SHIFT 6 +#define M98090_SPK_SLAVE_WIDTH 1 +#define M98090_MIXSPR_MIC2_MASK (1<<5) +#define M98090_MIXSPR_MIC2_SHIFT 5 +#define M98090_MIXSPR_MIC2_WIDTH 1 +#define M98090_MIXSPR_MIC1_MASK (1<<4) +#define M98090_MIXSPR_MIC1_SHIFT 4 +#define M98090_MIXSPR_MIC1_WIDTH 1 +#define M98090_MIXSPR_LINEB_MASK (1<<3) +#define M98090_MIXSPR_LINEB_SHIFT 3 +#define M98090_MIXSPR_LINEB_WIDTH 1 +#define M98090_MIXSPR_LINEA_MASK (1<<2) +#define M98090_MIXSPR_LINEA_SHIFT 2 +#define M98090_MIXSPR_LINEA_WIDTH 1 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 +#define M98090_MIXSPR_DACL_MASK (1<<0) +#define M98090_MIXSPR_DACL_SHIFT 0 +#define M98090_MIXSPR_DACL_WIDTH 1 +#define M98090_MIXSPR_MASK (63<<0) +#define M98090_MIXSPR_SHIFT 0 +#define M98090_MIXSPR_WIDTH 6 + +/* + * M98090_REG_MIX_SPK_CNTL + */ +#define M98090_MIXSPRG_MASK (3<<2) +#define M98090_MIXSPRG_SHIFT 2 +#define M98090_MIXSPRG_WIDTH 2 +#define M98090_MIXSPRG_NUM (1<<M98090_MIXSPRG_WIDTH) +#define M98090_MIXSPLG_MASK (3<<0) +#define M98090_MIXSPLG_SHIFT 0 +#define M98090_MIXSPLG_WIDTH 2 +#define M98090_MIXSPLG_NUM (1<<M98090_MIXSPLG_WIDTH) + +/* + * M98090_REG_LVL_SPK_LEFT + */ +#define M98090_SPLM_MASK (1<<7) +#define M98090_SPLM_SHIFT 7 +#define M98090_SPLM_WIDTH 1 +#define M98090_SPVOLL_MASK (63<<0) +#define M98090_SPVOLL_SHIFT 0 +#define M98090_SPVOLL_WIDTH 6 +#define M98090_SPVOLL_NUM 40 + +/* + * M98090_REG_LVL_SPK_RIGHT + */ +#define M98090_SPRM_MASK (1<<7) +#define M98090_SPRM_SHIFT 7 +#define M98090_SPRM_WIDTH 1 +#define M98090_SPVOLR_MASK (63<<0) +#define M98090_SPVOLR_SHIFT 0 +#define M98090_SPVOLR_WIDTH 6 +#define M98090_SPVOLR_NUM 40 + +/* + * M98090_REG_ALC_TIMING + */ +#define M98090_ALCEN_MASK (1<<7) +#define M98090_ALCEN_SHIFT 7 +#define M98090_ALCEN_WIDTH 1 +#define M98090_ALCEN_NUM (1<<M98090_ALCEN_WIDTH) +#define M98090_ALCRLS_MASK (7<<4) +#define M98090_ALCRLS_SHIFT 4 +#define M98090_ALCRLS_WIDTH 3 +#define M98090_ALCATK_MASK (7<<0) +#define M98090_ALCATK_SHIFT 0 +#define M98090_ALCATK_WIDTH 3 + +/* + * M98090_REG_ALC_CMPR + */ +#define M98090_ALCCMP_MASK (7<<5) +#define M98090_ALCCMP_SHIFT 5 +#define M98090_ALCCMP_WIDTH 3 +#define M98090_ALCTHC_MASK (31<<0) +#define M98090_ALCTHC_SHIFT 0 +#define M98090_ALCTHC_WIDTH 5 +#define M98090_ALCTHC_NUM (1<<M98090_ALCTHC_WIDTH) + +/* + * M98090_REG_ALC_EXP + */ +#define M98090_ALCEXP_MASK (7<<5) +#define M98090_ALCEXP_SHIFT 5 +#define M98090_ALCEXP_WIDTH 3 +#define M98090_ALCTHE_MASK (31<<0) +#define M98090_ALCTHE_SHIFT 0 +#define M98090_ALCTHE_WIDTH 5 +#define M98090_ALCTHE_NUM (1<<M98090_ALCTHE_WIDTH) + +/* + * M98090_REG_LVL_ALC + */ +#define M98090_ALCG_MASK (31<<0) +#define M98090_ALCG_SHIFT 0 +#define M98090_ALCG_WIDTH 5 +#define M98090_ALCG_NUM 13 + +/* + * M98090_REG_MIX_RCV_LEFT + */ +#define M98090_MIXRCVL_MIC2_MASK (1<<5) +#define M98090_MIXRCVL_MIC2_SHIFT 5 +#define M98090_MIXRCVL_MIC2_WIDTH 1 +#define M98090_MIXRCVL_MIC1_MASK (1<<4) +#define M98090_MIXRCVL_MIC1_SHIFT 4 +#define M98090_MIXRCVL_MIC1_WIDTH 1 +#define M98090_MIXRCVL_LINEB_MASK (1<<3) +#define M98090_MIXRCVL_LINEB_SHIFT 3 +#define M98090_MIXRCVL_LINEB_WIDTH 1 +#define M98090_MIXRCVL_LINEA_MASK (1<<2) +#define M98090_MIXRCVL_LINEA_SHIFT 2 +#define M98090_MIXRCVL_LINEA_WIDTH 1 +#define M98090_MIXRCVL_DACR_MASK (1<<1) +#define M98090_MIXRCVL_DACR_SHIFT 1 +#define M98090_MIXRCVL_DACR_WIDTH 1 +#define M98090_MIXRCVL_DACL_MASK (1<<0) +#define M98090_MIXRCVL_DACL_SHIFT 0 +#define M98090_MIXRCVL_DACL_WIDTH 1 +#define M98090_MIXRCVL_MASK (63<<0) +#define M98090_MIXRCVL_SHIFT 0 +#define M98090_MIXRCVL_WIDTH 6 + +/* + * M98090_REG_MIX_RCV_CTRL_LEFT + */ +#define M98090_MIXRCVLG_MASK (3<<0) +#define M98090_MIXRCVLG_SHIFT 0 +#define M98090_MIXRCVLG_WIDTH 2 +#define M98090_MIXRCVLG_NUM (1<<M98090_MIXRCVLG_WIDTH) + +/* + * M98090_REG_LVL_RCV_LEFT + */ +#define M98090_RCVLM_MASK (1<<7) +#define M98090_RCVLM_SHIFT 7 +#define M98090_RCVLM_WIDTH 1 +#define M98090_RCVLVOL_MASK (31<<0) +#define M98090_RCVLVOL_SHIFT 0 +#define M98090_RCVLVOL_WIDTH 5 +#define M98090_RCVLVOL_NUM (1<<M98090_RCVLVOL_WIDTH) + +/* + * M98090_REG_MIX_RCV_RIGHT + */ +#define M98090_LINMOD_MASK (1<<7) +#define M98090_LINMOD_SHIFT 7 +#define M98090_LINMOD_WIDTH 1 +#define M98090_MIXRCVR_MIC2_MASK (1<<5) +#define M98090_MIXRCVR_MIC2_SHIFT 5 +#define M98090_MIXRCVR_MIC2_WIDTH 1 +#define M98090_MIXRCVR_MIC1_MASK (1<<4) +#define M98090_MIXRCVR_MIC1_SHIFT 4 +#define M98090_MIXRCVR_MIC1_WIDTH 1 +#define M98090_MIXRCVR_LINEB_MASK (1<<3) +#define M98090_MIXRCVR_LINEB_SHIFT 3 +#define M98090_MIXRCVR_LINEB_WIDTH 1 +#define M98090_MIXRCVR_LINEA_MASK (1<<2) +#define M98090_MIXRCVR_LINEA_SHIFT 2 +#define M98090_MIXRCVR_LINEA_WIDTH 1 +#define M98090_MIXRCVR_DACR_MASK (1<<1) +#define M98090_MIXRCVR_DACR_SHIFT 1 +#define M98090_MIXRCVR_DACR_WIDTH 1 +#define M98090_MIXRCVR_DACL_MASK (1<<0) +#define M98090_MIXRCVR_DACL_SHIFT 0 +#define M98090_MIXRCVR_DACL_WIDTH 1 +#define M98090_MIXRCVR_MASK (63<<0) +#define M98090_MIXRCVR_SHIFT 0 +#define M98090_MIXRCVR_WIDTH 6 + +/* + * M98090_REG_MIX_RCV_CNTL_RIGHT + */ +#define M98090_MIXRCVRG_MASK (3<<0) +#define M98090_MIXRCVRG_SHIFT 0 +#define M98090_MIXRCVRG_WIDTH 2 +#define M98090_MIXRCVRG_NUM (1<<M98090_MIXRCVRG_WIDTH) + +/* + * M98090_REG_LVL_RCV_RIGHT + */ +#define M98090_RCVRM_MASK (1<<7) +#define M98090_RCVRM_SHIFT 7 +#define M98090_RCVRM_WIDTH 1 +#define M98090_RCVRVOL_MASK (31<<0) +#define M98090_RCVRVOL_SHIFT 0 +#define M98090_RCVRVOL_WIDTH 5 +#define M98090_RCVRVOL_NUM (1<<M98090_RCVRVOL_WIDTH) + +/* + * M98090_REG_CFG_JACK + */ +#define M98090_JDETEN_MASK (1<<7) +#define M98090_JDETEN_SHIFT 7 +#define M98090_JDETEN_WIDTH 1 +#define M98090_JDWK_MASK (1<<6) +#define M98090_JDWK_SHIFT 6 +#define M98090_JDWK_WIDTH 1 +#define M98090_JDEB_MASK (3<<0) +#define M98090_JDEB_SHIFT 0 +#define M98090_JDEB_WIDTH 2 +#define M98090_JDEB_25MS (0<<0) +#define M98090_JDEB_50MS (1<<0) +#define M98090_JDEB_100MS (2<<0) +#define M98090_JDEB_200MS (3<<0) + +/* + * M98090_REG_PWR_EN_IN + */ +#define M98090_PWR_MBEN_MASK (1<<4) +#define M98090_PWR_MBEN_SHIFT 4 +#define M98090_PWR_MBEN_WIDTH 1 +#define M98090_PWR_LINEAEN_MASK (1<<3) +#define M98090_PWR_LINEAEN_SHIFT 3 +#define M98090_PWR_LINEAEN_WIDTH 1 +#define M98090_PWR_LINEBEN_MASK (1<<2) +#define M98090_PWR_LINEBEN_SHIFT 2 +#define M98090_PWR_LINEBEN_WIDTH 1 +#define M98090_PWR_ADREN_MASK (1<<1) +#define M98090_PWR_ADREN_SHIFT 1 +#define M98090_PWR_ADREN_WIDTH 1 +#define M98090_PWR_ADLEN_MASK (1<<0) +#define M98090_PWR_ADLEN_SHIFT 0 +#define M98090_PWR_ADLEN_WIDTH 1 + +/* + * M98090_REG_PWR_EN_OUT + */ +#define M98090_PWR_HPREN_MASK (1<<7) +#define M98090_PWR_HPREN_SHIFT 7 +#define M98090_PWR_HPREN_WIDTH 1 +#define M98090_PWR_HPLEN_MASK (1<<6) +#define M98090_PWR_HPLEN_SHIFT 6 +#define M98090_PWR_HPLEN_WIDTH 1 +#define M98090_PWR_SPREN_MASK (1<<5) +#define M98090_PWR_SPREN_SHIFT 5 +#define M98090_PWR_SPREN_WIDTH 1 +#define M98090_PWR_SPLEN_MASK (1<<4) +#define M98090_PWR_SPLEN_SHIFT 4 +#define M98090_PWR_SPLEN_WIDTH 1 +#define M98090_PWR_RCVLEN_MASK (1<<3) +#define M98090_PWR_RCVLEN_SHIFT 3 +#define M98090_PWR_RCVLEN_WIDTH 1 +#define M98090_PWR_RCVREN_MASK (1<<2) +#define M98090_PWR_RCVREN_SHIFT 2 +#define M98090_PWR_RCVREN_WIDTH 1 +#define M98090_PWR_DAREN_MASK (1<<1) +#define M98090_PWR_DAREN_SHIFT 1 +#define M98090_PWR_DAREN_WIDTH 1 +#define M98090_PWR_DALEN_MASK (1<<0) +#define M98090_PWR_DALEN_SHIFT 0 +#define M98090_PWR_DALEN_WIDTH 1 + +/* + * M98090_REG_CFG_LVL + */ +#define M98090_ZDENN_MASK (1<<2) +#define M98090_ZDENN_SHIFT 2 +#define M98090_ZDENN_WIDTH 1 +#define M98090_VS2ENN_MASK (1<<1) +#define M98090_VS2ENN_SHIFT 1 +#define M98090_VS2ENN_WIDTH 1 +#define M98090_VSENN_MASK (1<<0) +#define M98090_VSENN_SHIFT 0 +#define M98090_VSENN_WIDTH 1 + +/* + * M98090_REG_DSP_EQ_EN + */ +#define M98090_DMIC34BQEN_MASK (1<<4) +#define M98090_DMIC34BQEN_SHIFT 4 +#define M98090_DMIC34BQEN_WIDTH 1 +#define M98090_DMIC34BQEN_NUM (1<<M98090_DMIC34BQEN_WIDTH) +#define M98090_ADCBQEN_MASK (1<<3) +#define M98090_ADCBQEN_SHIFT 3 +#define M98090_ADCBQEN_WIDTH 1 +#define M98090_ADCBQEN_NUM (1<<M98090_ADCBQEN_WIDTH) +#define M98090_EQ3BANDEN_MASK (1<<2) +#define M98090_EQ3BANDEN_SHIFT 2 +#define M98090_EQ3BANDEN_WIDTH 1 +#define M98090_EQ3BANDEN_NUM (1<<M98090_EQ3BANDEN_WIDTH) +#define M98090_EQ5BANDEN_MASK (1<<1) +#define M98090_EQ5BANDEN_SHIFT 1 +#define M98090_EQ5BANDEN_WIDTH 1 +#define M98090_EQ5BANDEN_NUM (1<<M98090_EQ5BANDEN_WIDTH) +#define M98090_EQ7BANDEN_MASK (1<<0) +#define M98090_EQ7BANDEN_SHIFT 0 +#define M98090_EQ7BANDEN_WIDTH 1 +#define M98090_EQ7BANDEN_NUM (1<<M98090_EQ7BANDEN_WIDTH) + +/* + * M98090_REG_BIAS_CNTL + */ +#define M98090_VCM_MODE_MASK (1<<0) +#define M98090_VCM_MODE_SHIFT 0 +#define M98090_VCM_MODE_WIDTH 1 +#define M98090_VCM_MODE_NUM (1<<M98090_VCM_MODE_WIDTH) + +/* + * M98090_REG_DAC_CFG + */ +#define M98090_DAC_PERFMODE_MASK (1<<1) +#define M98090_DAC_PERFMODE_SHIFT 1 +#define M98090_DAC_PERFMODE_WIDTH 1 +#define M98090_DAC_PERFMODE_NUM (1<<M98090_DAC_PERFMODE_WIDTH) +#define M98090_DACHP_MASK (1<<0) +#define M98090_DACHP_SHIFT 0 +#define M98090_DACHP_WIDTH 1 +#define M98090_DACHP_NUM (1<<M98090_DACHP_WIDTH) + +/* + * M98090_REG_ADC_CFG + */ +#define M98090_ADC_OSR128_MASK (1<<2) +#define M98090_ADC_OSR128_SHIFT 2 +#define M98090_ADC_OSR128_WIDTH 1 +#define M98090_ADCDITHER_MASK (1<<1) +#define M98090_ADCDITHER_SHIFT 1 +#define M98090_ADCDITHER_WIDTH 1 +#define M98090_ADCDITHER_NUM (1<<M98090_ADCDITHER_WIDTH) +#define M98090_ADCHP_MASK (1<<0) +#define M98090_ADCHP_SHIFT 0 +#define M98090_ADCHP_WIDTH 1 +#define M98090_ADCHP_NUM (1<<M98090_ADCHP_WIDTH) + +/* + * M98090_REG_PWR_SYS + */ +#define M98090_PWR_SHDNN_MASK (1<<7) +#define M98090_PWR_SHDNN_SHIFT 7 +#define M98090_PWR_SHDNN_WIDTH 1 + +/* + * M98090_REG_EQ_BASE + */ +#define M98090_B0_1_HI_MASK (255<<0) +#define M98090_B0_1_HI_SHIFT 0 +#define M98090_B0_1_HI_WIDTH 8 +#define M98090_B0_1_MID_MASK (255<<0) +#define M98090_B0_1_MID_SHIFT 0 +#define M98090_B0_1_MID_WIDTH 8 +#define M98090_B0_1_LO_MASK (255<<0) +#define M98090_B0_1_LO_SHIFT 0 +#define M98090_B0_1_LO_WIDTH 8 +#define M98090_B1_1_HI_MASK (255<<0) +#define M98090_B1_1_HI_SHIFT 0 +#define M98090_B1_1_HI_WIDTH 8 +#define M98090_B1_1_MID_MASK (255<<0) +#define M98090_B1_1_MID_SHIFT 0 +#define M98090_B1_1_MID_WIDTH 8 +#define M98090_B1_1_LO_MASK (255<<0) +#define M98090_B1_1_LO_SHIFT 0 +#define M98090_B1_1_LO_WIDTH 8 +#define M98090_B2_1_HI_MASK (255<<0) +#define M98090_B2_1_HI_SHIFT 0 +#define M98090_B2_1_HI_WIDTH 8 +#define M98090_B2_1_MID_MASK (255<<0) +#define M98090_B2_1_MID_SHIFT 0 +#define M98090_B2_1_MID_WIDTH 8 +#define M98090_B2_1_LO_MASK (255<<0) +#define M98090_B2_1_LO_SHIFT 0 +#define M98090_B2_1_LO_WIDTH 8 +#define M98090_A1_1_HI_MASK (255<<0) +#define M98090_A1_1_HI_SHIFT 0 +#define M98090_A1_1_HI_WIDTH 8 +#define M98090_A1_1_MID_MASK (255<<0) +#define M98090_A1_1_MID_SHIFT 0 +#define M98090_A1_1_MID_WIDTH 8 +#define M98090_A1_1_LO_MASK (255<<0) +#define M98090_A1_1_LO_SHIFT 0 +#define M98090_A1_1_LO_WIDTH 8 +#define M98090_A2_1_HI_MASK (255<<0) +#define M98090_A2_1_HI_SHIFT 0 +#define M98090_A2_1_HI_WIDTH 8 +#define M98090_A2_1_MID_MASK (255<<0) +#define M98090_A2_1_MID_SHIFT 0 +#define M98090_A2_1_MID_WIDTH 8 +#define M98090_A2_1_LO_MASK (255<<0) +#define M98090_A2_1_LO_SHIFT 0 +#define M98090_A2_1_LO_WIDTH 8 + +#define M98090_COEFS_PER_BAND 5 +#define M98090_COEFS_BLK_SZ (M98090_COEFS_PER_BAND * 3) +#define M98090_COEFS_MAX_SZ (M98090_COEFS_BLK_SZ * 7) + +/* + * M98090_REG_BIQUAD_BASE + */ +#define M98090_ADC_B0_HI_MASK (255<<0) +#define M98090_ADC_B0_HI_SHIFT 0 +#define M98090_ADC_B0_HI_WIDTH 8 +#define M98090_ADC_B0_MID_MASK (255<<0) +#define M98090_ADC_B0_MID_SHIFT 0 +#define M98090_ADC_B0_MID_WIDTH 8 +#define M98090_ADC_B0_LO_MASK (255<<0) +#define M98090_ADC_B0_LO_SHIFT 0 +#define M98090_ADC_B0_LO_WIDTH 8 +#define M98090_ADC_B1_HI_MASK (255<<0) +#define M98090_ADC_B1_HI_SHIFT 0 +#define M98090_ADC_B1_HI_WIDTH 8 +#define M98090_ADC_B1_MID_MASK (255<<0) +#define M98090_ADC_B1_MID_SHIFT 0 +#define M98090_ADC_B1_MID_WIDTH 8 +#define M98090_ADC_B1_LO_MASK (255<<0) +#define M98090_ADC_B1_LO_SHIFT 0 +#define M98090_ADC_B1_LO_WIDTH 8 +#define M98090_ADC_B2_HI_MASK (255<<0) +#define M98090_ADC_B2_HI_SHIFT 0 +#define M98090_ADC_B2_HI_WIDTH 8 +#define M98090_ADC_B2_MID_MASK (255<<0) +#define M98090_ADC_B2_MID_SHIFT 0 +#define M98090_ADC_B2_MID_WIDTH 8 +#define M98090_ADC_B2_LO_MASK (255<<0) +#define M98090_ADC_B2_LO_SHIFT 0 +#define M98090_ADC_B2_LO_WIDTH 8 +#define M98090_ADC_A1_HI_MASK (255<<0) +#define M98090_ADC_A1_HI_SHIFT 0 +#define M98090_ADC_A1_HI_WIDTH 8 +#define M98090_ADC_A1_MID_MASK (255<<0) +#define M98090_ADC_A1_MID_SHIFT 0 +#define M98090_ADC_A1_MID_WIDTH 8 +#define M98090_ADC_A1_LO_MASK (255<<0) +#define M98090_ADC_A1_LO_SHIFT 0 +#define M98090_ADC_A1_LO_WIDTH 8 +#define M98090_ADC_A2_HI_MASK (255<<0) +#define M98090_ADC_A2_HI_SHIFT 0 +#define M98090_ADC_A2_HI_WIDTH 8 +#define M98090_ADC_A2_MID_MASK (255<<0) +#define M98090_ADC_A2_MID_SHIFT 0 +#define M98090_ADC_A2_MID_WIDTH 8 +#define M98090_ADC_A2_LO_MASK (255<<0) +#define M98090_ADC_A2_LO_SHIFT 0 +#define M98090_ADC_A2_LO_WIDTH 8 + +/* + * M98090_REG_DMIC3_VOLUME + */ +#define M98090_DMIC_AV3G_MASK (7<<4) +#define M98090_DMIC_AV3G_SHIFT 4 +#define M98090_DMIC_AV3G_WIDTH 3 +#define M98090_DMIC_AV3G_NUM (1<<M98090_DMIC_AV3G_WIDTH) +#define M98090_DMIC_AV3_MASK (15<<0) +#define M98090_DMIC_AV3_SHIFT 0 +#define M98090_DMIC_AV3_WIDTH 4 +#define M98090_DMIC_AV3_NUM (1<<M98090_DMIC_AV3_WIDTH) + +/* + * M98090_REG_DMIC4_VOLUME + */ +#define M98090_DMIC_AV4G_MASK (7<<4) +#define M98090_DMIC_AV4G_SHIFT 4 +#define M98090_DMIC_AV4G_WIDTH 3 +#define M98090_DMIC_AV4G_NUM (1<<M98090_DMIC_AV4G_WIDTH) +#define M98090_DMIC_AV4_MASK (15<<0) +#define M98090_DMIC_AV4_SHIFT 0 +#define M98090_DMIC_AV4_WIDTH 4 +#define M98090_DMIC_AV4_NUM (1<<M98090_DMIC_AV4_WIDTH) + +/* + * M98090_REG_DMIC34_BQ_PREATTEN + */ +#define M98090_AV34BQ_MASK (15<<0) +#define M98090_AV34BQ_SHIFT 0 +#define M98090_AV34BQ_WIDTH 4 +#define M98090_AV34BQ_NUM (1<<M98090_AV34BQ_WIDTH) + +/* + * M98090_REG_RECORD_TDM_SLOT + */ +#define M98090_DAI_TDM_SLOTADCL_MASK (3<<6) +#define M98090_DAI_TDM_SLOTADCL_SHIFT 6 +#define M98090_DAI_TDM_SLOTADCL_WIDTH 2 +#define M98090_DAI_TDM_SLOTADCL_NUM (1<<M98090_DAI_TDM_SLOTADCL_WIDTH) +#define M98090_DAI_TDM_SLOTADCR_MASK (3<<4) +#define M98090_DAI_TDM_SLOTADCR_SHIFT 4 +#define M98090_DAI_TDM_SLOTADCR_WIDTH 2 +#define M98090_DAI_TDM_SLOTADCR_NUM (1<<M98090_DAI_TDM_SLOTADCR_WIDTH) +#define M98090_DAI_TDM_SLOTDMIC3_MASK (3<<2) +#define M98090_DAI_TDM_SLOTDMIC3_SHIFT 2 +#define M98090_DAI_TDM_SLOTDMIC3_WIDTH 2 +#define M98090_DAI_TDM_SLOTDMIC3_NUM (1<<M98090_DAI_TDM_SLOTDMIC3_WIDTH) +#define M98090_DAI_TDM_SLOTDMIC4_MASK (3<<0) +#define M98090_DAI_TDM_SLOTDMIC4_SHIFT 0 +#define M98090_DAI_TDM_SLOTDMIC4_WIDTH 2 +#define M98090_DAI_TDM_SLOTDMIC4_NUM (1<<M98090_DAI_TDM_SLOTDMIC4_WIDTH) + +/* + * M98090_REG_SAMPLE_RATE + */ +#define M98090_DMIC34_ZEROPAD_MASK (1<<4) +#define M98090_DMIC34_ZEROPAD_SHIFT 4 +#define M98090_DMIC34_ZEROPAD_WIDTH 1 +#define M98090_DMIC34_ZEROPAD_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DMIC34_SRDIV_MASK (7<<0) +#define M98090_DMIC34_SRDIV_SHIFT 0 +#define M98090_DMIC34_SRDIV_WIDTH 3 + +/* + * M98090_REG_DMIC34_BIQUAD_BASE + */ +#define M98090_DMIC34_B0_HI_MASK (255<<0) +#define M98090_DMIC34_B0_HI_SHIFT 0 +#define M98090_DMIC34_B0_HI_WIDTH 8 +#define M98090_DMIC34_B0_MID_MASK (255<<0) +#define M98090_DMIC34_B0_MID_SHIFT 0 +#define M98090_DMIC34_B0_MID_WIDTH 8 +#define M98090_DMIC34_B0_LO_MASK (255<<0) +#define M98090_DMIC34_B0_LO_SHIFT 0 +#define M98090_DMIC34_B0_LO_WIDTH 8 +#define M98090_DMIC34_B1_HI_MASK (255<<0) +#define M98090_DMIC34_B1_HI_SHIFT 0 +#define M98090_DMIC34_B1_HI_WIDTH 8 +#define M98090_DMIC34_B1_MID_MASK (255<<0) +#define M98090_DMIC34_B1_MID_SHIFT 0 +#define M98090_DMIC34_B1_MID_WIDTH 8 +#define M98090_DMIC34_B1_LO_MASK (255<<0) +#define M98090_DMIC34_B1_LO_SHIFT 0 +#define M98090_DMIC34_B1_LO_WIDTH 8 +#define M98090_DMIC34_B2_HI_MASK (255<<0) +#define M98090_DMIC34_B2_HI_SHIFT 0 +#define M98090_DMIC34_B2_HI_WIDTH 8 +#define M98090_DMIC34_B2_MID_MASK (255<<0) +#define M98090_DMIC34_B2_MID_SHIFT 0 +#define M98090_DMIC34_B2_MID_WIDTH 8 +#define M98090_DMIC34_B2_LO_MASK (255<<0) +#define M98090_DMIC34_B2_LO_SHIFT 0 +#define M98090_DMIC34_B2_LO_WIDTH 8 +#define M98090_DMIC34_A1_HI_MASK (255<<0) +#define M98090_DMIC34_A1_HI_SHIFT 0 +#define M98090_DMIC34_A1_HI_WIDTH 8 +#define M98090_DMIC34_A1_MID_MASK (255<<0) +#define M98090_DMIC34_A1_MID_SHIFT 0 +#define M98090_DMIC34_A1_MID_WIDTH 8 +#define M98090_DMIC34_A1_LO_MASK (255<<0) +#define M98090_DMIC34_A1_LO_SHIFT 0 +#define M98090_DMIC34_A1_LO_WIDTH 8 +#define M98090_DMIC34_A2_HI_MASK (255<<0) +#define M98090_DMIC34_A2_HI_SHIFT 0 +#define M98090_DMIC34_A2_HI_WIDTH 8 +#define M98090_DMIC34_A2_MID_MASK (255<<0) +#define M98090_DMIC34_A2_MID_SHIFT 0 +#define M98090_DMIC34_A2_MID_WIDTH 8 +#define M98090_DMIC34_A2_LO_MASK (255<<0) +#define M98090_DMIC34_A2_LO_SHIFT 0 +#define M98090_DMIC34_A2_LO_WIDTH 8 + +#define M98090_JACK_STATE_NO_HEADSET 0 +#define M98090_JACK_STATE_NO_HEADSET_2 1 +#define M98090_JACK_STATE_HEADPHONE 2 +#define M98090_JACK_STATE_HEADSET 3 + +/* + * M98090_REG_REV_ID + */ +#define M98090_REVID_MASK (255<<0) +#define M98090_REVID_SHIFT 0 +#define M98090_REVID_WIDTH 8 +#define M98090_REVID_NUM (1<<M98090_REVID_WIDTH) + +#define M98090_BYTE1(w) ((w >> 8) & 0xff) +#define M98090_BYTE0(w) (w & 0xff) + +/* Silicon revision number */ +#define M98090_REVA 0x40 +#define M98091_REVA 0x50 + +enum max98090_type { + MAX98090, + MAX98091, +}; + +struct max98090_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98090_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + enum max98090_type devtype; + void *control_data; + struct max98090_pdata *pdata; + unsigned int sysclk; + unsigned int bclk; + unsigned int lrclk; + struct max98090_cdata dai[1]; + int irq; + int jack_state; + struct delayed_work jack_work; + struct snd_soc_jack *jack; + unsigned int dai_fmt; + int tdm_slots; + int tdm_width; + u8 lin_state; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mux; + unsigned int sidetone; +}; + +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 1.7.0.4

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Re: [alsa-devel] HD-audio generic parser merged to linux-next
by Takashi Iwai 23 Jan '13

23 Jan '13

At Wed, 23 Jan 2013 15:01:43 +0100, Manolo Díaz wrote: > > El mié, 23 ene 2013 a las 14:54 horas > Takashi Iwai escribió: > > >At Wed, 23 Jan 2013 14:49:32 +0100, > >Manolo Díaz wrote: > >> > >> El mié, 23 ene 2013 a las 9:30 horas > >> Takashi Iwai escribió: > >> > >> >Hi, > >> > > >> >as most of feedbacks about the new generic parser migration were > >> >positive, I decided to take a risk: now all commit that have been in > >> >test/hda-migrate branch in sound-unstable git tree are merged to > >> >for-next branch of sound git tree for better test coverage. > >> >It means that all changes are merged to linux-next, i.e. will be > >> >included in 3.9 kernel. > >> > > >> >For the early adapter people, I'm going to keep test/hda-migrate > >> >branch of sound-unstable git tree updated for a while, but the branch > >> >will be abandoned some time later. > >> > > >> >If you encounter any regression by the recent update, please let me > >> >know. > >> > > >> > > >> >thanks, > >> > > >> >Takashi > >> >_______________________________________________ > >> >Alsa-devel mailing list > >> >Alsa-devel(a)alsa-project.org > >> >http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >> > >> Hi, > >> > >> Recording and (multi stream) playing seem to work fine, but auto-mute > >> doesn't. For example, front speakers may remain muted after > >> disconnecting the HP. > >> > >> cat /sys/kernel/debug/tracing/trace > >> # tracer: nop > >> # > >> # entries-in-buffer/entries-written: 0/0 #P:2 > >> # > >> # _-----=> irqs-off > >> # / _----=> need-resched > >> # | / _---=> hardirq/softirq > >> # || / _--=> preempt-depth > >> # ||| / delay > >> # TASK-PID CPU# |||| TIMESTAMP FUNCTION > >> # | | | |||| | | > >> > >> I've connected and disconnected the HP jack several times before. > > > >If nothing appears in the tracing (suppose you enabled the events > >properly via /sys/kernel/debug/tracing/events/hda/enable), it's rather > >a hardware issue. > > Sorry, I've forgotten this step. > > >Did the auto-mute work reliably before the code change? > > > > > >Takashi > > Yes, it did. Did you mean when it worked? Was it before moving to the generic parser, or after it but before upgrading to the latest tree? In the latter case, try git bisection to figure out the culprit. But, it's anyway better to update the git branch. You are testing the too old version (there have been already 75 commits after your testing one). > New try, this time with hda events enabled. > > cat /sys/kernel/debug/tracing/trace > # tracer: nop > # > # entries-in-buffer/entries-written: 871/871 #P:2 > # > # _-----=> irqs-off > # / _----=> need-resched > # | / _---=> hardirq/softirq > # || / _--=> preempt-depth > # ||| / delay > # TASK-PID CPU# |||| TIMESTAMP FUNCTION > # | | | |||| | | > <idle>-0 [001] d.h. 4509.450172: hda_unsol_event: [0] res=4000000, res_ex=10 > <idle>-0 [001] d.h. 4513.991711: hda_unsol_event: [0] res=4000000, res_ex=10 OK, so the unsol event is generated properly. Is the alsa-info.sh output you attached previously at the state after unplugging? If so, I see no problem in the code register. The path 0x04 -> 0x29 -> 0x12 looks fine, all unmuted, power D0 and the pin set properly. If something doesn't work as expected, we need two alsa-info outputs for comparison: the working one and non-working one. Takashi

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[alsa-devel] [PATCH] ASoC: wm2000: Lock state machine updates
by Mark Brown 23 Jan '13

23 Jan '13

Need to ensure we don't get confused by simultaneous updates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 31 ++++++++++++++++++++++++++++--- 1 file changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 655f43e..4d809ab 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -77,6 +77,8 @@ struct wm2000_priv { int anc_download_size; char *anc_download; + + struct mutex lock; }; static int wm2000_write(struct i2c_client *i2c, unsigned int reg, @@ -614,13 +616,20 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int anc_active = ucontrol->value.enumerated.item[0]; + int ret; if (anc_active > 1) return -EINVAL; + mutex_lock(&wm2000->lock); + wm2000->anc_active = anc_active; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, @@ -640,13 +649,20 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int val = ucontrol->value.enumerated.item[0]; + int ret; if (val > 1) return -EINVAL; + mutex_lock(&wm2000->lock); + wm2000->spk_ena = val; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static const struct snd_kcontrol_new wm2000_controls[] = { @@ -663,6 +679,9 @@ static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + int ret; + + mutex_lock(&wm2000->lock); if (SND_SOC_DAPM_EVENT_ON(event)) wm2000->anc_eng_ena = 1; @@ -670,7 +689,11 @@ static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_OFF(event)) wm2000->anc_eng_ena = 0; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { @@ -800,6 +823,8 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } + mutex_init(&wm2000->lock); + dev_set_drvdata(&i2c->dev, wm2000); wm2000->regmap = devm_regmap_init_i2c(i2c, &wm2000_regmap); -- 1.7.10.4

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[alsa-devel] repeated hda_codec_cleanup_stream
by Raj Upadhyaya 23 Jan '13

23 Jan '13

(Bugzilla bug title..) [Fwd: [Bug 894259] Sound is garbled on playback of OGG, MP3s, WAV files and videos with sound.] I am trying to debug why I have some sound problems with my set up. I have recently recompiled the kernel to 3.8.0-rc4. Here is a suggestion that I got recently from the debugging session. The message -> ALSA hda_codec.c:1499 hda_codec_cleanup_stream: is coming up in dmesg quite often. Is this OK, given my kernel compile options? - Raj -------- Forwarded Message -------- From: bugzilla(a)redhat.com To: raj(a)upadhyaya.com Subject: [Bug 894259] Sound is garbled on playback of OGG, MP3s, WAV files and videos with sound. Date: Wed, 23 Jan 2013 06:04:20 +0000 Product: Fedora https://bugzilla.redhat.com/show_bug.cgi?id=894259 --- Comment #17 from Raymond <superquad.vortex2(a)gmail.com> --- i think you better report the repeated hda_codec_cleanup_stream to alsa development mailing list .374225] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.374575] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.374581] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.374585] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.374588] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.374592] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.374916] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.374921] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.374925] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.374929] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.374932] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.375602] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.375607] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.375611] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.375615] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.375619] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.376435] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.376440] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.376444] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.376448] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.376452] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 -- You are receiving this mail because: You reported the bug.

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[alsa-devel] sound/pci/hda/patch_analog.c:4987 patch_ad1882() error: potentially derefencing uninitialized 'spec'.
by Dan Carpenter 22 Jan '13

22 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 96349b211586f3a6f2750bc4ea195bd9739465b6 commit: e87b24f711c797f1277188dc85b866bec17704a0 ALSA: hda - Rearrange for dropping static quirk codes in AD codec driver New smatch warnings: sound/pci/hda/patch_analog.c:4987 patch_ad1882() error: potentially derefencing uninitialized 'spec'. git remote add sound-unstable git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git git remote update sound-unstable git checkout e87b24f711c797f1277188dc85b866bec17704a0 vim +/spec +4987 sound/pci/hda/patch_analog.c e87b24f7 Takashi Iwai 2013-01-22 4971 #ifdef ENABLE_AD_STATIC_QUIRKS 0ac8551e Takashi Iwai 2007-06-20 4972 static int patch_ad1882(struct hda_codec *codec) 0ac8551e Takashi Iwai 2007-06-20 4973 { 0ac8551e Takashi Iwai 2007-06-20 4974 struct ad198x_spec *spec; c5a4bcd0 Takashi Iwai 2009-02-06 4975 int err, board_config; 0ac8551e Takashi Iwai 2007-06-20 4976 78bb3cb0 Takashi Iwai 2012-12-21 4977 board_config = snd_hda_check_board_config(codec, AD1882_MODELS, 78bb3cb0 Takashi Iwai 2012-12-21 4978 ad1882_models, NULL); e87b24f7 Takashi Iwai 2013-01-22 4979 if (board_config == AD1882_AUTO) e87b24f7 Takashi Iwai 2013-01-22 4980 return ad1882_parse_auto_config(codec); 78bb3cb0 Takashi Iwai 2012-12-21 4981 c5a4bcd0 Takashi Iwai 2009-02-06 4982 err = snd_hda_attach_beep_device(codec, 0x10); c5a4bcd0 Takashi Iwai 2009-02-06 4983 if (err < 0) { c5a4bcd0 Takashi Iwai 2009-02-06 4984 ad198x_free(codec); c5a4bcd0 Takashi Iwai 2009-02-06 4985 return err; c5a4bcd0 Takashi Iwai 2009-02-06 4986 } c5a4bcd0 Takashi Iwai 2009-02-06 @4987 set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); c5a4bcd0 Takashi Iwai 2009-02-06 4988 0ac8551e Takashi Iwai 2007-06-20 4989 spec->multiout.max_channels = 6; 0ac8551e Takashi Iwai 2007-06-20 4990 spec->multiout.num_dacs = 3; 0ac8551e Takashi Iwai 2007-06-20 4991 spec->multiout.dac_nids = ad1882_dac_nids; 0ac8551e Takashi Iwai 2007-06-20 4992 spec->multiout.dig_out_nid = AD1882_SPDIF_OUT; 0ac8551e Takashi Iwai 2007-06-20 4993 spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); 0ac8551e Takashi Iwai 2007-06-20 4994 spec->adc_nids = ad1882_adc_nids; 0ac8551e Takashi Iwai 2007-06-20 4995 spec->capsrc_nids = ad1882_capsrc_nids; --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

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[alsa-devel] [PATCH 00/12] More updates on HD-audio generic parser
by Takashi Iwai 22 Jan '13

22 Jan '13

Hi, not surprisingly, here is another batch for fixing / improving the HD-audio generic parser code. With these, now AD and IDT codecs have proper analog-loopback features, and VIA and Conexant codecs have better aamix handling. The conflict of aamixer and indep-hp mode seen on VIA codecs should have been fixed now. As usual, all changes are found in test/hda-gen-parser (or test/hda-migrate) branch of sound-unstable tree. It's already merged in master branch as well. Takashi

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