May 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] pcm: Fix typo in documentation: PREPARE → PREPARED.
by Jordi Mallach 23 May '13

23 May '13

Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 438fb3f..7ec1f0e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -437,7 +437,7 @@ to #SND_PCM_STATE_SETUP if successfully finishes, otherwise the state #SND_PCM_STATE_OPEN is entered. When it is brought to SETUP state, this function automatically -calls #snd_pcm_prepare() function to bring to the PREPARE state +calls #snd_pcm_prepare() function to bring to the PREPARED state as below. \par snd_pcm_prepare -- 1.7.10.4

2 1

[alsa-devel] [PATCH alsa-utils] Require alsa-lib 1.0.27.
by Jordi Mallach 23 May '13

23 May '13

Both aplay and speaker-test use the new snd_pcm_abort() function, which was introduced in alsa-lib 1.0.27. Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- configure.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.in b/configure.in index aae3e66..0023e4d 100644 --- a/configure.in +++ b/configure.in @@ -32,7 +32,7 @@ AC_PROG_MKDIR_P AC_PROG_LN_S AC_PROG_SED PKG_PROG_PKG_CONFIG -AM_PATH_ALSA(1.0.24) +AM_PATH_ALSA(1.0.27) if test "x$enable_alsatest" = "xyes"; then AC_CHECK_FUNC([snd_ctl_elem_add_enumerated], , [AC_ERROR([No user enum control support in alsa-lib])]) -- 1.7.10.4

2 1

[alsa-devel] [PATCH] usb-6fire: Modify firmware version check
by Torsten Schenk 23 May '13

23 May '13

Check only the uppermost 16 bits instead of the whole 32 bits of the version information. Do this because all firmware version tested with this version information worked correctly and the strict check causes problems for several users. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c --- a/sound/usb/6fire/firmware.c 2013-05-23 13:32:39.432545897 +0200 +++ b/sound/usb/6fire/firmware.c 2013-05-23 13:33:21.998193012 +0200 @@ -42,8 +42,8 @@ 0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */ }; -static const u8 known_fw_versions[][4] = { - { 0x03, 0x01, 0x0b, 0x00 } +static const u8 known_fw_versions[][2] = { + { 0x03, 0x01 } }; struct ihex_record { @@ -343,7 +343,7 @@ int i; for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++) - if (!memcmp(version, known_fw_versions + i, 4)) + if (!memcmp(version, known_fw_versions + i, 2)) return 0; snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "

2 1

[alsa-devel] [PATCH 01/11] ALSA: pxa2xx-ac97: Remove redundant platform_set_drvdata()
by Sachin Kamat 23 May '13

23 May '13

Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Eric Miao <eric.y.miao(a)gmail.com> --- sound/arm/pxa2xx-ac97.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index ec54be4..ce431e6 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -230,7 +230,6 @@ static int pxa2xx_ac97_remove(struct platform_device *dev) if (card) { snd_card_free(card); - platform_set_drvdata(dev, NULL); pxa2xx_ac97_hw_remove(dev); } -- 1.7.9.5

3 12

[alsa-devel] [PATCH] ASoC: fsl: expand the size of the name in fsl_ssi_private struct
by Nicolin Chen 23 May '13

23 May '13

strcpy(ssi_private->name, p) in probe() sets "name" by "p", gotten from dts, while the length of "p", if the devicetree node name of SSI is commonly set, would always be larger than 1 char size, so need a larger size for "name". Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6a5f94 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -152,5 +152,5 @@ struct fsl_ssi_private { } stats; - char name[1]; + char name[32]; }; -- 1.7.1

3 2

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] crec: Initial version of a compressed capture utility
by Richard Fitzgerald 23 May '13

23 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- Android.mk | 10 ++ crec.c | 339 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 362 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index b9c52e2..5baccaf 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..4f13878 --- /dev/null +++ b/crec.c @@ -0,0 +1,339 @@ +/* + * BSD LICENSE + * + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * tinyplay command line player for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n\n" + "-v\tverbose mode\n" + "-h\tPrints this help list\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.raw\n" + "\tcrec -f 5 test.raw\n"); + + exit(EXIT_FAILURE); +} + +static int sample_rate_to_alsa(unsigned int *alsa_rate, int rate) +{ + switch (rate) { + case 5512: + *alsa_rate = SNDRV_PCM_RATE_5512; + return 0; + case 8000: + *alsa_rate = SNDRV_PCM_RATE_8000; + return 0; + case 11025: + *alsa_rate = SNDRV_PCM_RATE_11025; + return 0; + case 16000: + *alsa_rate = SNDRV_PCM_RATE_16000; + return 0; + case 220500: + *alsa_rate = SNDRV_PCM_RATE_22050; + return 0; + case 32000: + *alsa_rate = SNDRV_PCM_RATE_32000; + return 0; + case 44100: + *alsa_rate = SNDRV_PCM_RATE_44100; + return 0; + case 48000: + *alsa_rate = SNDRV_PCM_RATE_48000; + return 0; + case 64000: + *alsa_rate = SNDRV_PCM_RATE_64000; + return 0; + case 88200: + *alsa_rate = SNDRV_PCM_RATE_88200; + return 0; + case 96000: + *alsa_rate = SNDRV_PCM_RATE_96000; + return 0; + case 176400: + *alsa_rate = SNDRV_PCM_RATE_176400; + return 0; + case 192000: + *alsa_rate = SNDRV_PCM_RATE_192000; + return 0; + default: + return -EINVAL; + } +} + +static int alsa_to_sample_rate(unsigned int alsa_rate) +{ + switch (alsa_rate) { + case SNDRV_PCM_RATE_5512: + return 5512; + case SNDRV_PCM_RATE_8000: + return 8000; + case SNDRV_PCM_RATE_11025: + return 11025; + case SNDRV_PCM_RATE_16000: + return 16000; + case SNDRV_PCM_RATE_22050: + return 22050; + case SNDRV_PCM_RATE_32000: + return 32000; + case SNDRV_PCM_RATE_44100: + return 44100; + case SNDRV_PCM_RATE_48000: + return 48000; + case SNDRV_PCM_RATE_64000: + return 64000; + case SNDRV_PCM_RATE_88200: + return 88200; + case SNDRV_PCM_RATE_96000: + return 96000; + case SNDRV_PCM_RATE_176400: + return 176400; + case SNDRV_PCM_RATE_192000: + return 192000; + default: + return -EINVAL; + } +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else + fprintf(stderr, "DSP played %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, unsigned int rate) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + FILE *file; + char *buffer; + int size, num_read, wrote; + + if (verbose) + printf("%s: entry\n", __func__); + file = fopen(name, "wb"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = 1; + codec.ch_out = 1; + if (sample_rate_to_alsa(&codec.sample_rate, rate) != 0) { + fprintf(stderr, "unknown sample rate %d\n", rate); + goto FILE_EXIT; + } + + codec.bit_rate = 0; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = 0; + 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_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto FILE_EXIT; + }; + if (verbose) + printf("%s: Opened compress device\n", __func__); + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto COMP_EXIT; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, + alsa_to_sample_rate(codec.sample_rate)); + + compress_start(compress); + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + num_read = compress_read(compress, buffer, size); + if (num_read > 0) { + wrote = fwrite(buffer, 1, num_read, file); + if (wrote < 0) { + fprintf(stderr, "Error writing output file\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto BUF_EXIT; + } + if (wrote != num_read) { + /* TODO: Buffer pointer needs to be set here */ + fprintf(stderr, "DSP supplied %d, we wrote %d\n", num_read, wrote); + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, num_read); + } + } + } while (num_read > 0); + + if (verbose) + printf("%s: exit success\n", __func__); + free(buffer); + fclose(file); + compress_close(compress); + return; +BUF_EXIT: + free(buffer); +COMP_EXIT: + compress_close(compress); +FILE_EXIT: + fclose(file); + if (verbose) + printf("%s: exit failure\n", __func__); + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, rate = 16000; + + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "rhvb:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'r': + rate = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, rate); + + fprintf(stderr, "Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 60d6c04..9404fd8 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c +RECSRC = crec.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

3 2

[alsa-devel] [PATCH] ASoC: fsl: expand the size of the name in fsl_ssi_private struct
by Nicolin Chen 23 May '13

23 May '13

strcpy(ssi_private->name, p) in probe() sets "name" by "p", gotten from dts, while the length of "p", if the devicetree node name of SSI is commonly set, would always be larger than 1 char size, so need a larger size for "name". Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6a5f94 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -152,5 +152,5 @@ struct fsl_ssi_private { } stats; - char name[1]; + char name[32]; }; -- 1.7.1

1 0

[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: fix hpointer error when no sample rate
by Richard Fitzgerald 23 May '13

23 May '13

Fixes the oops() in compress_get_hpointer() to return -ENODATA instead of errno if the sample rate is zero. The value in errno here is not related to the reason we are returning an error. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/compress.c b/compress.c index e80e598..abd0799 100644 --- a/compress.c +++ b/compress.c @@ -315,7 +315,7 @@ int compress_get_hpointer(struct compress *compress, if (ioctl(compress->fd, SNDRV_COMPRESS_AVAIL, &kavail)) return oops(compress, errno, "cannot get avail"); if (0 == kavail.tstamp.sampling_rate) - return oops(compress, errno, "invalid paramter"); + return oops(compress, -ENODATA, "sample rate unknown"); *avail = (unsigned int)kavail.avail; time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; tstamp->tv_sec = time; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: Fix early event callback list iteration
by Lars-Peter Clausen 23 May '13

23 May '13

The power_list field is used when adding a widget to a power sequence list. Use the same field when iterating the list using list_for_each_entry, otherwise we'll see undefined behavior. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 071579b..3507346 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1738,11 +1738,11 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); - list_for_each_entry(w, &down_list, list) { + list_for_each_entry(w, &down_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMD); } - list_for_each_entry(w, &up_list, list) { + list_for_each_entry(w, &up_list, power_list) { dapm_seq_check_event(dapm, w, SND_SOC_DAPM_WILL_PMU); } -- 1.8.2.1

2 1

[alsa-devel] [PATCH] ASoC: Add ssm2518 support
by Lars-Peter Clausen 23 May '13

23 May '13

This patch adds a ASoC CODEC driver for the SSM2516. The SSM2516 is a stereo Class-D audio amplifier with an I2S interface for audio in and a built-in dynamic range control processor. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/ssm2518.txt | 20 + include/linux/platform_data/ssm2518.h | 22 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2518.c | 845 +++++++++++++++++++++ sound/soc/codecs/ssm2518.h | 20 + 6 files changed, 913 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h diff --git a/Documentation/devicetree/bindings/sound/ssm2518.txt b/Documentation/devicetree/bindings/sound/ssm2518.txt new file mode 100644 index 0000000..59381a7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm2518.txt @@ -0,0 +1,20 @@ +SSM2518 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm2518" + - reg : the I2C address of the device. This will either be 0x34 (ADDR pin low) + or 0x35 (ADDR pin high) + +Optional properties: + - gpios : GPIO connected to the nSD pin. If the property is not present it is + assumed that the nSD pin is hardwired to always on. + +Example: + + ssm2518: ssm2518@34 { + compatible = "adi,ssm2518"; + reg = <0x34>; + gpios = <&gpio 5 0>; + }; diff --git a/include/linux/platform_data/ssm2518.h b/include/linux/platform_data/ssm2518.h new file mode 100644 index 0000000..9a8e3ea --- /dev/null +++ b/include/linux/platform_data/ssm2518.h @@ -0,0 +1,22 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_SSM2518_H__ +#define __LINUX_PLATFORM_DATA_SSM2518_H__ + +/** + * struct ssm2518_platform_data - Platform data for the ssm2518 driver + * @enable_gpio: GPIO connected to the nSD pin. Set to -1 if the nSD pin is + * hardwired. + */ +struct ssm2518_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d76609a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF + select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C @@ -313,6 +314,9 @@ config SND_SOC_SN95031 config SND_SOC_SPDIF tristate +config SND_SOC_SSM2518 + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dae0aa6..7204c24 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-si476x-objs := si476x.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o @@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c new file mode 100644 index 0000000..5874ba5 --- /dev/null +++ b/sound/soc/codecs/ssm2518.c @@ -0,0 +1,845 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/platform_data/ssm2518.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ssm2518.h" + +#define SSM2518_REG_POWER1 0x00 +#define SSM2518_REG_CLOCK 0x01 +#define SSM2518_REG_SAI_CTRL1 0x02 +#define SSM2518_REG_SAI_CTRL2 0x03 +#define SSM2518_REG_CHAN_MAP 0x04 +#define SSM2518_REG_LEFT_VOL 0x05 +#define SSM2518_REG_RIGHT_VOL 0x06 +#define SSM2518_REG_MUTE_CTRL 0x07 +#define SSM2518_REG_FAULT_CTRL 0x08 +#define SSM2518_REG_POWER2 0x09 +#define SSM2518_REG_DRC_1 0x0a +#define SSM2518_REG_DRC_2 0x0b +#define SSM2518_REG_DRC_3 0x0c +#define SSM2518_REG_DRC_4 0x0d +#define SSM2518_REG_DRC_5 0x0e +#define SSM2518_REG_DRC_6 0x0f +#define SSM2518_REG_DRC_7 0x10 +#define SSM2518_REG_DRC_8 0x11 +#define SSM2518_REG_DRC_9 0x12 + +#define SSM2518_POWER1_RESET BIT(7) +#define SSM2518_POWER1_NO_BCLK BIT(5) +#define SSM2518_POWER1_MCS_MASK (0xf << 1) +#define SSM2518_POWER1_MCS_64FS (0x0 << 1) +#define SSM2518_POWER1_MCS_128FS (0x1 << 1) +#define SSM2518_POWER1_MCS_256FS (0x2 << 1) +#define SSM2518_POWER1_MCS_384FS (0x3 << 1) +#define SSM2518_POWER1_MCS_512FS (0x4 << 1) +#define SSM2518_POWER1_MCS_768FS (0x5 << 1) +#define SSM2518_POWER1_MCS_100FS (0x6 << 1) +#define SSM2518_POWER1_MCS_200FS (0x7 << 1) +#define SSM2518_POWER1_MCS_400FS (0x8 << 1) +#define SSM2518_POWER1_SPWDN BIT(0) + +#define SSM2518_CLOCK_ASR BIT(0) + +#define SSM2518_SAI_CTRL1_FMT_MASK (0x3 << 5) +#define SSM2518_SAI_CTRL1_FMT_I2S (0x0 << 5) +#define SSM2518_SAI_CTRL1_FMT_LJ (0x1 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_24BIT (0x2 << 5) +#define SSM2518_SAI_CTRL1_FMT_RJ_16BIT (0x3 << 5) + +#define SSM2518_SAI_CTRL1_SAI_MASK (0x7 << 2) +#define SSM2518_SAI_CTRL1_SAI_I2S (0x0 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_2 (0x1 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_4 (0x2 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_8 (0x3 << 2) +#define SSM2518_SAI_CTRL1_SAI_TDM_16 (0x4 << 2) +#define SSM2518_SAI_CTRL1_SAI_MONO (0x5 << 2) + +#define SSM2518_SAI_CTRL1_FS_MASK (0x3) +#define SSM2518_SAI_CTRL1_FS_8000_12000 (0x0) +#define SSM2518_SAI_CTRL1_FS_16000_24000 (0x1) +#define SSM2518_SAI_CTRL1_FS_32000_48000 (0x2) +#define SSM2518_SAI_CTRL1_FS_64000_96000 (0x3) + +#define SSM2518_SAI_CTRL2_BCLK_INTERAL BIT(7) +#define SSM2518_SAI_CTRL2_LRCLK_PULSE BIT(6) +#define SSM2518_SAI_CTRL2_LRCLK_INVERT BIT(5) +#define SSM2518_SAI_CTRL2_MSB BIT(4) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK (0x3 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_32 (0x0 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_24 (0x1 << 2) +#define SSM2518_SAI_CTRL2_SLOT_WIDTH_16 (0x2 << 2) +#define SSM2518_SAI_CTRL2_BCLK_INVERT BIT(1) + +#define SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET 4 +#define SSM2518_CHAN_MAP_RIGHT_SLOT_MASK 0xf0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET 0 +#define SSM2518_CHAN_MAP_LEFT_SLOT_MASK 0x0f + +#define SSM2518_MUTE_CTRL_ANA_GAIN BIT(5) +#define SSM2518_MUTE_CTRL_MUTE_MASTER BIT(0) + +#define SSM2518_POWER2_APWDN BIT(0) + +#define SSM2518_DAC_MUTE BIT(6) +#define SSM2518_DAC_FS_MASK 0x07 +#define SSM2518_DAC_FS_8000 0x00 +#define SSM2518_DAC_FS_16000 0x01 +#define SSM2518_DAC_FS_32000 0x02 +#define SSM2518_DAC_FS_64000 0x03 +#define SSM2518_DAC_FS_128000 0x04 + +struct ssm2518 { + struct regmap *regmap; + bool right_j; + + unsigned int sysclk; + const struct snd_pcm_hw_constraint_list *constraints; + + int enable_gpio; +}; + +static const struct reg_default ssm2518_reg_defaults[] = { + { 0x00, 0x05 }, + { 0x01, 0x00 }, + { 0x02, 0x02 }, + { 0x03, 0x00 }, + { 0x04, 0x10 }, + { 0x05, 0x40 }, + { 0x06, 0x40 }, + { 0x07, 0x81 }, + { 0x08, 0x0c }, + { 0x09, 0x99 }, + { 0x0a, 0x7c }, + { 0x0b, 0x5b }, + { 0x0c, 0x57 }, + { 0x0d, 0x89 }, + { 0x0e, 0x8c }, + { 0x0f, 0x77 }, + { 0x10, 0x26 }, + { 0x11, 0x1c }, + { 0x12, 0x97 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm2518_vol_tlv, -7125, 2400); +static const DECLARE_TLV_DB_SCALE(ssm2518_compressor_tlv, -3400, 200, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_expander_tlv, -8100, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_noise_gate_tlv, -9600, 300, 0); +static const DECLARE_TLV_DB_SCALE(ssm2518_post_drc_tlv, -2400, 300, 0); + +static const DECLARE_TLV_DB_RANGE(ssm2518_limiter_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-2200, 200, 0), + 7, 15, TLV_DB_SCALE_ITEM(-800, 100, 0), +); + +static const char * const ssm2518_drc_peak_detector_attack_time_text[] = { + "0 ms", "0.1 ms", "0.19 ms", "0.37 ms", "0.75 ms", "1.5 ms", "3 ms", + "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", "192 ms", "384 ms", + "768 ms", "1536 ms", +}; + +static const char * const ssm2518_drc_peak_detector_release_time_text[] = { + "0 ms", "1.5 ms", "3 ms", "6 ms", "12 ms", "24 ms", "48 ms", "96 ms", + "192 ms", "384 ms", "768 ms", "1536 ms", "3072 ms", "6144 ms", + "12288 ms", "24576 ms" +}; + +static const char * const ssm2518_drc_hold_time_text[] = { + "0 ms", "0.67 ms", "1.33 ms", "2.67 ms", "5.33 ms", "10.66 ms", + "21.32 ms", "42.64 ms", "85.28 ms", "170.56 ms", "341.12 ms", + "682.24 ms", "1364 ms", +}; + +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, + SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, + SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, + SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, + SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, + SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, + SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); +static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, + SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); + +static const struct snd_kcontrol_new ssm2518_snd_controls[] = { + SOC_SINGLE("Playback De-emphasis Switch", SSM2518_REG_MUTE_CTRL, + 4, 1, 0), + SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2518_REG_LEFT_VOL, + SSM2518_REG_RIGHT_VOL, 0, 0xff, 1, ssm2518_vol_tlv), + SOC_DOUBLE("Master Playback Switch", SSM2518_REG_MUTE_CTRL, 2, 1, 1, 1), + + SOC_SINGLE("Amp Low Power Mode Switch", SSM2518_REG_POWER2, 4, 1, 0), + SOC_SINGLE("DAC Low Power Mode Switch", SSM2518_REG_POWER2, 3, 1, 0), + + SOC_SINGLE("DRC Limiter Switch", SSM2518_REG_DRC_1, 5, 1, 0), + SOC_SINGLE("DRC Compressor Switch", SSM2518_REG_DRC_1, 4, 1, 0), + SOC_SINGLE("DRC Expander Switch", SSM2518_REG_DRC_1, 3, 1, 0), + SOC_SINGLE("DRC Noise Gate Switch", SSM2518_REG_DRC_1, 2, 1, 0), + SOC_DOUBLE("DRC Switch", SSM2518_REG_DRC_1, 0, 1, 1, 0), + + SOC_SINGLE_TLV("DRC Limiter Threshold Volume", + SSM2518_REG_DRC_3, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Compressor Lower Threshold Volume", + SSM2518_REG_DRC_3, 0, 15, 1, ssm2518_compressor_tlv), + SOC_SINGLE_TLV("DRC Expander Upper Threshold Volume", SSM2518_REG_DRC_4, + 4, 15, 1, ssm2518_expander_tlv), + SOC_SINGLE_TLV("DRC Noise Gate Threshold Volume", + SSM2518_REG_DRC_4, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Upper Output Threshold Volume", + SSM2518_REG_DRC_5, 4, 15, 1, ssm2518_limiter_tlv), + SOC_SINGLE_TLV("DRC Lower Output Threshold Volume", + SSM2518_REG_DRC_5, 0, 15, 1, ssm2518_noise_gate_tlv), + SOC_SINGLE_TLV("DRC Post Volume", SSM2518_REG_DRC_8, + 2, 15, 1, ssm2518_post_drc_tlv), + + SOC_ENUM("DRC Peak Detector Attack Time", + ssm2518_drc_peak_detector_attack_time_enum), + SOC_ENUM("DRC Peak Detector Release Time", + ssm2518_drc_peak_detector_release_time_enum), + SOC_ENUM("DRC Attack Time", ssm2518_drc_attack_time_enum), + SOC_ENUM("DRC Decay Time", ssm2518_drc_decay_time_enum), + SOC_ENUM("DRC Hold Time", ssm2518_drc_hold_time_enum), + SOC_ENUM("DRC Noise Gate Hold Time", + ssm2518_drc_noise_gate_hold_time_enum), + SOC_ENUM("DRC RMS Averaging Time", ssm2518_drc_rms_averaging_time_enum), +}; + +static const struct snd_soc_dapm_widget ssm2518_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SSM2518_REG_POWER2, 1, 1), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SSM2518_REG_POWER2, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route ssm2518_routes[] = { + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +struct ssm2518_mcs_lut { + unsigned int rate; + const unsigned int *sysclks; +}; + +static const unsigned int ssm2518_sysclks_2048000[] = { + 2048000, 4096000, 8192000, 12288000, 16384000, 24576000, + 3200000, 6400000, 12800000, 0 +}; + +static const unsigned int ssm2518_sysclks_2822000[] = { + 2822000, 5644800, 11289600, 16934400, 22579200, 33868800, + 4410000, 8820000, 17640000, 0 +}; + +static const unsigned int ssm2518_sysclks_3072000[] = { + 3072000, 6144000, 12288000, 16384000, 24576000, 38864000, + 4800000, 9600000, 19200000, 0 +}; + +static const struct ssm2518_mcs_lut ssm2518_mcs_lut[] = { + { 8000, ssm2518_sysclks_2048000, }, + { 11025, ssm2518_sysclks_2822000, }, + { 12000, ssm2518_sysclks_3072000, }, + { 16000, ssm2518_sysclks_2048000, }, + { 24000, ssm2518_sysclks_3072000, }, + { 22050, ssm2518_sysclks_2822000, }, + { 32000, ssm2518_sysclks_2048000, }, + { 44100, ssm2518_sysclks_2822000, }, + { 48000, ssm2518_sysclks_3072000, }, + { 96000, ssm2518_sysclks_3072000, }, +}; + +static const unsigned int ssm2518_rates_2048000[] = { + 8000, 16000, 32000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2048000 = { + .list = ssm2518_rates_2048000, + .count = ARRAY_SIZE(ssm2518_rates_2048000), +}; + +static const unsigned int ssm2518_rates_2822000[] = { + 11025, 22050, 44100, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_2822000 = { + .list = ssm2518_rates_2822000, + .count = ARRAY_SIZE(ssm2518_rates_2822000), +}; + +static const unsigned int ssm2518_rates_3072000[] = { + 12000, 24000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_3072000 = { + .list = ssm2518_rates_3072000, + .count = ARRAY_SIZE(ssm2518_rates_3072000), +}; + +static const unsigned int ssm2518_rates_12288000[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { + .list = ssm2518_rates_12288000, + .count = ARRAY_SIZE(ssm2518_rates_12288000), +}; + +static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, + unsigned int rate) +{ + const unsigned int *sysclks = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(ssm2518_mcs_lut); i++) { + if (ssm2518_mcs_lut[i].rate == rate) { + sysclks = ssm2518_mcs_lut[i].sysclks; + break; + } + } + if (!sysclks) + return -EINVAL; + + for (i = 0; sysclks[i]; i++) { + if (sysclks[i] == ssm2518->sysclk) + return i; + } + return -EINVAL; +} + +static int ssm2518_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 ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int ctrl1, ctrl1_mask; + int mcs; + int ret; + + mcs = ssm2518_lookup_mcs(ssm2518, rate); + if (mcs < 0) + return mcs; + + ctrl1_mask = SSM2518_SAI_CTRL1_FS_MASK; + + if (rate >= 8000 && rate <= 12000) + ctrl1 = SSM2518_SAI_CTRL1_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + ctrl1 = SSM2518_SAI_CTRL1_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + ctrl1 = SSM2518_SAI_CTRL1_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + ctrl1 = SSM2518_SAI_CTRL1_FS_64000_96000; + else + return -EINVAL; + + if (ssm2518->right_j) { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl1_mask |= SSM2518_SAI_CTRL1_FMT_MASK; + } + + /* Disable auto samplerate detection */ + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_CLOCK, + SSM2518_CLOCK_ASR, SSM2518_CLOCK_ASR); + if (ret < 0) + return ret; + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + ctrl1_mask, ctrl1); + if (ret < 0) + return ret; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_MCS_MASK, mcs << 1); +} + +static int ssm2518_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = SSM2518_MUTE_CTRL_MUTE_MASTER; + else + val = 0; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_MUTE_CTRL, + SSM2518_MUTE_CTRL_MUTE_MASTER, val); +} + +static int ssm2518_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1 = 0, ctrl2 = 0; + bool invert_fclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl2 |= SSM2518_SAI_CTRL2_BCLK_INVERT; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + ssm2518->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT; + ssm2518->right_j = true; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_I2S; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_PULSE; + ctrl1 |= SSM2518_SAI_CTRL1_FMT_LJ; + invert_fclk = false; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl2 |= SSM2518_SAI_CTRL2_LRCLK_INVERT; + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, ctrl1); + if (ret) + return ret; + return regmap_write(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, ctrl2); +} + +static int ssm2518_set_power(struct ssm2518 *ssm2518, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN, SSM2518_POWER1_SPWDN); + regcache_mark_dirty(ssm2518->regmap); + } + + if (gpio_is_valid(ssm2518->enable_gpio)) + gpio_set_value(ssm2518->enable_gpio, enable); + + regcache_cache_only(ssm2518->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_SPWDN | SSM2518_POWER1_RESET, 0x00); + regcache_sync(ssm2518->regmap); + } + + return ret; +} + +static int ssm2518_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm2518_set_power(ssm2518, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm2518_set_power(ssm2518, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + return 0; +} + +static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl1, ctrl2; + int left_slot, right_slot; + int ret; + + if (slots == 0) { + return regmap_update_bits(ssm2518->regmap, + SSM2518_REG_SAI_CTRL1, SSM2518_SAI_CTRL1_SAI_MASK, + SSM2518_SAI_CTRL1_SAI_I2S); + } + + if (tx_mask == 0 || tx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + /* We assume the left channel < right channel */ + left_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = ffs(tx_mask); + tx_mask &= ~(1 << tx_mask); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (width) { + case 16: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_16; + break; + case 24: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_24; + break; + case 32: + ctrl2 = SSM2518_SAI_CTRL2_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 1: + ctrl1 = SSM2518_SAI_CTRL1_SAI_MONO; + break; + case 2: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_2; + break; + case 4: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_4; + break; + case 8: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_8; + break; + case 16: + ctrl1 = SSM2518_SAI_CTRL1_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_write(ssm2518->regmap, SSM2518_REG_CHAN_MAP, + (left_slot << SSM2518_CHAN_MAP_LEFT_SLOT_OFFSET) | + (right_slot << SSM2518_CHAN_MAP_RIGHT_SLOT_OFFSET)); + if (ret) + return ret; + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL1, + SSM2518_SAI_CTRL1_SAI_MASK, ctrl1); + if (ret) + return ret; + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_SAI_CTRL2, + SSM2518_SAI_CTRL2_SLOT_WIDTH_MASK, ctrl2); +} + +static int ssm2518_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(dai->codec); + + if (ssm2518->constraints) { + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, ssm2518->constraints); + } + + return 0; +} + +#define SSM2518_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm2518_dai_ops = { + .startup = ssm2518_startup, + .hw_params = ssm2518_hw_params, + .digital_mute = ssm2518_mute, + .set_fmt = ssm2518_set_dai_fmt, + .set_tdm_slot = ssm2518_set_tdm_slot, +}; + +static struct snd_soc_dai_driver ssm2518_dai = { + .name = "ssm2518-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SSM2518_FORMATS, + }, + .ops = &ssm2518_dai_ops, +}; + +static int ssm2518_probe(struct snd_soc_codec *codec) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = ssm2518->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER2, + SSM2518_POWER2_APWDN, 0x00); + if (ret) + return ret; + + return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int ssm2518_remove(struct snd_soc_codec *codec) +{ + ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (clk_id != SSM2518_SYSCLK) + return -EINVAL; + + switch (source) { + case SSM2518_SYSCLK_SRC_MCLK: + val = 0; + break; + case SSM2518_SYSCLK_SRC_BCLK: + /* In this case the bitclock is used as the system clock, and + * the bitclock signal needs to be connected to the MCLK pin and + * the BCLK pin is left unconnected */ + val = SSM2518_POWER1_NO_BCLK; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 2048000: + case 4096000: + case 8192000: + case 3200000: + case 6400000: + case 12800000: + ssm2518->constraints = &ssm2518_constraints_2048000; + break; + case 2822000: + case 5644800: + case 11289600: + case 16934400: + case 22579200: + case 33868800: + case 4410000: + case 8820000: + case 17640000: + ssm2518->constraints = &ssm2518_constraints_2822000; + break; + case 3072000: + case 6144000: + case 38864000: + case 4800000: + case 9600000: + case 19200000: + ssm2518->constraints = &ssm2518_constraints_3072000; + break; + case 12288000: + case 16384000: + case 24576000: + ssm2518->constraints = &ssm2518_constraints_12288000; + break; + default: + return -EINVAL; + } + + ssm2518->sysclk = freq; + + return regmap_update_bits(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_NO_BCLK, val); +} + +static struct snd_soc_codec_driver ssm2518_codec_driver = { + .probe = ssm2518_probe, + .remove = ssm2518_remove, + .set_bias_level = ssm2518_set_bias_level, + .set_sysclk = ssm2518_set_sysclk, + .idle_bias_off = true, + + .controls = ssm2518_snd_controls, + .num_controls = ARRAY_SIZE(ssm2518_snd_controls), + .dapm_widgets = ssm2518_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm2518_dapm_widgets), + .dapm_routes = ssm2518_routes, + .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), +}; + +static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config ssm2518_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM2518_REG_DRC_9, + .volatile_reg = ssm2518_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm2518_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm2518_reg_defaults), +}; + +static int ssm2518_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm2518_platform_data *pdata = i2c->dev.platform_data; + struct ssm2518 *ssm2518; + int ret; + + ssm2518 = devm_kzalloc(&i2c->dev, sizeof(*ssm2518), GFP_KERNEL); + if (ssm2518 == NULL) + return -ENOMEM; + + if (pdata) { + ssm2518->enable_gpio = pdata->enable_gpio; + } else if (i2c->dev.of_node) { + ssm2518->enable_gpio = of_get_gpio(i2c->dev.of_node, 0); + if (ssm2518->enable_gpio == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + ssm2518->enable_gpio = -1; + } + + if (gpio_is_valid(ssm2518->enable_gpio)) { + ret = devm_gpio_request_one(&i2c->dev, ssm2518->enable_gpio, + GPIOF_OUT_INIT_HIGH, "SSM2518 nSD"); + if (ret) + return ret; + } + + i2c_set_clientdata(i2c, ssm2518); + + ssm2518->regmap = devm_regmap_init_i2c(i2c, &ssm2518_regmap_config); + if (IS_ERR(ssm2518->regmap)) + return PTR_ERR(ssm2518->regmap); + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(ssm2518->regmap, true); + ret = regmap_write(ssm2518->regmap, SSM2518_REG_POWER1, + SSM2518_POWER1_RESET); + if (ret) + return ret; + regcache_cache_bypass(ssm2518->regmap, false); + + return snd_soc_register_codec(&i2c->dev, &ssm2518_codec_driver, + &ssm2518_dai, 1); +} + +static int ssm2518_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2518_i2c_ids[] = { + { "ssm2518", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); + +static struct i2c_driver ssm2518_driver = { + .driver = { + .name = "ssm2518", + .owner = THIS_MODULE, + }, + .probe = ssm2518_i2c_probe, + .remove = ssm2518_i2c_remove, + .id_table = ssm2518_i2c_ids, +}; +module_i2c_driver(ssm2518_driver); + +MODULE_DESCRIPTION("ASoC SSM2518 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2518.h b/sound/soc/codecs/ssm2518.h new file mode 100644 index 0000000..62511d8 --- /dev/null +++ b/sound/soc/codecs/ssm2518.h @@ -0,0 +1,20 @@ +/* + * SSM2518 amplifier audio driver + * + * Copyright 2013 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SND_SOC_CODECS_SSM2518_H__ +#define __SND_SOC_CODECS_SSM2518_H__ + +#define SSM2518_SYSCLK 0 + +enum ssm2518_sysclk_src { + SSM2518_SYSCLK_SRC_MCLK = 0, + SSM2518_SYSCLK_SRC_BCLK = 1, +}; + +#endif -- 1.8.0

2 3

[alsa-devel] [PATCH] ALSA: ASoC: add codec driver for TI TAS5086
by Daniel Mack 22 May '13

22 May '13

This patch adds a driver for TI's TA5086 6-channel PWM processor. This chip has a very unusual register layout, specifically because the registers are of unequal size, and multi-byte registers require bulk writes to take effect. Regmap does not support these kind of mappings. Currently, the driver does not touch any of the registers >= 0x20, so it doesn't matter, because the register map is mapped to an 8-bit array. In case more features will be added in the future that require access to higher registers, the entire regmap H/W I/O routines have to be open-coded. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5086.c | 587 +++++++++++++++++++++ 4 files changed, 625 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 sound/soc/codecs/tas5086.c diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 0000000..8ea4f5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,32 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b7256..86b3524 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -63,6 +63,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C @@ -320,6 +321,9 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS5086 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3..8077bc2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o +snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -177,6 +178,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c new file mode 100644 index 0000000..cd3c267 --- /dev/null +++ b/sound/soc/codecs/tas5086.c @@ -0,0 +1,587 @@ +/* + * TAS5086 ASoC codec driver + * + * Copyright (c) 2013 Daniel Mack <zonque(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * TODO: + * - implement DAPM and input muxing + * - implement modulation limit + * - implement non-default PWM start + * + * Note that this chip has a very unusual register layout, specifically + * because the registers are of unequal size, and multi-byte registers + * require bulk writes to take effect. Regmap does not support that kind + * of devices. + * + * Currently, the driver does not touch any of the registers >= 0x20, so + * it doesn't matter because the entire map can be accessed as 8-bit + * array. In case more features will be added in the future + * that require access to higher registers, the entire regmap H/W I/O + * routines have to be open-coded. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE) + +#define TAS5086_PCM_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +/* + * TAS5086 registers + */ +#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */ +#define TAS5086_CLOCK_RATE(val) (val << 5) +#define TAS5086_CLOCK_RATE_MASK (0x7 << 5) +#define TAS5086_CLOCK_RATIO(val) (val << 2) +#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2) +#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1) +#define TAS5086_CLOCK_VALID (1 << 0) + +#define TAS5086_DEV_ID 0x01 /* Device ID register */ +#define TAS5086_ERROR_STATUS 0x02 /* Error status register */ +#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */ +#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */ +#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */ +#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */ +#define TAS5086_MASTER_VOL 0x07 /* Master volume */ +#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */ +#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */ +#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */ +#define TAS5086_PWM_START 0x18 /* PWM start register */ +#define TAS5086_SURROUND 0x19 /* Surround register */ +#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ +#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ +#define TAS5086_BKNDERR 0x1c + +#define TAS5086_DEEMPH_MASK 0x03 + +/* + * Default TAS5086 power-up configuration + */ +static const struct reg_default tas5086_reg_defaults[] = { + { 0x00, 0x6c }, + { 0x01, 0x03 }, + { 0x02, 0x00 }, + { 0x03, 0xa0 }, + { 0x04, 0x05 }, + { 0x05, 0x60 }, + { 0x06, 0x00 }, + { 0x07, 0xff }, + { 0x08, 0x30 }, + { 0x09, 0x30 }, + { 0x0a, 0x30 }, + { 0x0b, 0x30 }, + { 0x0c, 0x30 }, + { 0x0d, 0x30 }, + { 0x0e, 0xb1 }, + { 0x0f, 0x00 }, + { 0x10, 0x02 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x3f }, + { 0x19, 0x00 }, + { 0x1a, 0x18 }, + { 0x1b, 0x82 }, + { 0x1c, 0x05 }, +}; + +static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); +} + +static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == TAS5086_DEV_ID || + reg == TAS5086_ERROR_STATUS; +} + +static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) +{ + return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); +} + +struct tas5086_private { + struct regmap *regmap; + unsigned int mclk, sclk; + unsigned int format; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; +}; + +static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; + +static int tas5086_set_deemph(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int i, val = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) + if (tas5086_deemph[i] == priv->rate) + val = i; + + return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, + TAS5086_DEEMPH_MASK, val); +} + +static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return tas5086_set_deemph(codec); +} + + +static int tas5086_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case 0: /* MCLK */ + priv->mclk = freq; + break; + case 1: /* SCLK */ + priv->sclk = freq; + break; + } + + return 0; +} + +static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The TAS5086 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + /* we need to refer to the data format from hw_params() */ + priv->format = format; + + return 0; +} + +static const int tas5086_sample_rates[] = { + 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const int tas5086_ratios[] = { + 64, 128, 192, 256, 384, 512 +}; + +static int index_in_array(const int *array, int len, int needle) +{ + int i; + + for (i = 0; i < len; i++) + if (array[i] == needle) + return i; + + return -ENOENT; +} + +static int tas5086_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 tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + priv->rate = params_rate(params); + + /* Look up the sample rate and refer to the offset in the list */ + val = index_in_array(tas5086_sample_rates, + ARRAY_SIZE(tas5086_sample_rates), priv->rate); + + if (val < 0) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATE_MASK, + TAS5086_CLOCK_RATE(val)); + if (ret < 0) + return ret; + + /* MCLK / Fs ratio */ + val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), + priv->mclk / priv->rate); + if (val < 0) { + dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATIO_MASK, + TAS5086_CLOCK_RATIO(val)); + if (ret < 0) + return ret; + + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_SCLK_RATIO_48, + (priv->sclk == 48 * priv->rate) ? + TAS5086_CLOCK_SCLK_RATIO_48 : 0); + if (ret < 0) + return ret; + + /* + * The chip has a very unituitive register mapping and muxes information + * about data format and sample depth into the same register, but not on + * a logical bit-boundary. Hence, we have to refer to the format passed + * in the set_dai_fmt() callback and set up everything from here. + * + * First, determine the 'base' value, using the format ... + */ + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x03; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x06; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* ... then add the offset for the sample bit depth. */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val += 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val += 1; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + val += 2; + break; + default: + dev_err(codec->dev, "Invalid bit width\n"); + return -EINVAL; + }; + + ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); + if (ret < 0) + return ret; + + /* clock is considered valid now */ + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID); + if (ret < 0) + return ret; + + return tas5086_set_deemph(codec); +} + +static int tas5086_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + mute ? 0x3f : 0x00); +} + +/* TAS5086 controls */ +static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas5086_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL, + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + tas5086_get_deemph, tas5086_put_deemph), +}; + +static const struct snd_soc_dai_ops tas5086_dai_ops = { + .hw_params = tas5086_hw_params, + .set_sysclk = tas5086_set_dai_sysclk, + .set_fmt = tas5086_set_dai_fmt, + .digital_mute = tas5086_digital_mute, +}; + +static struct snd_soc_dai_driver tas5086_dai = { + .name = "tas5086-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = TAS5086_PCM_RATES, + .formats = TAS5086_PCM_FORMATS, + }, + .ops = &tas5086_dai_ops, +}; + +#ifdef CONFIG_PM +static int tas5086_soc_suspend(struct snd_soc_codec *codec) +{ + return 0; +} + +static int tas5086_soc_resume(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* Restore codec state */ + return regcache_sync(priv->regmap); +} +#else +#define tas5086_soc_suspend NULL +#define tas5086_soc_resume NULL +#endif /* CONFIG_PM */ + +#ifdef CONFIG_OF +static const struct of_device_id tas5086_dt_ids[] = { + { .compatible = "ti,tas5086", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5086_dt_ids); +#endif + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_probe(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int charge_period = 1300000; /* hardware default is 1300 ms */ + int gpio_nreset = -EINVAL; + int i, ret; + + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { + struct device_node *of_node = codec->dev->of_node; + + gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0); + of_property_read_u32(of_node, "ti,charge-period", &charge_period); + } + + if (gpio_nreset >= 0) + if (devm_gpio_request(codec->dev, gpio_nreset, "TAS5086 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } +msleep(300); + priv->gpio_nreset = gpio_nreset; + + /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ + ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); + if (ret < 0) + return ret; + + if (i != 0x3) { + dev_err(codec->dev, + "Failed to identify TAS5086 codec (got %02x)\n", i); + return -ENODEV; + } + + /* lookup and set split-capacitor charge period */ + if (charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(codec->dev, + "Invalid split-cap charge period of %d ns.\n", + charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* set master volume to 0 dB */ + ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); + if (ret < 0) + return ret; + + return 0; +} + +static int tas5086_remove(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->gpio_nreset)) + /* Set codec to the reset state */ + gpio_set_value(priv->gpio_nreset, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { + .probe = tas5086_probe, + .remove = tas5086_remove, + .suspend = tas5086_soc_suspend, + .resume = tas5086_soc_resume, + .controls = tas5086_controls, + .num_controls = ARRAY_SIZE(tas5086_controls), +}; + +static const struct i2c_device_id tas5086_i2c_id[] = { + { "tas5086", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); + +static const struct regmap_config tas5086_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .reg_defaults = tas5086_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5086_volatile_reg, + .writeable_reg = tas5086_writeable_reg, + .readable_reg = tas5086_accessible_reg, +}; + +static int tas5086_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tas5086_private *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); +} + +static int tas5086_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver tas5086_i2c_driver = { + .driver = { + .name = "tas5086", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas5086_dt_ids), + }, + .id_table = tas5086_i2c_id, + .probe = tas5086_i2c_probe, + .remove = tas5086_i2c_remove, +}; + +static int __init tas5086_modinit(void) +{ + return i2c_add_driver(&tas5086_i2c_driver); +} +module_init(tas5086_modinit); + +static void __exit tas5086_modexit(void) +{ + i2c_del_driver(&tas5086_i2c_driver); +} +module_exit(tas5086_modexit); + +MODULE_AUTHOR("Daniel Mack <zonque(a)gmail.com>"); +MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- 1.8.1.4

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Re: [alsa-devel] [PATCH 3/3] sound/soc/fsl: remove leftover release_mem_region
by Mark Brown 22 May '13

22 May '13

On Wed, May 22, 2013 at 06:04:00PM +0200, Wolfram Sang wrote: > On Wed, May 22, 2013 at 10:56:28AM -0500, Mark Brown wrote: > > This doesn't apply against current code, can you please regenerate > > against topic/fsl or mainline? > ? Works fine against v3.10-rc2 here. Or is mainline v3.9 for you? To be honest I didn't actually try mainline as I wasn't expecting any conflicts from the branch and it didn't seem urgent as a bug fix.

1 0

[alsa-devel] [PATCH v4] ASoC: dfbmcs320: make the driver common for other BT modules
by Barry Song 22 May '13

22 May '13

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- v4: change "select SND_SOC_DFBMCS320" to "select SND_SOC_BT_SCO" for OpenMOKO user as v3 missed this sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++----------- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- 5 files changed, 29 insertions(+), 20 deletions(-) rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d8c4f3d..386292d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49ff127..82aa34e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/bt-sco.c similarity index 53% rename from sound/soc/codecs/dfbmcs320.c rename to sound/soc/codecs/bt-sco.c index 4f4f7f4..a081d9f 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/bt-sco.c @@ -1,5 +1,5 @@ /* - * Driver for the DFBM-CS320 bluetooth module + * Driver for generic Bluetooth SCO link * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it @@ -15,8 +15,8 @@ #include <sound/soc.h> -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", .playback = { .channels_min = 1, .channels_max = 1, @@ -31,32 +31,41 @@ static struct snd_soc_dai_driver dfbmcs320_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; -static int dfbmcs320_probe(struct platform_device *pdev) +static int bt_sco_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); } -static int dfbmcs320_remove(struct platform_device *pdev) +static int bt_sco_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver dfmcs320_driver = { +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { .driver = { - .name = "dfbmcs320", + .name = "bt-sco", .owner = THIS_MODULE, }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, }; -module_platform_driver(dfmcs320_driver); +module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 475fb0d..ae0ea87 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,7 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_DFBMCS320 + select SND_SOC_SCO help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- 1.8.2.3

3 4

Re: [alsa-devel] [PATCH 3/3] sound/soc/fsl: remove leftover release_mem_region
by Mark Brown 22 May '13

22 May '13

On Wed, May 22, 2013 at 03:21:26PM +0200, Wolfram Sang wrote: > When converting this driver to devm_ioremap_resource, the removal of this now > unneeded function has been forgotten. This doesn't apply against current code, can you please regenerate against topic/fsl or mainline?

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang Xingchao 22 May '13

22 May '13

Haswell doesnot support runtime pm by default. This patch let haswell Display HD-A controller enter runtime suspend, and bring more power saving whith power-well. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bf27693..eb25888 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + else if (chip->driver_caps + & AZX_DCAPS_I915_POWERWELL) { + /* Haswell doesnot support runtime pm by default */ + pm_runtime_put_noidle(&pci->dev); + pm_runtime_allow(&pci->dev); + } dev++; complete_all(&chip->probe_wait); -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: davinci: fix sample rotation
by Daniel Mack 21 May '13

21 May '13

McASP serial audio engine needs different rotation values on TX and RX channels. Commit dde109fb462 ("ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback") changed the calculation to fix the playback format, but broke the capture stream by doing it for both TXFMT and RXFMT. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Michal Bachraty <michal.bachraty(a)streamunlimited.com> Cc: stable(a)vger.kernel.org [3.9 only] --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 63633e9..8d805ec 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -628,7 +628,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (word_length / 4) & 0x7; + u32 tx_rotate = (word_length / 4) & 0x7; + u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; /* @@ -651,9 +652,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(rotate), TXROT(7)); + TXROT(tx_rotate), TXROT(7)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rotate), RXROT(7)); + RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); } -- 1.8.1.4

2 1

[alsa-devel] [PATCH 0/2] Set ab8500-codec dai slots from machine drivers
by Fabio Baltieri 21 May '13

21 May '13

Hi Mark, these two patches rework the slot selection code in ab8500-codec to use the slots masks provided by the machine drivers instead of the hardcoded ones as suggested. The patches replaces some of the previous macros with a parametrized version in the effort of making the actual code compact and readable, and should be applied in order. Also, the second patch drops the hardcoded defintions swapped in -rc2, so you may want to rebase the ux500 topic branch to avoid the conflict, or I can resend the patch based on the topic branch directly. Just let me know. Big thanks to Ola for helping me figuring out the weird ab8500 slot mapping logic. Fabio Fabio Baltieri (2): ASoC: ab8500-codec: Set tx dai slots from tx_mask ASoC: ab8500-codec: Set rx dai slots from rx_mask sound/soc/codecs/ab8500-codec.c | 60 ++++++++++++++++++++++++++--------------- sound/soc/codecs/ab8500-codec.h | 42 +++++++++++++++-------------- 2 files changed, 61 insertions(+), 41 deletions(-) -- 1.8.2

2 5

[alsa-devel] [PATCH 0/2 V4] Power-well API implementation for Haswell
by Wang Xingchao 21 May '13

21 May '13

Hi all, This is V4 and here're some changes notes: change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): i915/drm: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 +++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 98 +++++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 ++++++++++++++ sound/pci/hda/Kconfig | 10 ++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 75 +++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 ++++++++++++++ sound/pci/hda/hda_intel.c | 41 ++++++++++++++-- 10 files changed, 310 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 7

Re: [alsa-devel] [Alsa-devel] add support for syba 7.1 surround sound card
by Clemens Ladisch 21 May '13

21 May '13

good1.2guy(a)gmail.com wrote: > Summary: add support for syba 7.1 surround sound card > Keywords: syba via VT1723 Tremor > Kernel: 3.8.8-203.fc18.x86_64 What are these three lines for? > add support for syba 7.1 surround sound card > actually the card only has 6 dmas, so it is really only 5.1 The website shows a daughterboard. > added code to force_rate setting to not fail EBUSY when the dmas are running. > suggest using force_rate with pulseaudio to avoid loops with error logging. This sounds as if it has nothing to do with this particular SYBA card but affects all VT1723 cards. Please move it into a separate patch. And there should be no reason for a module parameter; if this is a bug in the driver and/or in PulseAudio, it should be just fixed. > Signed-off-by: psmith2004(a)yahoo.com Real names only. Please read Documentation/SubmittingPatches. > +static void syba_quirk_open(struct snd_ice1712 *ice) > +{ > + unsigned long flags; It's common to have an empty line here. > + spin_lock_irqsave(&ice->reg_lock, flags); > + outb(0xff, ICEREG1724(ice, AC97_CFG)); > + udelay(500); > + outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); > + spin_unlock_irqrestore(&ice->reg_lock, flags); > +} What is the reason for this quirk? Is it possible that this might be needed for other cards using a AC'97 codec? Regards, Clemens

1 0

[alsa-devel] what is the difference between SNDRV_PCM_IOCTL_WRITEN_FRAMES and SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Zhang wei 21 May '13

21 May '13

I know the format likes below when use SNDRV_PCM_IOCTL_WRITEI_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> L+R+L+R+L+R... L:a frame left channel data. R:a frame right channel data. Right? But what is the kind of format when use SNDRV_PCM_IOCTL_WRITEN_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> ? Thank you:-) Br zw

2 4

[alsa-devel] [PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A
by Mark Brown 20 May '13

20 May '13

On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Samreen Nilofer <samreen.nilofer(a)intel.com> --- include/linux/mfd/wm8994/registers.h | 8 ++++++++ sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index 0535489..db8cef3 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -2668,6 +2668,10 @@ /* * R772 (0x304) - AIF1ADC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1ADC_LRCLK_DIR 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_MASK 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_SHIFT 11 /* AIF1ADC_LRCLK_DIR */ @@ -2679,6 +2683,10 @@ /* * R773 (0x305) - AIF1DAC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1DAC_LRCLK_DIR 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_MASK 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_SHIFT 11 /* AIF1DAC_LRCLK_DIR */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index cb67603..0bcbd12 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2574,17 +2574,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2603,6 +2610,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2641,12 +2649,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2677,6 +2687,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2] ASoC: soc-compress: Send correct stream event for capture start
by Charles Keepax 20 May '13

20 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7e..06a8000 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- 1.7.2.5

2 1

Re: [alsa-devel] [PATCH] ASoC: max98090: request IRQF_ONESHOT interrupt
by Mark Brown 20 May '13

20 May '13

On Sun, May 19, 2013 at 10:58:07PM -0700, Andrew Bresticker wrote: > request_threaded_irq() rejects calls which both do not specify a handler > (indicating that the primary IRQ handler should be used) and do not set > IRQF_ONESHOT because the combination is unsafe with level-triggered > interrupts. It is safe in this case, though, since max98090 IRQs are Applied, thanks. If request_threaded_irq() is rejecting these that's not ideal - some threaded interrupts can only be generated from within threaded context and therefore can't have primary handlers. Unless it only rejects in the case where the parent IRQ is a hard IRQ of course.

1 0

[alsa-devel] [PATCH 0/2 V3] Power-well API implementation for Haswell
by Wang Xingchao 20 May '13

20 May '13

Hi all, This is V3 and some fixes after Takashi's review. change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): drm/915: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/intel_pm.c | 76 +++++++++++++++++++++++++++++++++++---- include/drm/i915_powerwell.h | 36 +++++++++++++++++++ sound/pci/hda/Kconfig | 10 ++++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 ++++++++++++++++++ sound/pci/hda/hda_intel.c | 41 +++++++++++++++++++-- 7 files changed, 262 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 6

Re: [alsa-devel] [PATCH v2] ASoC: dfbmcs320: make the driver common for other BT modules
by Mark Brown 18 May '13

18 May '13

On Fri, May 17, 2013 at 10:19:37AM +0800, Barry Song wrote: > DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, > we don't want everyone to have a seperate codec driver. anyway, the > feature of Bluetooth handsfree is same on all platforms, so this patch > makes the DFBM-CS320 driver become a common BT handsfree driver. Sorry, should have thought of this before but why are we calling this "handsfree"? It's just a generic bluetooth link as far as the host system is concerned, "bt-sco" might be more appropriate? Otherwise this looks OK for ASoC though I'd expect to see a change to registration of the OpenMoko BT device (perhaps I missed it though).

3 4

[alsa-devel] [patch] ASoC: wm8994: missing break in wm8994_get_fll_config()
by Dan Carpenter 18 May '13

18 May '13

Smatch complains that: sound/soc/codecs/wm8994.c:2087 wm8994_get_fll_config() warn: missing break? reassigning 'fll->k' Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9f32dd8..303d755 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2080,6 +2080,7 @@ static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll, fll->k = K / 10; pr_debug("N=%x K=%x\n", fll->n, fll->k); + break; default: gcd_fll = gcd(freq_out, freq_in);

2 1

[alsa-devel] [PATCH v3] ASoC: dfbmcs320: make the driver common for other BT modules
by Barry Song 18 May '13

18 May '13

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v3: rename to sco according to Mark's comment sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++----------- sound/soc/samsung/neo1973_wm8753.c | 2 +- 4 files changed, 28 insertions(+), 19 deletions(-) rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d8c4f3d..386292d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49ff127..82aa34e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/bt-sco.c similarity index 53% rename from sound/soc/codecs/dfbmcs320.c rename to sound/soc/codecs/bt-sco.c index 4f4f7f4..a081d9f 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/bt-sco.c @@ -1,5 +1,5 @@ /* - * Driver for the DFBM-CS320 bluetooth module + * Driver for generic Bluetooth SCO link * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it @@ -15,8 +15,8 @@ #include <sound/soc.h> -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", .playback = { .channels_min = 1, .channels_max = 1, @@ -31,32 +31,41 @@ static struct snd_soc_dai_driver dfbmcs320_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; -static int dfbmcs320_probe(struct platform_device *pdev) +static int bt_sco_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); } -static int dfbmcs320_remove(struct platform_device *pdev) +static int bt_sco_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver dfmcs320_driver = { +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { .driver = { - .name = "dfbmcs320", + .name = "bt-sco", .owner = THIS_MODULE, }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, }; -module_platform_driver(dfmcs320_driver); +module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- 1.8.2.3

1 0

[alsa-devel] [PATCH 0/6] Second set of fixes for ux500 ASoC drivers
by Fabio Baltieri 18 May '13

18 May '13

Hi, These are some others fixes for various small issues I found while testing the ux500 ASoC driver on -next. Patch 1 adds some missing declarations for AD controls that were causing some weird behaviour in alsamixer, as the default state was outside the declared range. Patch 2 fixes a kernel crash when opening and closing the audio device without sending any data. Patch 3 drops pinctrl code altogether from the driver. The actual implementation is buggy as the pins are only registered to the playback interfaces, which gives a bunch of warnings during kernel startup and also kills the capture interface by setting the shared pins to hi-z mode even if that's still active. As putting those pins in high-z is not really needed and was removed from the internal STE driver anyway, I'm just dropping that code form here as well. In parallel, I'm sending a pinctrl patch to declare those pin as a hog. Patches 4 to 6 fixes some weirdness with time slot usage. After this series the driver seems to work fine for both capture and playback interface (tested on a snowball v11). Thanks, Fabio Fabio Baltieri (6): ASoC: ab8500-codec: Add missing ad_to_slot definitions ASoC: ux500: Do not clear state if already idle ASoC: ux500: Drop pinctrl sleep support ASoC: ux500: Update tx tdm slots configuration ASoC: ux500: Swap even/odd AD slot definitions ASoC: ux500: Use the first two AD slots for capture sound/soc/codecs/ab8500-codec.c | 39 +++++++++++++++------------ sound/soc/codecs/ab8500-codec.h | 36 ++++++++++++------------- sound/soc/ux500/mop500_ab8500.c | 4 +-- sound/soc/ux500/ux500_msp_i2s.c | 58 +++-------------------------------------- sound/soc/ux500/ux500_msp_i2s.h | 6 ----- 5 files changed, 46 insertions(+), 97 deletions(-) -- 1.8.2

4 54

[alsa-devel] [PATCH] ASoC: soc-compress: Send correct stream event for capture start
by Charles Keepax 17 May '13

17 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Sorry little bug I missed due to some integration bugs in our setup here. Again this applies onto Takashi's sound for-next branch. Thanks, Charles sound/soc/soc-compress.c | 9 +++++++-- 1 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7e..a5107df 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,13 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + } else { + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); + } /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A
by Mark Brown 17 May '13

17 May '13

On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ab7ebd9..74238bd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2573,17 +2573,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2602,6 +2609,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2640,12 +2648,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2676,6 +2686,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5110: Add missing speaker initialisation
by Mark Brown 17 May '13

17 May '13

Add callback to initialise the speaker in the core following the recent changes to handling of integration with the thermal interrupts. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index e77410e..91a7015 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1252,6 +1252,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; -- 1.7.10.4

1 0

[alsa-devel] module dependency between i915 and snd-hda-intel
by Wang, Xingchao 17 May '13

17 May '13

Hi Takashi, Seems kernel building will determine the module loading sequence, this would fix the dependency for i915 and snd-hda-intel. If we call the i915 module API directly in snd-hda-intel side, kernel building will change the module.dep accordingly like: " kernel/sound/pci/hda/snd-hda-intel.ko: kernel/drivers/gpu/drm/i915/i915.ko kernel/drivers/gpu/dr m/drm_kms_helper.ko kernel/drivers/gpu/drm/drm.ko kernel/drivers/i2c/algos/i2c-algo-bit.ko kerne l/drivers/acpi/video.ko kernel/sound/pci/hda/snd-hda-codec.ko kernel/sound/core/snd-hwdep.ko ker nel/sound/core/snd-pcm.ko kernel/sound/core/snd-timer.ko kernel/sound/core/snd.ko kernel/sound/s oundcore.ko kernel/sound/core/snd-page-alloc.ko " So snd-hda-intel will wait for i915 loading. Even I added i915 into blacklist.conf, it will force load i915.ko. So I think the only bad case is no i915 module built-in, that would cause haswell hda initialize fail, and the patch would output error message. thanks --xingchao

2 6

[alsa-devel] [PATCH 1/3] ASoC: arizona: Provide simple DAI ops for autoconfiguring interfaces
by Mark Brown 17 May '13

17 May '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ sound/soc/codecs/arizona.h | 1 + 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 90c3db1..4a42041 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1198,6 +1198,13 @@ const struct snd_soc_dai_ops arizona_dai_ops = { }; EXPORT_SYMBOL_GPL(arizona_dai_ops); +const struct snd_soc_dai_ops arizona_simple_dai_ops = { + .startup = arizona_startup, + .hw_params = arizona_hw_params_rate, + .set_sysclk = arizona_dai_set_sysclk, +}; +EXPORT_SYMBOL_GPL(arizona_simple_dai_ops); + int arizona_init_dai(struct arizona_priv *priv, int id) { struct arizona_dai_priv *dai_priv = &priv->dai[id]; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index af39f10..a1a5dad 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -213,6 +213,7 @@ extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); extern const struct snd_soc_dai_ops arizona_dai_ops; +extern const struct snd_soc_dai_ops arizona_simple_dai_ops; #define ARIZONA_FLL_NAME_LEN 20 -- 1.7.10.4

1 2

[alsa-devel] [PATCH] ASoC: simplify registration of snd-soc-dummy device
by Uwe Kleine-König 17 May '13

17 May '13

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/soc-utils.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4b3be6c..29b211e 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -159,15 +159,10 @@ int __init snd_soc_util_init(void) { int ret; - soc_dummy_dev = platform_device_alloc("snd-soc-dummy", -1); - if (!soc_dummy_dev) - return -ENOMEM; - - ret = platform_device_add(soc_dummy_dev); - if (ret != 0) { - platform_device_put(soc_dummy_dev); - return ret; - } + soc_dummy_dev = + platform_device_register_simple("snd-soc-dummy", -1, NULL, 0); + if (IS_ERR(soc_dummy_dev)) + return PTR_ERR(soc_dummy_dev); ret = platform_driver_register(&soc_dummy_driver); if (ret != 0) -- 1.8.2.1

2 1

Re: [alsa-devel] [PATCH v2] ASoC: max98090: add digital mic mux to record path
by Mark Brown 17 May '13

17 May '13

On Thu, May 16, 2013 at 12:03:54PM -0700, Andrew Bresticker wrote: > The max98090 driver currently treats the digital mic enable as a supply > on the record path, causing the digital mic enable to always be turned on > when attempting to record. This is incorrect, however, since the digital Applied, thanks.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - add PCI IDs for Intel BayTrail
by Chew Chiau Ee 17 May '13

17 May '13

From: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> Add HD Audio Device PCI ID for the Intel BayTrail platform. Signed-off-by: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> Signed-off-by: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bcd40ee..fa902a4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3847,6 +3847,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* BayTrail */ + { PCI_DEVICE(0x8086, 0x0f04), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- 1.7.4.4

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format
by Takashi Iwai 17 May '13

17 May '13

At Thu, 16 May 2013 20:26:17 +0200, Torstein Hegge wrote: > > freqshift is only set for the data endpoint and syncmaxsize is only set > for the sync endpoint. This results in a syncmaxsize of zero used in the > proc output feedback format calculation, which gives a feedback format > incorrectly shown as 8.16 for UAC2 devices. > > As neither the data nor the sync endpoint gives all the relevant > content, output the two combined. > > Also remove the sync_endpoint "packet size" which is always zero > and the sync_endpoint "momentary freq" which is constant. > > Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async. > > Reported-by: B. Zhang <bb.zhang(a)free.fr> > Signed-off-by: Torstein Hegge <hegge(a)resisty.net> Thanks, applied. Takashi > --- > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Daniel Mack <zonque(a)gmail.com> > Cc: Clemens Ladisch <clemens(a)ladisch.de> > Cc: Davor Herga <davor.herga(a)guest.arnes.si> > > sound/usb/proc.c | 22 +++++++++++----------- > 1 file changed, 11 insertions(+), 11 deletions(-) > > diff --git a/sound/usb/proc.c b/sound/usb/proc.c > index d218f76..462a5ed 100644 > --- a/sound/usb/proc.c > +++ b/sound/usb/proc.c > @@ -117,21 +117,22 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s > } > > static void proc_dump_ep_status(struct snd_usb_substream *subs, > - struct snd_usb_endpoint *ep, > + struct snd_usb_endpoint *data_ep, > + struct snd_usb_endpoint *sync_ep, > struct snd_info_buffer *buffer) > { > - if (!ep) > + if (!data_ep) > return; > - snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize); > + snd_iprintf(buffer, " Packet Size = %d\n", data_ep->curpacksize); > snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", > subs->speed == USB_SPEED_FULL > - ? get_full_speed_hz(ep->freqm) > - : get_high_speed_hz(ep->freqm), > - ep->freqm >> 16, ep->freqm & 0xffff); > - if (ep->freqshift != INT_MIN) { > - int res = 16 - ep->freqshift; > + ? get_full_speed_hz(data_ep->freqm) > + : get_high_speed_hz(data_ep->freqm), > + data_ep->freqm >> 16, data_ep->freqm & 0xffff); > + if (sync_ep && data_ep->freqshift != INT_MIN) { > + int res = 16 - data_ep->freqshift; > snd_iprintf(buffer, " Feedback Format = %d.%d\n", > - (ep->syncmaxsize > 3 ? 32 : 24) - res, res); > + (sync_ep->syncmaxsize > 3 ? 32 : 24) - res, res); > } > } > > @@ -141,8 +142,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn > snd_iprintf(buffer, " Status: Running\n"); > snd_iprintf(buffer, " Interface = %d\n", subs->interface); > snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); > - proc_dump_ep_status(subs, subs->data_endpoint, buffer); > - proc_dump_ep_status(subs, subs->sync_endpoint, buffer); > + proc_dump_ep_status(subs, subs->data_endpoint, subs->sync_endpoint, buffer); > } else { > snd_iprintf(buffer, " Status: Stop\n"); > } > -- > 1.8.3.rc1.377.g7010c6b >

1 0

Re: [alsa-devel] [PATCH] ASoC: dfbmcs320: make the driver common for other BT modules
by Lars-Peter Clausen 16 May '13

16 May '13

On 05/16/2013 08:53 AM, Barry Song wrote: > DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, > we don't want everyone to have a seperate codec driver. anyway, the > feature of Bluetooth handsfree is same on all platforms, so this patch > makes the DFBM-CS320 driver become a common BT handsfree driver. [...] > -static struct platform_driver dfmcs320_driver = { > +static struct platform_driver bt_handsfree_driver = { > .driver = { > - .name = "dfbmcs320", > + .name = "bt-handsfree", > .owner = THIS_MODULE, In my opinion it's a good idea to keep a platform_id list of all the supported devices. > }, > - .probe = dfbmcs320_probe, > - .remove = dfbmcs320_remove, > + .probe = bt_handsfree_probe, > + .remove = bt_handsfree_remove, > }; > > -module_platform_driver(dfmcs320_driver); > +module_platform_driver(bt_handsfree_driver);

3 4

[alsa-devel] [PATCH] ASoC: wm5102: Correct OSR control name for EPOUT
by Charles Keepax 16 May '13

16 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index cb03cc4..803d1b8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -814,7 +814,7 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), -SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), +SOC_VALUE_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- 1.7.2.5

2 1

[alsa-devel] Debian alsa-plugins patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-plugins that were pending upstreaming Thanks, Jordi

3 8

[alsa-devel] Only stereo output possible with ATI Tahiti XT HDMI Audio [Radeon HD 7970 Series] [1002:aaa0]
by Olivier Langlois 16 May '13

16 May '13

I have an HDMI chain with a Radeon card on 1 side and a 1080P monitor at the other end with a Pioner SC-09TX A/V Receiver in the middle. Video is transmitted fine. 2 channels pcm plays fine on the receiver. I am totally unable to output sound with 8 channels. lano1106@whippet2 /proc/asound/card0 $ speaker-test -c8 speaker-test 1.0.27 Playback device is default Stream parameters are 48000Hz, S16_LE, 8 channels Using 16 octaves of pink noise Channels count (8) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument My neophyte instinct tells me that the problem is interoperability issue between the Radeon Audio controller and the SC-09TX receiver. That would be great to be able to tweak the hda driver to workaround the problem but if I was able to manually program the hda codec module to accept 8 channel LPCM input, that would be good enough for me. My understanding is that in the advent of HDMI handshake failure, the sound source fallbacks to the bare minimum requirement that the HDMI standard requires. I have started to read Intel HDA manual. What I would like to know is: Is it known if my particular HDA device is able to output 48KHz 8 channels LPCM? would it configurable manually? Which HDA verbs should get familiar with if I want to tweak the code to configure the controller manually? Thank you, Olivier lano1106@whippet2 ~ $ lspci -nn | grep Audio 01:00.1 Audio device [0403]: Advanced Micro Devices [AMD] nee ATI Tahiti XT HDMI Audio [Radeon HD 7970 Series] [1002:aaa0] Linux: 3.8.12 alsalib:1.0.27 lano1106@whippet2 /proc/asound/card0 $ dmesg | grep ALSA [ 6.351490] ALSA sound/pci/hda/hda_intel.c:2848 0000:01:00.1: Handle VGA-switcheroo audio client [ 6.351491] ALSA sound/pci/hda/hda_intel.c:3040 0000:01:00.1: Using LPIB position fix [ 6.351493] ALSA sound/pci/hda/hda_intel.c:3157 0000:01:00.1: Force to non-snoop mode [ 6.351619] ALSA sound/pci/hda/hda_intel.c:3282 0000:01:00.1: chipset global capabilities = 0x0 [ 6.354036] ALSA sound/pci/hda/hda_intel.c:1131 0000:01:00.1: codec_mask = 0x1 [ 6.354061] ALSA sound/pci/hda/hda_intel.c:1600 0000:01:00.1: codec #0 probed OK [ 6.354062] ALSA sound/pci/hda/hda_intel.c:1716 0000:01:00.1: Enable sync_write for stable communication [ 6.411369] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=3 Presence_Detect=0 ELD_Valid=0 [ 6.411397] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 6.411424] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=7 Presence_Detect=0 ELD_Valid=0 [ 6.411453] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=9 Presence_Detect=0 ELD_Valid=0 [ 6.411482] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=11 Presence_Detect=0 ELD_Valid=0 [ 6.411512] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=13 Presence_Detect=0 ELD_Valid=0 [ 29.487577] ALSA sound/pci/hda/patch_hdmi.c:951 HDMI hot plug event: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=0 [ 29.487593] ALSA sound/pci/hda/patch_hdmi.c:1173 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 29.487605] ALSA sound/pci/hda/hda_eld.c:337 HDMI: ELD buf size is 0, force 128 [ 29.487617] ALSA sound/pci/hda/hda_eld.c:356 HDMI: invalid ELD data byte 0 lano1106@whippet2 /proc/asound/card0 $ cat eld#0.0 monitor_present 1 eld_valid 0 lano1106@whippet2 /proc/asound/card0 :( $ cat codec#0 Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100300 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=1, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=3 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="ELD", index=0, device=3 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x02 Node 0x04 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=7 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Control: name="ELD", index=0, device=7 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=8 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Control: name="ELD", index=0, device=8 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=9 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Control: name="ELD", index=0, device=9 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=10 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=4, device=0 Control: name="IEC958 Playback Pro Mask", index=4, device=0 Control: name="IEC958 Playback Default", index=4, device=0 Control: name="IEC958 Playback Switch", index=4, device=0 Control: name="ELD", index=0, device=10 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="HDMI/DP,pcm=11 Jack", index=0, device=0 Control: name="IEC958 Playback Con Mask", index=5, device=0 Control: name="IEC958 Playback Pro Mask", index=5, device=0 Control: name="IEC958 Playback Default", index=5, device=0 Control: name="IEC958 Playback Switch", index=5, device=0 Control: name="ELD", index=0, device=11 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=06, enabled=1 Connection: 1 0x0c whippet2 /sys/kernel/debug/tracing # cat trace_pipe kworker/u:5-130 [002] ...1 79390.094652: hda_send_cmd: [0:0] val=3f0900 kworker/u:5-130 [002] ...1 79390.094665: hda_get_response: [0:0] val=ffffffff kworker/u:5-130 [002] ...1 79390.094667: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.094677: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.094679: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.094689: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.394001: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.394045: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.394050: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.394071: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.694227: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [002] ...1 79390.694281: hda_get_response: [0:0] val=0 kworker/u:5-130 [002] ...1 79390.694286: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [002] ...1 79390.694307: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.617406: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28163.617420: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.617425: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28163.617436: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.917628: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28163.917642: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28163.917647: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28163.917657: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.217846: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.217860: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.217865: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.217875: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.518101: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.518115: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.518119: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.518130: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.818310: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28164.818323: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28164.818328: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28164.818339: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28165.118549: hda_send_cmd: [0:0] val=3f2e08 kworker/u:5-130 [000] ...1 28165.118563: hda_get_response: [0:0] val=0 kworker/u:5-130 [000] ...1 28165.118568: hda_send_cmd: [0:0] val=3f2f00 kworker/u:5-130 [000] ...1 28165.118579: hda_get_response: [0:0] val=0

1 2

[alsa-devel] 7.1 surround channels missing on CS46xx
by Ondrej Zary 16 May '13

16 May '13

Hello, I got a Hercules Fortissimo 7.1 sound card (CS4624 + two CS4294) but snd-cs46xx driver supports only 6 channels (front, rear, center/lfe). The surround output (7&8) is not supported - as seen in dsp_spos_scb_lib.c: case DSP_PCM_S71_CHANNEL: /* TODO */ snd_BUG(); break; Tried to add a code for this, mostly copying the center/lfe code but I miss some data: - which slots (ACOSV_SLVx) should be used (maybe 10 and 12?) - what are the channel_disp and fifo_addr values for cs46xx_dsp_create_codec_out_scb() function - what values should S71_MIXER_SCB_ADDR and S71_CODEC_SCB_ADDR be (maybe 0x1b0 and 0x1c0?) -- Ondrej Zary

1 0

[alsa-devel] Debian alsa-tools patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-tools that were pending upstreaming. Please review patch 8, as this one is new and the general state of configure scripts in alsa-tools is less than optimal. I've left out the patch to correct envy24control's configuration directory, as the additional check for the old dir is not implemented. If anyone wants to pick that up, please contact me or fetch the (trivial) patch from Debian sources. Thanks, Jordi

3 15

[alsa-devel] Debian alsa-utils patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-utils that were pending upstreaming. Thanks, Jordi

2 5

[alsa-devel] Debian alsa-lib patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-lib that were pending upstreaming. Thanks, Jordi

2 4

[alsa-devel] Debian alsa-lib patches
by Jordi Mallach 16 May '13

16 May '13

As previously discussed, I'm posting all Debian patches against alsa-oss that were pending upstreaming. Thanks, Jordi [PATCH 1/3] Fix path to libaoss.so. [PATCH 2/3] Show a warning if the alsa wrapper failed. [PATCH 3/3] Add AM_MAINTAINER_MODE([enable]) to configure.in.

2 6

[alsa-devel] [PATCH - pcm_plug] Add details about what pcm_plug does in doc
by olivier＠trillion01.com 16 May '13

16 May '13

From: Olivier Langlois <olivier(a)trillion01.com> Current doc only mention conversion to channels, rate and format. It alsa takes care of access mode mismatch. This is what I was looking for and I had to look the source code to figure out. Signed-off-by: Olivier Langlois <olivier(a)trillion01.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index fa84eaa..8a639af 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -1155,7 +1155,8 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, \section pcm_plugins_plug Automatic conversion plugin -This plugin converts channels, rate and format on request. +This plugin converts channels, rate and format on request. It will also +implicitly adapt to the slave plugin access mode if different than our own. \code pcm.name { -- 1.8.2.3

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - add PCI IDs for Intel BayTrail
by Takashi Iwai 16 May '13

16 May '13

At Thu, 16 May 2013 13:27:18 +0800, Chew Chiau Ee wrote: > > From: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> > > Add HD Audio Device PCI ID for the Intel BayTrail platform. It's rather the successor of Oaktrail & co, right? Then put after Oaktrail entry, please. thanks, Takashi > Signed-off-by: Chew, Chiau Ee <chiau.ee.chew(a)intel.com> > Signed-off-by: Artem Bityutskiy <artem.bityutskiy(a)linux.intel.com> > --- > sound/pci/hda/hda_intel.c | 3 +++ > 1 files changed, 3 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index bcd40ee..18640d9 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -3811,6 +3811,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > /* CPT */ > { PCI_DEVICE(0x8086, 0x1c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > + /* BayTrail */ > + { PCI_DEVICE(0x8086, 0x0f04), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > -- > 1.7.4.4 >

1 0

[alsa-devel] [PATCH 0/2 V2] Power-well API implementation for Haswell
by Wang Xingchao 16 May '13

16 May '13

Hi all, This is V2 and some fixes after first round review: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): drm/915: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/intel_pm.c | 76 +++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 +++++++++++++++ sound/pci/hda/Kconfig | 8 ++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 92 +++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 33 ++++++++++++-- 7 files changed, 273 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 7

[alsa-devel] [PATCH] ALSA: hda - Add headset mic support for another Dell machine
by David Henningsson 15 May '13

15 May '13

BugLink: https://bugs.launchpad.net/bugs/1180351 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 6bf47f7..59d2e91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3482,6 +3482,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1