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[alsa-devel] [PATCH v2 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 13 May '16

13 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 68 +++++++++++++++++++++++++++++++++++++--------- sound/soc/soc-pcm.c | 3 +- 3 files changed, 61 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..fe5bc93 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..075f0d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,13 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); + if (custom_stop_condition) { + con = custom_stop_condition(widget, dir); + if (con >= 0) + return con; + con = 0; + } + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1117,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1130,41 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1172,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1211,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1310,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1970,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

2 5

[alsa-devel] [PATCH v3 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 13 May '16

13 May '16

Updated driver for TI's TAS5720L/M digital audio amplifiers incorporating previous feedback (thanks Mark Brown). Biggest change is that the driver is no longer using interrupts for codec fault reporting, but instead uses a polling-based approach to check for (and clear) codec fault conditions every 200ms. This is much more graceful and actually lower overhead than trying to keep the actual TAS5720 fault interrupts (that come in every 300us!) under control which is what v2 of the driver was doing. Because of the removal of interrupts the DT doc had to be updated as well, so I did not carry forward Rob Herring's earlier Acked-by because of said change. -- Andreas Dannenberg Texas Instruments Inc Changes since v2: - Switched fault handling from using interrupts to polling mode - Remove interrupt related description from DT bindings doc - Remove unlikely() to simplify an expression - Remove unnesseary tas5720_set_dai_sysclk() function stub Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 26 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 638 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 766 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

3 14

[alsa-devel] [PATCH] ASoC: rt5668: add rt5668 codec driver
by John Lin 13 May '16

13 May '16

This is the initial codec driver for rt5668. Signed-off-by: John Lin <john.lin(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5668.txt | 28 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5668.c | 3280 ++++++++++++++++++++ sound/soc/codecs/rt5668.h | 781 +++++ 5 files changed, 4097 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5668.txt create mode 100644 sound/soc/codecs/rt5668.c create mode 100644 sound/soc/codecs/rt5668.h diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt new file mode 100644 index 0000000..50d7759 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -0,0 +1,28 @@ +RT5668/RT5663 audio CODECs + +These devices support I2C only. + +Required properties: + +- compatible : "realtek,rt5668" or "realtek,rt5663" + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Pins on the device (for linking into audio routes) for RT5668/RT5663: + + * IN1P + * IN1N + * IN2P + * IN2N + * HPOL + * HPOR + +Example: + +codec: rt5668@1a { + compatible = "realtek,rt5668"; + reg = <0x1a>; + interrupts = <347>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ac7317e..283c4bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -107,6 +107,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C + select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -610,6 +611,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y + default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -618,6 +620,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m + default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -660,6 +663,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5668 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 94873ee..88b16d9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -105,6 +105,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o +snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -318,6 +319,7 @@ obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5659) += snd-soc-rt5659.o +obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c new file mode 100644 index 0000000..271fd01 --- /dev/null +++ b/sound/soc/codecs/rt5668.c @@ -0,0 +1,3280 @@ +/* + * rt5668.c -- RT5668/RT5663 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: John Lin <john.lin(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5668.h" +#include "rl6231.h" + +#define RT5668_DEVICE_ID 0x6451 +#define RT5663_DEVICE_ID 0x6406 + +enum { + CODEC_TYPE_RT5668, + CODEC_TYPE_RT5663, +}; + +struct rt5668_priv { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct delayed_work jack_detect_work; + struct snd_soc_jack *hs_jack; + struct timer_list btn_check_timer; + + int codec_type; + int sysclk; + int sysclk_src; + int lrck; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_default rt5668_reg[] = { + { 0x0000, 0x0000 }, + { 0x0001, 0xc8c8 }, + { 0x0002, 0x8080 }, + { 0x0003, 0x8000 }, + { 0x0004, 0xc80a }, + { 0x0005, 0x0000 }, + { 0x0006, 0x0000 }, + { 0x0007, 0x0000 }, + { 0x000a, 0x0000 }, + { 0x000b, 0x0000 }, + { 0x000c, 0x0000 }, + { 0x000d, 0x0000 }, + { 0x000f, 0x0808 }, + { 0x0010, 0x4000 }, + { 0x0011, 0x0000 }, + { 0x0012, 0x1404 }, + { 0x0013, 0x1000 }, + { 0x0014, 0xa00a }, + { 0x0015, 0x0404 }, + { 0x0016, 0x0404 }, + { 0x0017, 0x0011 }, + { 0x0018, 0xafaf }, + { 0x0019, 0xafaf }, + { 0x001a, 0xafaf }, + { 0x001b, 0x0011 }, + { 0x001c, 0x2f2f }, + { 0x001d, 0x2f2f }, + { 0x001e, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0020, 0x0000 }, + { 0x0021, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0024, 0x000b }, + { 0x0026, 0xc0c0 }, + { 0x0027, 0xc0c0 }, + { 0x0028, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xaaaa }, + { 0x002b, 0xaaaa }, + { 0x002c, 0xaba8 }, + { 0x002d, 0x0000 }, + { 0x002e, 0x0000 }, + { 0x002f, 0x0000 }, + { 0x0030, 0x0000 }, + { 0x0031, 0x5000 }, + { 0x0032, 0x0000 }, + { 0x0033, 0x0000 }, + { 0x0034, 0x0000 }, + { 0x0035, 0x0000 }, + { 0x003a, 0x0000 }, + { 0x003b, 0x0000 }, + { 0x003c, 0x00ff }, + { 0x003d, 0x0000 }, + { 0x003e, 0x00ff }, + { 0x003f, 0x0000 }, + { 0x0040, 0x0000 }, + { 0x0041, 0x00ff }, + { 0x0042, 0x0000 }, + { 0x0043, 0x00ff }, + { 0x0044, 0x0c0c }, + { 0x0049, 0xc00b }, + { 0x004a, 0x0000 }, + { 0x004b, 0x031f }, + { 0x004d, 0x0000 }, + { 0x004e, 0x001f }, + { 0x004f, 0x0000 }, + { 0x0050, 0x001f }, + { 0x0052, 0xf000 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x003f }, + { 0x0067, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006d, 0xff00 }, + { 0x006e, 0x2808 }, + { 0x006f, 0x000a }, + { 0x0070, 0x8000 }, + { 0x0071, 0x8000 }, + { 0x0072, 0x8000 }, + { 0x0073, 0x7000 }, + { 0x0074, 0x7770 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0078, 0x00f0 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0000 }, + { 0x007b, 0x0000 }, + { 0x007c, 0x0000 }, + { 0x007d, 0x0123 }, + { 0x007e, 0x4500 }, + { 0x007f, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0085, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x0088, 0x0000 }, + { 0x0089, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0060 }, + { 0x008f, 0x1000 }, + { 0x0091, 0x0c26 }, + { 0x0092, 0x0073 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0098, 0x0000 }, + { 0x0099, 0x0000 }, + { 0x009a, 0x0007 }, + { 0x009f, 0x0000 }, + { 0x00a0, 0x0000 }, + { 0x00a1, 0x0002 }, + { 0x00a2, 0x0001 }, + { 0x00a3, 0x0002 }, + { 0x00a4, 0x0001 }, + { 0x00ae, 0x2040 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00b9, 0x0000 }, + { 0x00ba, 0x0002 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0aaa }, + { 0x00c2, 0xaa80 }, + { 0x00c3, 0x0003 }, + { 0x00c4, 0x0000 }, + { 0x00d0, 0x0000 }, + { 0x00d1, 0x2244 }, + { 0x00d2, 0x0000 }, + { 0x00d3, 0x3300 }, + { 0x00d4, 0x2200 }, + { 0x00d9, 0x0809 }, + { 0x00da, 0x0000 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x600c }, + { 0x00ea, 0xb320 }, + { 0x00eb, 0x0000 }, + { 0x00ec, 0xb300 }, + { 0x00ed, 0x0000 }, + { 0x00ee, 0xb320 }, + { 0x00ef, 0x0000 }, + { 0x00f0, 0x0201 }, + { 0x00f1, 0x0ddd }, + { 0x00f2, 0x0ddd }, + { 0x00f6, 0x0000 }, + { 0x00f7, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x0000 }, + { 0x00fb, 0x0000 }, + { 0x00fc, 0x0000 }, + { 0x00fd, 0x0000 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6451 }, + { 0x0100, 0xaaaa }, + { 0x0101, 0x000a }, + { 0x010a, 0xaaaa }, + { 0x010b, 0xa0a0 }, + { 0x010c, 0xaeae }, + { 0x010d, 0xaaaa }, + { 0x010e, 0xaaaa }, + { 0x010f, 0xaaaa }, + { 0x0110, 0xe002 }, + { 0x0111, 0xa602 }, + { 0x0112, 0xaaaa }, + { 0x0113, 0x2000 }, + { 0x0117, 0x0f00 }, + { 0x0125, 0x0420 }, + { 0x0132, 0x0000 }, + { 0x0133, 0x0000 }, + { 0x0136, 0x5555 }, + { 0x0137, 0x5540 }, + { 0x0138, 0x3700 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x2020 }, + { 0x013b, 0x2020 }, + { 0x013c, 0x2005 }, + { 0x013f, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0147, 0x0000 }, + { 0x0148, 0x0000 }, + { 0x0160, 0x4ec0 }, + { 0x0161, 0x0080 }, + { 0x0162, 0x0200 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x000f }, + { 0x0168, 0x000f }, + { 0x0170, 0x4e80 }, + { 0x0171, 0x0080 }, + { 0x0172, 0x0200 }, + { 0x0173, 0x0800 }, + { 0x0174, 0x00ff }, + { 0x0175, 0x0000 }, + { 0x0190, 0x4131 }, + { 0x0191, 0x4131 }, + { 0x0192, 0x4131 }, + { 0x0193, 0x4131 }, + { 0x0194, 0x0000 }, + { 0x0195, 0x0000 }, + { 0x0196, 0x0000 }, + { 0x0197, 0x0000 }, + { 0x0198, 0x0000 }, + { 0x0199, 0x0000 }, + { 0x01a0, 0x1e64 }, + { 0x01a1, 0x06a3 }, + { 0x01a2, 0x0000 }, + { 0x01a3, 0x0000 }, + { 0x01a4, 0x0000 }, + { 0x01a5, 0x0000 }, + { 0x01a6, 0x0000 }, + { 0x01a7, 0x0000 }, + { 0x01a8, 0x0000 }, + { 0x01a9, 0x0000 }, + { 0x01aa, 0x0000 }, + { 0x01ab, 0x0000 }, + { 0x01b5, 0x0000 }, + { 0x01b6, 0x01c3 }, + { 0x01b7, 0x02a0 }, + { 0x01b8, 0x03e9 }, + { 0x01b9, 0x1389 }, + { 0x01ba, 0xc351 }, + { 0x01bb, 0x0009 }, + { 0x01bc, 0x0018 }, + { 0x01bd, 0x002a }, + { 0x01be, 0x004c }, + { 0x01bf, 0x0097 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0000 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0000 }, + { 0x01c6, 0x0000 }, + { 0x01c7, 0x0000 }, + { 0x01c8, 0x40af }, + { 0x01c9, 0x0702 }, + { 0x01ca, 0x0000 }, + { 0x01cb, 0x0000 }, + { 0x01cc, 0x5757 }, + { 0x01cd, 0x5757 }, + { 0x01ce, 0x5757 }, + { 0x01cf, 0x5757 }, + { 0x01d0, 0x5757 }, + { 0x01d1, 0x5757 }, + { 0x01d2, 0x5757 }, + { 0x01d3, 0x5757 }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x003c }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dc, 0x0000 }, + { 0x01de, 0x7c00 }, + { 0x01df, 0x0320 }, + { 0x01e0, 0x06a1 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0000 }, + { 0x01e6, 0x0001 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x01f3, 0x0000 }, + { 0x01f4, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x0000 }, + { 0x0202, 0x0000 }, + { 0x0203, 0x0000 }, + { 0x0204, 0x0000 }, + { 0x0205, 0x0000 }, + { 0x0206, 0x0000 }, + { 0x0207, 0x0000 }, + { 0x0208, 0x0000 }, + { 0x0210, 0x60b1 }, + { 0x0211, 0xa000 }, + { 0x0212, 0x024c }, + { 0x0213, 0xf7ff }, + { 0x0214, 0x024c }, + { 0x0215, 0x0102 }, + { 0x0216, 0x00a3 }, + { 0x0217, 0x0048 }, + { 0x0218, 0x92c0 }, + { 0x0219, 0x0000 }, + { 0x021a, 0x00c8 }, + { 0x021b, 0x0020 }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x02ff, 0x0110 }, + { 0x0300, 0x001f }, + { 0x0301, 0x032c }, + { 0x0302, 0x5f21 }, + { 0x0303, 0x4000 }, + { 0x0304, 0x4000 }, + { 0x0305, 0x06d5 }, + { 0x0306, 0x8000 }, + { 0x0307, 0x0700 }, + { 0x0310, 0x4560 }, + { 0x0311, 0xa4a8 }, + { 0x0312, 0x7418 }, + { 0x0313, 0x0000 }, + { 0x0314, 0x0006 }, + { 0x0315, 0xffff }, + { 0x0316, 0xc400 }, + { 0x0317, 0x0000 }, + { 0x0330, 0x00a6 }, + { 0x0331, 0x04c3 }, + { 0x0332, 0x27c8 }, + { 0x0333, 0xbf50 }, + { 0x0334, 0x0045 }, + { 0x0335, 0x0007 }, + { 0x0336, 0x7418 }, + { 0x0337, 0x0501 }, + { 0x0338, 0x0000 }, + { 0x0339, 0x0010 }, + { 0x033a, 0x1010 }, + { 0x03c0, 0x7e00 }, + { 0x03c1, 0x8000 }, + { 0x03c2, 0x8000 }, + { 0x03c3, 0x8000 }, + { 0x03c4, 0x8000 }, + { 0x03c5, 0x8000 }, + { 0x03c6, 0x8000 }, + { 0x03c7, 0x8000 }, + { 0x03c8, 0x8000 }, + { 0x03c9, 0x8000 }, + { 0x03ca, 0x8000 }, + { 0x03cb, 0x8000 }, + { 0x03cc, 0x8000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, + { 0x03fe, 0x0000 }, + { 0x03ff, 0x0000 }, + { 0x07f0, 0x0000 }, + { 0x07fa, 0x0000 }, +}; + +static const struct reg_default rt5663_reg[] = { + { 0x0000, 0x0000 }, + { 0x0002, 0x0008 }, + { 0x0005, 0x1000 }, + { 0x0006, 0x1000 }, + { 0x000a, 0x0000 }, + { 0x0010, 0x000f }, + { 0x0015, 0x42c1 }, + { 0x0016, 0x0000 }, + { 0x0018, 0x000b }, + { 0x0019, 0xafaf }, + { 0x001c, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0026, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xa0a0 }, + { 0x002c, 0x000c }, + { 0x002d, 0x0000 }, + { 0x0040, 0x0808 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006e, 0x0000 }, + { 0x006f, 0x0000 }, + { 0x0070, 0x8020 }, + { 0x0073, 0x1000 }, + { 0x0074, 0xe400 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0077, 0x00f0 }, + { 0x0078, 0x0000 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0123 }, + { 0x007b, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0004 }, + { 0x008f, 0x1000 }, + { 0x0090, 0x0646 }, + { 0x0091, 0x0e3e }, + { 0x0092, 0x1071 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0097, 0x0000 }, + { 0x0098, 0x0000 }, + { 0x009a, 0x0000 }, + { 0x009f, 0x0000 }, + { 0x00ae, 0x2000 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00ba, 0x0000 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00bf, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0000 }, + { 0x00c5, 0x0000 }, + { 0x00cb, 0xa02f }, + { 0x00cc, 0x0000 }, + { 0x00cd, 0x0e02 }, + { 0x00d9, 0x08f9 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x0043 }, + { 0x00e4, 0x400b }, + { 0x00e5, 0x8031 }, + { 0x00e6, 0x3080 }, + { 0x00e7, 0x4100 }, + { 0x00e8, 0x1400 }, + { 0x00e9, 0xe00a }, + { 0x00ea, 0x0404 }, + { 0x00eb, 0x0404 }, + { 0x00ec, 0xb320 }, + { 0x00ed, 0x0000 }, + { 0x00f4, 0x0000 }, + { 0x00f6, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x8000 }, + { 0x00fd, 0x0001 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6406 }, + { 0x0100, 0xa0a0 }, + { 0x0108, 0x4444 }, + { 0x0109, 0x4444 }, + { 0x010a, 0xaaaa }, + { 0x010b, 0x00a0 }, + { 0x010c, 0x8aaa }, + { 0x010d, 0xaaaa }, + { 0x010e, 0x2aaa }, + { 0x010f, 0x002a }, + { 0x0110, 0xa0a4 }, + { 0x0111, 0x4602 }, + { 0x0112, 0x0101 }, + { 0x0113, 0x2000 }, + { 0x0114, 0x0000 }, + { 0x0116, 0x0000 }, + { 0x0117, 0x0f00 }, + { 0x0118, 0x0006 }, + { 0x0125, 0x2224 }, + { 0x0126, 0x5550 }, + { 0x0127, 0x0400 }, + { 0x0128, 0x7711 }, + { 0x0132, 0x0004 }, + { 0x0137, 0x5441 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x30c0 }, + { 0x013b, 0x2000 }, + { 0x013c, 0x2005 }, + { 0x013d, 0x30c0 }, + { 0x013e, 0x0000 }, + { 0x0140, 0x3700 }, + { 0x0141, 0x1f00 }, + { 0x0144, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0160, 0x0e80 }, + { 0x0161, 0x0020 }, + { 0x0162, 0x0080 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x1417 }, + { 0x0168, 0x0017 }, + { 0x0169, 0x0017 }, + { 0x0180, 0x2000 }, + { 0x0181, 0x0000 }, + { 0x0182, 0x0000 }, + { 0x0183, 0x2000 }, + { 0x0184, 0x0000 }, + { 0x0185, 0x0000 }, + { 0x01b0, 0x4b30 }, + { 0x01b1, 0x0000 }, + { 0x01b2, 0xd870 }, + { 0x01b3, 0x0000 }, + { 0x01b4, 0x0030 }, + { 0x01b5, 0x5757 }, + { 0x01b6, 0x5757 }, + { 0x01b7, 0x5757 }, + { 0x01b8, 0x5757 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0540 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0009 }, + { 0x01c6, 0x0018 }, + { 0x01c7, 0x002a }, + { 0x01c8, 0x004c }, + { 0x01c9, 0x0097 }, + { 0x01ca, 0x01c3 }, + { 0x01cb, 0x03e9 }, + { 0x01cc, 0x1389 }, + { 0x01cd, 0xc351 }, + { 0x01ce, 0x0000 }, + { 0x01cf, 0x0000 }, + { 0x01d0, 0x0000 }, + { 0x01d1, 0x0000 }, + { 0x01d2, 0x0000 }, + { 0x01d3, 0x003c }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x5757 }, + { 0x01d7, 0x5757 }, + { 0x01d8, 0x5757 }, + { 0x01d9, 0x5757 }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dd, 0x0009 }, + { 0x01de, 0x7f00 }, + { 0x01df, 0x00c8 }, + { 0x01e0, 0x0691 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0040 }, + { 0x01e6, 0x0000 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x2244 }, + { 0x0202, 0xaaaa }, + { 0x0250, 0x8010 }, + { 0x0251, 0x0000 }, + { 0x0252, 0x028a }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x0300, 0x0000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, +}; + +static bool rt5668_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0011: + case 0x0082: + case 0x00ae: + case 0x00be: + case 0x00db: + case 0x00df: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x01de: + case 0x01e7: + return true; + default: + return false; + } +} + +static bool rt5668_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0001: + case 0x0002: + case 0x0003: + case 0x0004: + case 0x0005: + case 0x0006: + case 0x0007: + case 0x000a: + case 0x000b: + case 0x000c: + case 0x000d: + case 0x000f: + case 0x0010: + case 0x0011: + case 0x0012: + case 0x0013: + case 0x0014: + case 0x0015: + case 0x0016: + case 0x0017: + case 0x0018: + case 0x0019: + case 0x001a: + case 0x001b: + case 0x001c: + case 0x001d: + case 0x001e: + case 0x001f: + case 0x0020: + case 0x0021: + case 0x0022: + case 0x0023: + case 0x0024: + case 0x0026: + case 0x0027: + case 0x0028: + case 0x0029: + case 0x002a: + case 0x002b: + case 0x002c: + case 0x002d: + case 0x002e: + case 0x002f: + case 0x0030: + case 0x0031: + case 0x0032: + case 0x0033: + case 0x0034: + case 0x0035: + case 0x003a: + case 0x003b: + case 0x003c: + case 0x003d: + case 0x003e: + case 0x003f: + case 0x0040: + case 0x0041: + case 0x0042: + case 0x0043: + case 0x0044: + case 0x0049: + case 0x004a: + case 0x004b: + case 0x004d: + case 0x004e: + case 0x004f: + case 0x0050: + case 0x0052: + case 0x0061: + case 0x0062: + case 0x0063: + case 0x0064: + case 0x0065: + case 0x0066: + case 0x0067: + case 0x006b: + case 0x006d: + case 0x006e: + case 0x006f: + case 0x0070: + case 0x0071: + case 0x0072: + case 0x0073: + case 0x0074: + case 0x0075: + case 0x0076: + case 0x0078: + case 0x0079: + case 0x007a: + case 0x007b: + case 0x007c: + case 0x007d: + case 0x007e: + case 0x007f: + case 0x0080: + case 0x0081: + case 0x0082: + case 0x0083: + case 0x0084: + case 0x0085: + case 0x0086: + case 0x0087: + case 0x0088: + case 0x0089: + case 0x008a: + case 0x008b: + case 0x008c: + case 0x008e: + case 0x008f: + case 0x0091: + case 0x0092: + case 0x0093: + case 0x0094: + case 0x0098: + case 0x0099: + case 0x009a: + case 0x009f: + case 0x00a0: + case 0x00a1: + case 0x00a2: + case 0x00a3: + case 0x00a4: + case 0x00ae: + case 0x00af: + case 0x00b6: + case 0x00b7: + case 0x00b8: + case 0x00b9: + case 0x00ba: + case 0x00bb: + case 0x00be: + case 0x00c0: + case 0x00c1: + case 0x00c2: + case 0x00c3: + case 0x00c4: + case 0x00d0: + case 0x00d1: + case 0x00d2: + case 0x00d3: + case 0x00d4: + case 0x00d9: + case 0x00da: + case 0x00db: + case 0x00dc: + case 0x00dd: + case 0x00de: + case 0x00df: + case 0x00e0: + case 0x00e1: + case 0x00e2: + case 0x00ea: + case 0x00eb: + case 0x00ec: + case 0x00ed: + case 0x00ee: + case 0x00ef: + case 0x00f0: + case 0x00f1: + case 0x00f2: + case 0x00f6: + case 0x00f7: + case 0x00f8: + case 0x00fa: + case 0x00fb: + case 0x00fc: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x0100: + case 0x0101: + case 0x010a: + case 0x010b: + case 0x010c: + case 0x010d: + case 0x010e: + case 0x010f: + case 0x0110: + case 0x0111: + case 0x0112: + case 0x0113: + case 0x0117: + case 0x0125: + case 0x0132: + case 0x0133: + case 0x0136: + case 0x0137: + case 0x0138: + case 0x0139: + case 0x013a: + case 0x013b: + case 0x013c: + case 0x013f: + case 0x0145: + case 0x0146: + case 0x0147: + case 0x0148: + case 0x0160: + case 0x0161: + case 0x0162: + case 0x0163: + case 0x0164: + case 0x0165: + case 0x0166: + case 0x0167: + case 0x0168: + case 0x0170: + case 0x0171: + case 0x0172: + case 0x0173: + case 0x0174: + case 0x0175: + case 0x0190: + case 0x0191: + case 0x0192: + case 0x0193: + case 0x0194: + case 0x0195: + case 0x0196: + case 0x0197: + case 0x0198: + case 0x0199: + case 0x01a0: + case 0x01a1: + case 0x01a2: + case 0x01a3: + case 0x01a4: + case 0x01a5: + case 0x01a6: + case 0x01a7: + case 0x01a8: + case 0x01a9: + case 0x01aa: + case 0x01ab: + case 0x01b5: + case 0x01b6: + case 0x01b7: + case 0x01b8: + case 0x01b9: + case 0x01ba: + case 0x01bb: + case 0x01bc: + case 0x01bd: + case 0x01be: + case 0x01bf: + case 0x01c0: + case 0x01c1: + case 0x01c2: + case 0x01c3: + case 0x01c4: + case 0x01c5: + case 0x01c6: + case 0x01c7: + case 0x01c8: + case 0x01c9: + case 0x01ca: + case 0x01cb: + case 0x01cc: + case 0x01cd: + case 0x01ce: + case 0x01cf: + case 0x01d0: + case 0x01d1: + case 0x01d2: + case 0x01d3: + case 0x01d4: + case 0x01d5: + case 0x01d6: + case 0x01da: + case 0x01db: + case 0x01dc: + case 0x01de: + case 0x01df: + case 0x01e0: + case 0x01e1: + case 0x01e2: + case 0x01e3: + case 0x01e4: + case 0x01e5: + case 0x01e6: + case 0x01e7: + case 0x01e8: + case 0x01ea: + case 0x01eb: + case 0x01ec: + case 0x01ed: + case 0x01ee: + case 0x01ef: + case 0x01f0: + case 0x01f1: + case 0x01f2: + case 0x01f3: + case 0x01f4: + case 0x0200: + case 0x0201: + case 0x0202: + case 0x0203: + case 0x0204: + case 0x0205: + case 0x0206: + case 0x0207: + case 0x0208: + case 0x0210: + case 0x0211: + case 0x0212: + case 0x0213: + case 0x0214: + case 0x0215: + case 0x0216: + case 0x0217: + case 0x0218: + case 0x0219: + case 0x021a: + case 0x021b: + case 0x02fa: + case 0x02fb: + case 0x02fc: + case 0x02ff: + case 0x0300: + case 0x0301: + case 0x0302: + case 0x0303: + case 0x0304: + case 0x0305: + case 0x0306: + case 0x0307: + case 0x0310: + case 0x0311: + case 0x0312: + case 0x0313: + case 0x0314: + case 0x0315: + case 0x0316: + case 0x0317: + case 0x0330: + case 0x0331: + case 0x0332: + case 0x0333: + case 0x0334: + case 0x0335: + case 0x0336: + case 0x0337: + case 0x0338: + case 0x0339: + case 0x033a: + case 0x03c0: + case 0x03c1: + case 0x03c2: + case 0x03c3: + case 0x03c4: + case 0x03c5: + case 0x03c6: + case 0x03c7: + case 0x03c8: + case 0x03c9: + case 0x03ca: + case 0x03cb: + case 0x03cc: + case 0x03d0: + case 0x03d1: + case 0x03d2: + case 0x03d3: + case 0x03d4: + case 0x03d5: + case 0x03d6: + case 0x03d7: + case 0x03d8: + case 0x03d9: + case 0x03da: + case 0x03db: + case 0x03dc: + case 0x03dd: + case 0x03de: + case 0x03df: + case 0x03e0: + case 0x03e1: + case 0x03e2: + case 0x03e3: + case 0x03e4: + case 0x03e5: + case 0x03e6: + case 0x03e7: + case 0x03e8: + case 0x03e9: + case 0x03ea: + case 0x03eb: + case 0x03ec: + case 0x03ed: + case 0x03ee: + case 0x03ef: + case 0x03f0: + case 0x03f1: + case 0x03f2: + case 0x03f3: + case 0x03fe: + case 0x03ff: + case 0x07f0: + case 0x07fa: + return true; + default: + return false; + } +} + +static bool rt5663_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0015: + case 0x0023: + case 0x0029: + case 0x0076: + case 0x0093: + case 0x0097: + case 0x00ae: + case 0x00be: + case 0x00bf: + case 0x00c5: + case 0x00cb: + case 0x00db: + case 0x00df: + case 0x00e4: + case 0x00e6: + case 0x00e7: + case 0x00e8: + case 0x00f8: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x0139: + case 0x0141: + case 0x0160: + case 0x0164: + case 0x0165: + case 0x0166: + case 0x0180: + case 0x0183: + case 0x01c0: + case 0x01c2: + case 0x01c3: + case 0x01c4: + case 0x01de: + case 0x01e6: + case 0x01ea: + case 0x01eb: + case 0x01ec: + case 0x01ed: + case 0x01ee: + case 0x01ef: + case 0x01f0: + case 0x01f1: + case 0x01f2: + case 0x0300: + return true; + default: + return false; + } +} + +static bool rt5663_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0002: + case 0x0005: + case 0x0006: + case 0x000a: + case 0x0010: + case 0x0015: + case 0x0016: + case 0x0018: + case 0x0019: + case 0x001c: + case 0x001f: + case 0x0022: + case 0x0023: + case 0x0026: + case 0x0029: + case 0x002a: + case 0x002c: + case 0x002d: + case 0x0040: + case 0x0061: + case 0x0062: + case 0x0063: + case 0x0064: + case 0x0065: + case 0x0066: + case 0x006b: + case 0x006e: + case 0x006f: + case 0x0070: + case 0x0073: + case 0x0074: + case 0x0075: + case 0x0076: + case 0x0077: + case 0x0078: + case 0x0079: + case 0x007a: + case 0x007b: + case 0x0080: + case 0x0081: + case 0x0082: + case 0x0083: + case 0x0084: + case 0x0086: + case 0x0087: + case 0x008a: + case 0x008b: + case 0x008c: + case 0x008e: + case 0x008f: + case 0x0090: + case 0x0091: + case 0x0092: + case 0x0093: + case 0x0094: + case 0x0097: + case 0x0098: + case 0x009a: + case 0x009f: + case 0x00ae: + case 0x00af: + case 0x00b6: + case 0x00b7: + case 0x00b8: + case 0x00ba: + case 0x00bb: + case 0x00be: + case 0x00bf: + case 0x00c0: + case 0x00c1: + case 0x00c5: + case 0x00cb: + case 0x00cc: + case 0x00cd: + case 0x00d9: + case 0x00db: + case 0x00dc: + case 0x00dd: + case 0x00de: + case 0x00df: + case 0x00e0: + case 0x00e1: + case 0x00e2: + case 0x00e4: + case 0x00e5: + case 0x00e6: + case 0x00e7: + case 0x00e8: + case 0x00e9: + case 0x00ea: + case 0x00eb: + case 0x00ec: + case 0x00ed: + case 0x00f4: + case 0x00f6: + case 0x00f8: + case 0x00fa: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x0100: + case 0x0108: + case 0x0109: + case 0x010a: + case 0x010b: + case 0x010c: + case 0x010d: + case 0x010e: + case 0x010f: + case 0x0110: + case 0x0111: + case 0x0112: + case 0x0113: + case 0x0114: + case 0x0116: + case 0x0117: + case 0x0118: + case 0x0125: + case 0x0126: + case 0x0127: + case 0x0128: + case 0x0132: + case 0x0137: + case 0x0139: + case 0x013a: + case 0x013b: + case 0x013c: + case 0x013d: + case 0x013e: + case 0x0140: + case 0x0141: + case 0x0144: + case 0x0145: + case 0x0146: + case 0x0160: + case 0x0161: + case 0x0162: + case 0x0163: + case 0x0164: + case 0x0165: + case 0x0166: + case 0x0167: + case 0x0168: + case 0x0169: + case 0x0180: + case 0x0181: + case 0x0182: + case 0x0183: + case 0x0184: + case 0x0185: + case 0x01b0: + case 0x01b1: + case 0x01b2: + case 0x01b3: + case 0x01b4: + case 0x01b5: + case 0x01b6: + case 0x01b7: + case 0x01b8: + case 0x01c0: + case 0x01c1: + case 0x01c2: + case 0x01c3: + case 0x01c4: + case 0x01c5: + case 0x01c6: + case 0x01c7: + case 0x01c8: + case 0x01c9: + case 0x01ca: + case 0x01cb: + case 0x01cc: + case 0x01cd: + case 0x01ce: + case 0x01cf: + case 0x01d0: + case 0x01d1: + case 0x01d2: + case 0x01d3: + case 0x01d4: + case 0x01d5: + case 0x01d6: + case 0x01d7: + case 0x01d8: + case 0x01d9: + case 0x01da: + case 0x01db: + case 0x01dd: + case 0x01de: + case 0x01df: + case 0x01e0: + case 0x01e1: + case 0x01e2: + case 0x01e3: + case 0x01e4: + case 0x01e5: + case 0x01e6: + case 0x01e7: + case 0x01e8: + case 0x01ea: + case 0x01eb: + case 0x01ec: + case 0x01ed: + case 0x01ee: + case 0x01ef: + case 0x01f0: + case 0x01f1: + case 0x01f2: + case 0x0200: + case 0x0201: + case 0x0202: + case 0x0250: + case 0x0251: + case 0x0252: + case 0x02fa: + case 0x02fb: + case 0x02fc: + case 0x0300: + case 0x03d0: + case 0x03d1: + case 0x03d2: + case 0x03d3: + case 0x03d4: + case 0x03d5: + case 0x03d6: + case 0x03d7: + case 0x03d8: + case 0x03d9: + case 0x03da: + case 0x03db: + case 0x03dc: + case 0x03dd: + case 0x03de: + case 0x03df: + case 0x03e0: + case 0x03e1: + case 0x03e2: + case 0x03e3: + case 0x03e4: + case 0x03e5: + case 0x03e6: + case 0x03e7: + case 0x03e8: + case 0x03e9: + case 0x03ea: + case 0x03eb: + case 0x03ec: + case 0x03ed: + case 0x03ee: + case 0x03ef: + case 0x03f0: + case 0x03f1: + case 0x03f2: + case 0x03f3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(rt5663_hp_vol_tlv, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5668_hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(in_bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5663_if1_adc_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5668_TDM_1, + RT5663_DATA_SWAP_ADCDAT1_SHIFT, rt5663_if1_adc_data_select); + +static void rt5668_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + if (enable) { + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_EN); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + if (rt5668->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5668_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_NOR); + else + snd_soc_update_bits(codec, RT5668_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_NOR); + } else { + if (rt5668->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5668_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_BYP); + else + snd_soc_update_bits(codec, RT5668_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_BYP); + + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_DIS); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + } +} + +/** + * rt5668_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5668_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + if (jack_insert) { + snd_soc_write(codec, RT5668_CBJ_TYPE_2, 0x8040); + snd_soc_write(codec, RT5668_CBJ_TYPE_3, 0x1484); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_update_bits(codec, RT5668_MICBIAS_2, + RT5668_DIG_1M_CLK_MASK, RT5668_DIG_1M_CLK_EN); + snd_soc_update_bits(codec, RT5668_CBJ_TYPE_1, 0x8, 0x8); + + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5668_CBJ_TYPE_2) & 0x0003; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-0011 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + i++; + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5668_enable_push_button_irq(codec, true); + break; + default: + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + rt5668->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + snd_soc_update_bits(codec, RT5668_CBJ_TYPE_1, 0x8, 0x0); + + if (rt5668->jack_type == SND_JACK_HEADSET) { + rt5668_enable_push_button_irq(codec, false); + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + } + rt5668->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +/** + * rt5663_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + + if (jack_insert) { + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & + 0x0003; + i++; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-00e7 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5668_enable_push_button_irq(codec, true); + break; + default: + rt5668->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + if (rt5668->jack_type == SND_JACK_HEADSET) + rt5668_enable_push_button_irq(codec, false); + rt5668->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +int rt5668_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5668_4BTN_IL_CMD_1); + dev_dbg(codec->dev, "%s: val=0x%x\n", __func__, val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5668_4BTN_IL_CMD_1, val); + + return btn_type; +} +EXPORT_SYMBOL(rt5668_button_detect); + +static irqreturn_t rt5668_irq(int irq, void *data) +{ + struct rt5668_priv *rt5668 = data; + + dev_dbg(rt5668->codec->dev, "%s IRQ queue work\n", __func__); + + queue_delayed_work(system_wq, &rt5668->jack_detect_work, + msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5668_btn_check_callback(unsigned long data) +{ + struct rt5668_priv *rt5668 = (struct rt5668_priv *)data; + + queue_delayed_work(system_wq, &rt5668->jack_detect_work, + msecs_to_jiffies(5)); +} + +int rt5668_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + rt5668->hs_jack = hs_jack; + + rt5668_irq(0, rt5668); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5668_set_jack_detect); + +static int rt5668_check_jd_status(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val; + + /* JD1 */ + if (rt5668->codec_type == CODEC_TYPE_RT5668) + val = snd_soc_read(codec, RT5668_INT_ST_1) & 0x2000; + else + val = snd_soc_read(codec, RT5668_INT_ST_1) & 0x1000; + + if (!val) { /* jack insert */ + dev_dbg(codec->dev, "%s-Jack In\n", __func__); + return 1; + } + + /* jack out */ + dev_dbg(codec->dev, "%s-Jack Out\n", __func__); + return 0; +} + +static void rt5668_jack_detect_work(struct work_struct *work) +{ + struct rt5668_priv *rt5668 = + container_of(work, struct rt5668_priv, jack_detect_work.work); + struct snd_soc_codec *codec = rt5668->codec; + int btn_type, report = 0; + + if (!codec) + return; + + if (rt5668_check_jd_status(codec)) { + /* jack in */ + if (rt5668->jack_type == 0) { + /* jack was out, report jack type */ + if (rt5668->codec_type == CODEC_TYPE_RT5668) + report = rt5668_jack_detect(rt5668->codec, 1); + else + report = rt5663_jack_detect(rt5668->codec, 1); + } else { + /* jack is already in, report button event */ + report = SND_JACK_HEADSET; + btn_type = rt5668_button_detect(rt5668->codec); + /** + * rt5668 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5668->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + + /* button release or spurious interrput*/ + if (btn_type == 0) + report = rt5668->jack_type; + else + mod_timer(&rt5668->btn_check_timer, + msecs_to_jiffies(100)); + } + } else { + /* jack out */ + if (rt5668->codec_type == CODEC_TYPE_RT5668) + report = rt5668_jack_detect(rt5668->codec, 0); + else + report = rt5663_jack_detect(rt5668->codec, 0); + } + dev_dbg(codec->dev, "%s jack report: 0x%04x\n", __func__, report); + snd_soc_jack_report(rt5668->hs_jack, report, SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); +} + +static const struct snd_kcontrol_new rt5668_snd_controls[] = { + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC Playback Volume", RT5668_DAC1_DIG_VOL, + RT5668_DAC_L1_VOL_SHIFT + 1, RT5668_DAC_R1_VOL_SHIFT + 1, + 87, 0, dac_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5668_STO1_ADC_DIG_VOL, + RT5668_ADC_L_MUTE_SHIFT, RT5668_ADC_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5668_STO1_ADC_DIG_VOL, + RT5668_ADC_L_VOL_SHIFT + 1, RT5668_ADC_R_VOL_SHIFT + 1, + 63, 0, adc_vol_tlv), +}; + +static const struct snd_kcontrol_new rt5668_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5668_HPL_GAIN, + RT5668_HPR_GAIN, RT5668_GAIN_HP_SHIFT, 15, 1, + rt5668_hp_vol_tlv), + /* Mic Boost Volume */ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5668_AEC_BST, + RT5668_GAIN_CBJ_SHIFT, 8, 0, in_bst_tlv), +}; + +static const struct snd_kcontrol_new rt5663_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, + RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, + rt5663_hp_vol_tlv), + /* Mic Boost Volume*/ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5663_CBJ_2, + RT5663_GAIN_BST1_SHIFT, 8, 0, in_bst_tlv), + /* Data Swap for Slot0/1 in ADCDAT1 */ + SOC_ENUM("IF1 ADC Data Swap", rt5663_if1_adc_enum), +}; + +static int rt5668_is_sys_clk_from_pll(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + val = snd_soc_read(codec, RT5668_GLB_CLK); + val &= RT5668_SCLK_SRC_MASK; + if (val == RT5668_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int rt5668_is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + switch (w->shift) { + case RT5668_ADC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_3; + shift = RT5668_AD_STO1_TRACK_SHIFT; + break; + case RT5668_DAC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_2; + shift = RT5668_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } else { + switch (w->shift) { + case RT5663_ADC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_2; + shift = RT5663_AD_STO1_TRACK_SHIFT; + break; + case RT5663_DAC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_2; + shift = RT5668_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } + + val = (snd_soc_read(codec, reg) >> shift) & 0x7; + + if (val) + return 1; + + return 0; +} + +static int rt5668_i2s_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int da_asrc_en, ad_asrc_en; + + da_asrc_en = (snd_soc_read(codec, RT5668_ASRC_2) & + RT5668_DA_STO1_TRACK_MASK) ? 1 : 0; + if (rt5668->codec_type == CODEC_TYPE_RT5668) + ad_asrc_en = (snd_soc_read(codec, RT5668_ASRC_3) & + RT5668_AD_STO1_TRACK_MASK) ? 1 : 0; + else + ad_asrc_en = (snd_soc_read(codec, RT5668_ASRC_2) & + RT5663_AD_STO1_TRACK_MASK) ? 1 : 0; + + + if (da_asrc_en || ad_asrc_en) + if (rt5668->sysclk > rt5668->lrck * 384) + return 1; + + dev_err(codec->dev, "sysclk < 384 x fs, disable i2s asrc\n"); + + return 0; +} + +/** + * rt5668_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5668 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5668_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc2_mask = 0; + unsigned int asrc2_value = 0; + unsigned int asrc3_mask = 0; + unsigned int asrc3_value = 0; + + switch (clk_src) { + case RT5668_CLK_SEL_SYS: + case RT5668_CLK_SEL_I2S1_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5668_DA_STEREO_FILTER) { + asrc2_mask |= RT5668_DA_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5668_DA_STO1_TRACK_SHIFT; + } + + if (filter_mask & RT5668_AD_STEREO_FILTER) { + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + asrc3_mask |= RT5668_AD_STO1_TRACK_MASK; + asrc3_value |= clk_src << RT5668_AD_STO1_TRACK_SHIFT; + } else { + asrc2_mask |= RT5663_AD_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5663_AD_STO1_TRACK_SHIFT; + } + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5668_ASRC_2, asrc2_mask, + asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5668_ASRC_3, asrc3_mask, + asrc3_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5668_sel_asrc_clk_src); + +/* Analog Mixer */ +static const struct snd_kcontrol_new rt5668_recmix1l[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST2_SHIFT, 1, 1), + SOC_DAPM_SINGLE("BST1 CBJ Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST1_CBJ_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_recmix1r[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1R, + RT5668_RECMIX1R_BST2_SHIFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5668_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_adda_l_mix[] = { + SOC_DAPM_SINGLE("ADC L Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC L Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_adda_r_mix[] = { + SOC_DAPM_SINGLE("ADC R Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_R_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_R_SHIFT, 1, 1), +}; + +/* Out Switch */ +static const struct snd_kcontrol_new rt5668_hpo_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_AMP_2, + RT5668_EN_DAC_HPO_SHIFT, 1, 0); + +/* Stereo ADC source */ +static const char * const rt5668_sto1_adc_src[] = { + "ADC L", "ADC R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcl_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_L_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcl_mux = + SOC_DAPM_ENUM("STO1 ADC L Mux", rt5668_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcr_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_R_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcr_mux = + SOC_DAPM_ENUM("STO1 ADC R Mux", rt5668_sto1_adcr_enum); + +/* RT5663: Analog DACL1 input source */ +static const char * const rt5663_alg_dacl_src[] = { + "DAC L", "STO DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacl_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACL1_SRC_SHIFT, rt5663_alg_dacl_src); + +static const struct snd_kcontrol_new rt5668_alg_dacl_mux = + SOC_DAPM_ENUM("DAC L Mux", rt5663_alg_dacl_enum); + +/* RT5663: Analog DACR1 input source */ +static const char * const rt5663_alg_dacr_src[] = { + "DAC R", "STO DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacr_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACR1_SRC_SHIFT, rt5663_alg_dacr_src); + +static const struct snd_kcontrol_new rt5668_alg_dacr_mux = + SOC_DAPM_ENUM("DAC R Mux", rt5663_alg_dacr_enum); + +static int rt5668_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_HP_CHARGE_PUMP_1, + RT5668_SEL_PM_HP_SHIFT, RT5668_SEL_PM_HP_HIGH); + snd_soc_update_bits(codec, RT5668_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_SILENCE); + } else { + snd_soc_write(codec, RT5668_DEPOP_2, 0x3003); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x000b, + 0x000b); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x0030, + 0x0030); + snd_soc_update_bits(codec, RT5668_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_DIS); + snd_soc_write(codec, RT5668_HP_CHARGE_PUMP_2, 0x1371); + snd_soc_write(codec, RT5663_HP_BIAS, 0xabba); + snd_soc_write(codec, RT5663_CHARGE_PUMP_1, 0x2224); + snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7766); + snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); + snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_write(codec, RT5663_VREF_RECMIX, 0x0007); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x3000, + 0x3000); + } + break; + + case SND_SOC_DAPM_PRE_PMD: + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_REG); + } else { + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x3000, 0x0); + snd_soc_update_bits(codec, RT5668_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_EN); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x0030, 0x0); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x000b, + 0x000b); + } + break; + + default: + return 0; + } + + return 0; +} + +static int rt5668_bst2_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, + RT5668_PWR_BST2 | RT5668_PWR_BST2_OP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5663_pre_div_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0xff00); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0xfffc); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0x0000); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0x0000); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5668_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5668_PWR_ANLG_3, RT5668_PWR_PLL_SHIFT, 0, + NULL, 0), + + /* micbias */ + SND_SOC_DAPM_MICBIAS("MICBIAS1", RT5668_PWR_ANLG_2, + RT5668_PWR_MB1_SHIFT, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS2", RT5668_PWR_ANLG_2, + RT5668_PWR_MB2_SHIFT, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + + /* REC Mixer Power */ + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5668_PWR_ANLG_2, + RT5668_PWR_RECMIX1_SHIFT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC L Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", RT5668_CHOP_ADC, + RT5668_CKGEN_ADCC_SHIFT, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXL", SND_SOC_NOPM, + 0, 0, rt5668_sto1_adc_l_mix, + ARRAY_SIZE(rt5668_sto1_adc_l_mix)), + + /* ADC Filter Power */ + SND_SOC_DAPM_SUPPLY("STO1 ADC Filter", RT5668_PWR_DIG_2, + RT5668_PWR_ADC_S1F_SHIFT, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S", RT5668_PWR_DIG_1, RT5668_PWR_I2S1_SHIFT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "AIF Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("ADDA MIXL", SND_SOC_NOPM, 0, 0, rt5668_adda_l_mix, + ARRAY_SIZE(rt5668_adda_l_mix)), + SND_SOC_DAPM_MIXER("ADDA MIXR", SND_SOC_NOPM, 0, 0, rt5668_adda_r_mix, + ARRAY_SIZE(rt5668_adda_r_mix)), + SND_SOC_DAPM_PGA("DAC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("STO1 DAC Filter", RT5668_PWR_DIG_2, + RT5668_PWR_DAC_S1F_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("STO1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_l_mix, ARRAY_SIZE(rt5668_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("STO1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_r_mix, ARRAY_SIZE(rt5668_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("STO1 DAC L Power", RT5668_PWR_DIG_1, + RT5668_PWR_DAC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("STO1 DAC R Power", RT5668_PWR_DIG_1, + RT5668_PWR_DAC_R1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Headphone*/ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5668_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_widget rt5668_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5668_PWR_ANLG_3, + RT5668_PWR_LDO2_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5668_PWR_VOL, + RT5668_PWR_MIC_DET_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO DAC", RT5668_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5668_ASRC_1, + RT5668_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5668_ASRC_1, + RT5668_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5668_ASRC_1, + RT5668_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5668_PWR_ANLG_3, + RT5668_PWR_CBJ_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BST2 Power", SND_SOC_NOPM, 0, 0, + rt5668_bst2_power, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5668_recmix1l, + ARRAY_SIZE(rt5668_recmix1l)), + SND_SOC_DAPM_MIXER("RECMIX1R", SND_SOC_NOPM, 0, 0, rt5668_recmix1r, + ARRAY_SIZE(rt5668_recmix1r)), + SND_SOC_DAPM_SUPPLY("RECMIX1R Power", RT5668_PWR_ANLG_2, + RT5668_PWR_RECMIX2_SHIFT, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC R Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_R1_SHIFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_PGA("STO1 ADC L1", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_L1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R1", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_R1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_L2_SRC_SHIFT, 1, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R2", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_R2_SRC_SHIFT, 1, NULL, 0), + + SND_SOC_DAPM_MUX("STO1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcl_mux), + SND_SOC_DAPM_MUX("STO1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcr_mux), + + /* ADC Mix */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_adc_r_mix, ARRAY_SIZE(rt5668_sto1_adc_r_mix)), + + /* Analog DAC Clock */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RT5668_CHOP_DAC, + RT5668_CKGEN_DAC1_SHIFT, 0, NULL, 0), + + /* Headphone out */ + SND_SOC_DAPM_SWITCH("HPO Playback", SND_SOC_NOPM, 0, 0, + &rt5668_hpo_switch), +}; + +static const struct snd_soc_dapm_widget rt5663_specific_dapm_widgets[] = { + /* System Clock Pre Divider Gating */ + SND_SOC_DAPM_SUPPLY("Pre Div Power", SND_SOC_NOPM, 0, 0, + rt5663_pre_div_power, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + /* LDO */ + SND_SOC_DAPM_SUPPLY("LDO ADC", RT5668_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5668_ASRC_1, + RT5663_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5668_ASRC_1, + RT5663_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5668_ASRC_1, + RT5663_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* STO ADC */ + SND_SOC_DAPM_PGA("STO1 ADC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog DAC source */ + SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0, &rt5668_alg_dacl_mux), + SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0, &rt5668_alg_dacr_mux), +}; + +static const struct snd_soc_dapm_route rt5668_dapm_routes[] = { + /* PLL */ + { "I2S", NULL, "PLL", rt5668_is_sys_clk_from_pll }, + + /* ASRC */ + { "STO1 ADC Filter", NULL, "ADC ASRC", rt5668_is_using_asrc }, + { "STO1 DAC Filter", NULL, "DAC ASRC", rt5668_is_using_asrc }, + { "I2S", NULL, "I2S ASRC", rt5668_i2s_use_asrc }, + + { "ADC L", NULL, "ADC L Power" }, + { "ADC L", NULL, "ADC Clock" }, + + { "STO1 ADC L2", NULL, "STO1 DAC MIXL" }, + + { "STO1 ADC MIXL", "ADC1 Switch", "STO1 ADC L1" }, + { "STO1 ADC MIXL", "ADC2 Switch", "STO1 ADC L2" }, + { "STO1 ADC MIXL", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXL" }, + { "IF ADC", NULL, "IF1 ADC1" }, + { "AIFTX", NULL, "IF ADC" }, + { "AIFTX", NULL, "I2S" }, + + { "AIFRX", NULL, "I2S" }, + { "IF DAC", NULL, "AIFRX" }, + { "IF1 DAC1 L", NULL, "IF DAC" }, + { "IF1 DAC1 R", NULL, "IF DAC" }, + + { "ADDA MIXL", "ADC L Switch", "STO1 ADC MIXL" }, + { "ADDA MIXL", "DAC L Switch", "IF1 DAC1 L" }, + { "ADDA MIXL", NULL, "STO1 DAC Filter" }, + { "ADDA MIXL", NULL, "STO1 DAC L Power" }, + { "ADDA MIXR", "DAC R Switch", "IF1 DAC1 R" }, + { "ADDA MIXR", NULL, "STO1 DAC Filter" }, + { "ADDA MIXR", NULL, "STO1 DAC R Power" }, + + { "DAC L1", NULL, "ADDA MIXL" }, + { "DAC R1", NULL, "ADDA MIXR" }, + + { "STO1 DAC MIXL", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXL", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC L Power" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC Filter" }, + { "STO1 DAC MIXR", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXR", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC R Power" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC Filter" }, + + { "HP Amp", NULL, "DAC L" }, + { "HP Amp", NULL, "DAC R" }, +}; + +static const struct snd_soc_dapm_route rt5668_specific_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2" }, + { "MICBIAS2", NULL, "LDO2" }, + + { "BST1 CBJ", NULL, "IN1P" }, + { "BST1 CBJ", NULL, "IN1N" }, + { "BST1 CBJ", NULL, "CBJ Power" }, + + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + { "BST2", NULL, "BST2 Power" }, + + { "RECMIX1L", "BST2 Switch", "BST2" }, + { "RECMIX1L", "BST1 CBJ Switch", "BST1 CBJ" }, + { "RECMIX1L", NULL, "RECMIX1L Power" }, + { "RECMIX1R", "BST2 Switch", "BST2" }, + { "RECMIX1R", NULL, "RECMIX1R Power" }, + + { "ADC L", NULL, "RECMIX1L" }, + { "ADC R", NULL, "RECMIX1R" }, + { "ADC R", NULL, "ADC R Power" }, + { "ADC R", NULL, "ADC Clock" }, + + { "STO1 ADC L Mux", "ADC L", "ADC L" }, + { "STO1 ADC L Mux", "ADC R", "ADC R" }, + { "STO1 ADC L1", NULL, "STO1 ADC L Mux" }, + + { "STO1 ADC R Mux", "ADC L", "ADC L" }, + { "STO1 ADC R Mux", "ADC R", "ADC R" }, + { "STO1 ADC R1", NULL, "STO1 ADC R Mux" }, + { "STO1 ADC R2", NULL, "STO1 DAC MIXR" }, + + { "STO1 ADC MIXR", "ADC1 Switch", "STO1 ADC R1" }, + { "STO1 ADC MIXR", "ADC2 Switch", "STO1 ADC R2" }, + { "STO1 ADC MIXR", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXR" }, + + { "ADDA MIXR", "ADC R Switch", "STO1 ADC MIXR" }, + + { "DAC L", NULL, "STO1 DAC MIXL" }, + { "DAC L", NULL, "LDO DAC" }, + { "DAC L", NULL, "DAC Clock" }, + { "DAC R", NULL, "STO1 DAC MIXR" }, + { "DAC R", NULL, "LDO DAC" }, + { "DAC R", NULL, "DAC Clock" }, + + { "HPO Playback", "Switch", "HP Amp" }, + { "HPOL", NULL, "HPO Playback" }, + { "HPOR", NULL, "HPO Playback" }, +}; + +static const struct snd_soc_dapm_route rt5663_specific_dapm_routes[] = { + { "I2S", NULL, "Pre Div Power" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "RECMIX1L Power" }, + + { "ADC L", NULL, "BST1" }, + + { "STO1 ADC L1", NULL, "ADC L" }, + + { "DAC L Mux", "DAC L", "DAC L1" }, + { "DAC L Mux", "STO DAC MIXL", "STO1 DAC MIXL" }, + { "DAC R Mux", "DAC R", "DAC R1"}, + { "DAC R Mux", "STO DAC MIXR", "STO1 DAC MIXR" }, + + { "DAC L", NULL, "DAC L Mux" }, + { "DAC R", NULL, "DAC R Mux" }, + + { "HPOL", NULL, "HP Amp" }, + { "HPOR", NULL, "HP Amp" }, +}; + +static int rt5668_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 rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, frame_size; + + rt5668->lrck = params_rate(params); + pre_div = rl6231_get_clk_info(rt5668->sysclk, rt5668->lrck); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5668->lrck, dai->id); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5668->lrck * (32 << (frame_size > 32) ? 1 : 0), rt5668->lrck); + dev_dbg(dai->dev, "pre_div is %d for iis %d\n", pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5668_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5668_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5668_I2S_DL_8; + break; + default: + return -EINVAL; + } + + mask_clk = RT5668_I2S_PD1_MASK; + val_clk = pre_div << RT5668_I2S_PD1_SHIFT; + snd_soc_update_bits(codec, RT5668_I2S1_SDP, + RT5668_I2S_DL_MASK, val_len); + + snd_soc_update_bits(codec, RT5668_ADDA_CLK_1, mask_clk, val_clk); + + return 0; +} + +static int rt5668_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5668_I2S_MS_S; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5668_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5668_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5668_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5668_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5668_I2S1_SDP, RT5668_I2S_MS_MASK | + RT5668_I2S_BP_MASK | RT5668_I2S_DF_MASK, reg_val); + + return 0; +} + +static int rt5668_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5668->sysclk && clk_id == rt5668->sysclk_src) + return 0; + + switch (clk_id) { + case RT5668_SCLK_S_MCLK: + reg_val |= RT5668_SCLK_SRC_MCLK; + break; + case RT5668_SCLK_S_PLL1: + reg_val |= RT5668_SCLK_SRC_PLL1; + break; + case RT5668_SCLK_S_RCCLK: + reg_val |= RT5668_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5668_GLB_CLK, RT5668_SCLK_SRC_MASK, + reg_val); + rt5668->sysclk = freq; + rt5668->sysclk_src = clk_id; + + dev_dbg(codec->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5668_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + int mask, shift, val; + + if (source == rt5668->pll_src && freq_in == rt5668->pll_in && + freq_out == rt5668->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5668->pll_in = 0; + rt5668->pll_out = 0; + snd_soc_update_bits(codec, RT5668_GLB_CLK, + RT5668_SCLK_SRC_MASK, RT5668_SCLK_SRC_MCLK); + return 0; + } + + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + mask = RT5668_PLL1_SRC_MASK; + shift = RT5668_PLL1_SRC_SHIFT; + } else { + mask = RT5663_PLL1_SRC_MASK; + shift = RT5663_PLL1_SRC_SHIFT; + } + + switch (source) { + case RT5668_PLL1_S_MCLK: + val = 0x0; + break; + case RT5668_PLL1_S_BCLK1: + val = 0x1; + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5668_GLB_CLK, mask, (val << shift)); + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", pll_code.m_bp, + (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code, + pll_code.k_code); + + snd_soc_write(codec, RT5668_PLL_1, + pll_code.n_code << RT5668_PLL_N_SHIFT | pll_code.k_code); + snd_soc_write(codec, RT5668_PLL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5668_PLL_M_SHIFT | + pll_code.m_bp << RT5668_PLL_M_BP_SHIFT); + + rt5668->pll_in = freq_in; + rt5668->pll_out = freq_out; + rt5668->pll_src = source; + + return 0; +} + +static int rt5668_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, reg; + + if (rx_mask || tx_mask) + val |= RT5668_TDM_MODE_TDM; + + switch (slots) { + case 4: + val |= RT5668_TDM_IN_CH_4; + val |= RT5668_TDM_OUT_CH_4; + break; + case 6: + val |= RT5668_TDM_IN_CH_6; + val |= RT5668_TDM_OUT_CH_6; + break; + case 8: + val |= RT5668_TDM_IN_CH_8; + val |= RT5668_TDM_OUT_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= RT5668_TDM_IN_LEN_20; + val |= RT5668_TDM_OUT_LEN_20; + break; + case 24: + val |= RT5668_TDM_IN_LEN_24; + val |= RT5668_TDM_OUT_LEN_24; + break; + case 32: + val |= RT5668_TDM_IN_LEN_32; + val |= RT5668_TDM_OUT_LEN_32; + break; + case 16: + break; + default: + return -EINVAL; + } + + if (rt5668->codec_type == CODEC_TYPE_RT5668) + reg = RT5668_TDM_1; + else + reg = RT5663_TDM_1; + + snd_soc_update_bits(codec, reg, RT5668_TDM_MODE_MASK | + RT5668_TDM_IN_CH_MASK | RT5668_TDM_OUT_CH_MASK | + RT5668_TDM_IN_LEN_MASK | RT5668_TDM_OUT_LEN_MASK, val); + + return 0; +} + +static int rt5668_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s ratio = %d\n", __func__, ratio); + + if (rt5668->codec_type == CODEC_TYPE_RT5663) { + switch (ratio) { + case 32: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_16); + break; + case 40: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_20); + break; + case 48: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_24); + break; + case 64: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_32); + break; + default: + dev_err(codec->dev, "Invalid ratio!\n"); + return -EINVAL; + } + } + + return 0; +} + +static int rt5668_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_1, + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK, + RT5668_PWR_FV1 | RT5668_PWR_FV2); + break; + + case SND_SOC_BIAS_PREPARE: + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_EN); + snd_soc_update_bits(codec, RT5668_MCLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_AUTO | + RT5668_PWR_CLK_DET_EN); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (rt5668->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5668_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_DIS); + snd_soc_update_bits(codec, RT5668_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK | + RT5668_PWR_MB_MASK, RT5668_PWR_VREF1 | + RT5668_PWR_VREF2 | RT5668_PWR_MB); + usleep_range(10000, 10005); + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_MCLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_DIS | + RT5668_PWR_CLK_DET_DIS); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1 | RT5668_PWR_FV2, 0x0); + break; + + default: + break; + } + + return 0; +} + +static int rt5668_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + rt5668->codec = codec; + + switch (rt5668->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_dapm_new_controls(dapm, + rt5668_specific_dapm_widgets, + ARRAY_SIZE(rt5668_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5668_specific_dapm_routes, + ARRAY_SIZE(rt5668_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5668_specific_controls, + ARRAY_SIZE(rt5668_specific_controls)); + break; + case CODEC_TYPE_RT5663: + snd_soc_dapm_new_controls(dapm, + rt5663_specific_dapm_widgets, + ARRAY_SIZE(rt5663_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5663_specific_dapm_routes, + ARRAY_SIZE(rt5663_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5663_specific_controls, + ARRAY_SIZE(rt5663_specific_controls)); + break; + } + + return 0; +} + +static int rt5668_remove(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5668->regmap, RT5668_RESET, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5668_suspend(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5668->regmap, true); + regcache_mark_dirty(rt5668->regmap); + + return 0; +} + +static int rt5668_resume(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5668->regmap, false); + regcache_sync(rt5668->regmap); + + return 0; +} +#else +#define rt5668_suspend NULL +#define rt5668_resume NULL +#endif + +#define RT5668_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5668_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5668_aif_dai_ops = { + .hw_params = rt5668_hw_params, + .set_fmt = rt5668_set_dai_fmt, + .set_sysclk = rt5668_set_dai_sysclk, + .set_pll = rt5668_set_dai_pll, + .set_tdm_slot = rt5668_set_tdm_slot, + .set_bclk_ratio = rt5668_set_bclk_ratio, +}; + +struct snd_soc_dai_driver rt5668_dai[] = { + { + .name = "rt5668-aif", + .id = RT5668_AIF, + .playback = { + .stream_name = "AIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .capture = { + .stream_name = "AIF Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .ops = &rt5668_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5668 = { + .probe = rt5668_probe, + .remove = rt5668_remove, + .suspend = rt5668_suspend, + .resume = rt5668_resume, + .set_bias_level = rt5668_set_bias_level, + .idle_bias_off = true, + .controls = rt5668_snd_controls, + .num_controls = ARRAY_SIZE(rt5668_snd_controls), + .dapm_widgets = rt5668_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5668_dapm_widgets), + .dapm_routes = rt5668_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5668_dapm_routes), +}; + +static const struct regmap_config rt5668_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x07fa, + .volatile_reg = rt5668_volatile_register, + .readable_reg = rt5668_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5668_reg, + .num_reg_defaults = ARRAY_SIZE(rt5668_reg), +}; + +static const struct regmap_config rt5663_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .volatile_reg = rt5663_volatile_register, + .readable_reg = rt5663_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5663_reg, + .num_reg_defaults = ARRAY_SIZE(rt5663_reg), +}; + +static const struct regmap_config temp_regmap = { + .name = "nocache", + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt5668_i2c_id[] = { + { "rt5668", 0 }, + { "rt5663", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5668_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5668_of_match[] = { + { .compatible = "realtek,rt5668", }, + { .compatible = "realtek,rt5663", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5668_of_match); +#endif + +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5668_acpi_match[] = { + { "10EC5668", 0}, + { "10EC5663", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5668_acpi_match); +#endif + +static void rt5668_calibrate(struct rt5668_priv *rt5668) +{ + regmap_write(rt5668->regmap, 0x0111, 0xa402); + regmap_write(rt5668->regmap, 0x0061, 0x0100); + regmap_write(rt5668->regmap, 0x0010, 0x4040); + regmap_write(rt5668->regmap, 0x00fa, 0x0001); + regmap_write(rt5668->regmap, 0x0094, 0x0380); + regmap_write(rt5668->regmap, 0x0080, 0x8000); + regmap_write(rt5668->regmap, 0x0073, 0x1000); + regmap_write(rt5668->regmap, 0x013a, 0x3030); + regmap_write(rt5668->regmap, 0x013c, 0x3c05); + regmap_write(rt5668->regmap, 0x0063, 0xa23e); + msleep(40); + regmap_write(rt5668->regmap, 0x0063, 0xf23e); + regmap_write(rt5668->regmap, 0x01df, 0x0321); + regmap_write(rt5668->regmap, 0x01de, 0xfc00); + msleep(500); +} + +static void rt5663_calibrate(struct rt5668_priv *rt5668) +{ + regmap_write(rt5668->regmap, 0x00fa, 0x8001); + regmap_write(rt5668->regmap, 0x0063, 0xa2b8); + msleep(50); + regmap_write(rt5668->regmap, 0x0062, 0x8400); + regmap_write(rt5668->regmap, 0x0061, 0x8df8); + regmap_write(rt5668->regmap, 0x0064, 0x0003); + regmap_write(rt5668->regmap, 0x0065, 0x018c); + regmap_write(rt5668->regmap, 0x0073, 0x0000); + regmap_write(rt5668->regmap, 0x006e, 0xffff); + regmap_write(rt5668->regmap, 0x006f, 0xffff); + regmap_write(rt5668->regmap, 0x008f, 0x3003); + regmap_write(rt5668->regmap, 0x008e, 0x003b); + regmap_write(rt5668->regmap, 0x0091, 0x1e32); + regmap_write(rt5668->regmap, 0x0092, 0x1371); + regmap_write(rt5668->regmap, 0x0112, 0x3b0b); + regmap_write(rt5668->regmap, 0x002a, 0x2080); + regmap_write(rt5668->regmap, 0x002d, 0x000c); + regmap_write(rt5668->regmap, 0x0202, 0xabba); + regmap_write(rt5668->regmap, 0x0125, 0x2224); + regmap_write(rt5668->regmap, 0x0002, 0x8088); + regmap_write(rt5668->regmap, 0x0168, 0x0017); + regmap_write(rt5668->regmap, 0x0169, 0x0017); + regmap_write(rt5668->regmap, 0x0026, 0x4040); + regmap_write(rt5668->regmap, 0x013b, 0x3000); + regmap_write(rt5668->regmap, 0x0010, 0x0005); + regmap_write(rt5668->regmap, 0x0074, 0xc000); + regmap_write(rt5668->regmap, 0x00ec, 0x3320); + regmap_write(rt5668->regmap, 0x01df, 0x00c9); + regmap_write(rt5668->regmap, 0x02fa, 0x004c); + regmap_write(rt5668->regmap, 0x0108, 0x7766); + regmap_write(rt5668->regmap, 0x0111, 0x4702); + regmap_write(rt5668->regmap, 0x0063, 0xf2b8); + regmap_write(rt5668->regmap, 0x0118, 0x0007); + msleep(200); + regmap_write(rt5668->regmap, 0x01dd, 0x0069); + regmap_write(rt5668->regmap, 0x01e0, 0x06c2); + regmap_write(rt5668->regmap, 0x01de, 0x7b00); + regmap_write(rt5668->regmap, 0x01de, 0xfb00); + msleep(500); +} + +static int rt5668_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5668_priv *rt5668; + int ret; + unsigned int val; + struct regmap *regmap; + + rt5668 = devm_kzalloc(&i2c->dev, sizeof(struct rt5668_priv), + GFP_KERNEL); + + if (rt5668 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5668); + + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&i2c->dev, "Failed to allocate temp register map: %d\n", + ret); + return ret; + } + regmap_read(regmap, RT5668_VENDOR_ID_2, &val); + switch (val) { + case RT5668_DEVICE_ID: + rt5668->regmap = devm_regmap_init_i2c(i2c, &rt5668_regmap); + rt5668->codec_type = CODEC_TYPE_RT5668; + break; + case RT5663_DEVICE_ID: + rt5668->regmap = devm_regmap_init_i2c(i2c, &rt5663_regmap); + rt5668->codec_type = CODEC_TYPE_RT5663; + break; + default: + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663 or rt5668\n", + val); + return -ENODEV; + } + + if (IS_ERR(rt5668->regmap)) { + ret = PTR_ERR(rt5668->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + /* reset and calibrate */ + regmap_write(rt5668->regmap, RT5668_RESET, 0); + regcache_cache_bypass(rt5668->regmap, true); + if (rt5668->codec_type == CODEC_TYPE_RT5668) + rt5668_calibrate(rt5668); + else + rt5663_calibrate(rt5668); + regcache_cache_bypass(rt5668->regmap, false); + regmap_write(rt5668->regmap, RT5668_RESET, 0); + dev_dbg(&i2c->dev, "calibrate done\n"); + + /* GPIO1 as IRQ */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_1, RT5668_GP1_PIN_MASK, + RT5668_GP1_PIN_IRQ); + /* 4btn inline command debounce */ + regmap_update_bits(rt5668->regmap, RT5668_4BTN_IL_CMD_1, + RT5668_4BTN_CLK_DEB_MASK, RT5668_4BTN_CLK_DEB_65MS); + + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + regmap_write(rt5668->regmap, RT5668_BIAS_CUR_8, 0xa402); + /* JD1 */ + regmap_update_bits(rt5668->regmap, RT5668_AUTO_RC_CLK, + RT5668_IRQ_POW_SAV_MASK | RT5668_IRQ_POW_SAV_JD1_MASK, + RT5668_IRQ_POW_SAV_EN | RT5668_IRQ_POW_SAV_JD1_EN); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_2, + RT5668_PWR_JD1_MASK, RT5668_PWR_JD1); + regmap_update_bits(rt5668->regmap, RT5668_IRQ_1, + RT5668_EN_CB_JD_MASK, RT5668_EN_CB_JD_EN); + + regmap_update_bits(rt5668->regmap, RT5668_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, RT5668_HP_SIG_SRC1_REG); + regmap_update_bits(rt5668->regmap, RT5668_CBJ_TYPE_1, + RT5668_VREF_BIAS_MASK | RT5668_CBJ_DET_MASK | + RT5668_DET_TYPE_MASK, RT5668_VREF_BIAS_REG | + RT5668_CBJ_DET_EN | RT5668_DET_TYPE_QFN); + /* Set GPIO4 and GPIO8 as input for combo jack */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_2, + RT5668_GP4_PIN_CONF_MASK, RT5668_GP4_PIN_CONF_INPUT); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_3, + RT5668_GP8_PIN_CONF_MASK, RT5668_GP8_PIN_CONF_INPUT); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_LDO1_DVO_MASK | RT5668_AMP_HP_MASK, + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + } else { + regmap_update_bits(rt5668->regmap, RT5668_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, RT5668_DIG_GATE_CTRL_EN); + regmap_update_bits(rt5668->regmap, RT5668_HP_CHARGE_PUMP_1, + RT5663_SI_HP_MASK | RT5668_OSW_HP_L_MASK | + RT5668_OSW_HP_R_MASK, RT5663_SI_HP_EN | + RT5668_OSW_HP_L_DIS | RT5668_OSW_HP_R_DIS); + regmap_update_bits(rt5668->regmap, RT5663_DUMMY_1, + RT5663_EMB_CLK_MASK | RT5663_HPA_CPL_BIAS_MASK | + RT5663_HPA_CPR_BIAS_MASK, RT5663_EMB_CLK_EN | + RT5663_HPA_CPL_BIAS_1 | RT5663_HPA_CPR_BIAS_1); + regmap_update_bits(rt5668->regmap, RT5663_CBJ_1, + RT5663_INBUF_CBJ_BST1_MASK | RT5663_CBJ_SENSE_BST1_MASK, + RT5663_INBUF_CBJ_BST1_ON | RT5663_CBJ_SENSE_BST1_L); + regmap_update_bits(rt5668->regmap, RT5668_IL_CMD_2, + RT5663_PWR_MIC_DET_MASK, RT5663_PWR_MIC_DET_ON); + /* BST1 power on for JD */ + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_2, + RT5668_PWR_BST1_MASK, RT5668_PWR_BST1_ON); + regmap_update_bits(rt5668->regmap, RT5663_EM_JACK_TYPE_1, + RT5663_CBJ_DET_MASK | RT5663_EXT_JD_MASK | + RT5663_POL_EXT_JD_MASK, RT5663_CBJ_DET_EN | + RT5663_EXT_JD_EN | RT5663_POL_EXT_JD_EN); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB_MASK | RT5668_LDO1_DVO_MASK | + RT5668_AMP_HP_MASK, RT5668_PWR_MB | + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + regmap_update_bits(rt5668->regmap, RT5668_AUTO_RC_CLK, + RT5668_IRQ_POW_SAV_MASK, RT5668_IRQ_POW_SAV_EN); + regmap_update_bits(rt5668->regmap, RT5668_IRQ_1, + RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_2, + RT5663_GP1_PIN_CONF_MASK, RT5663_GP1_PIN_CONF_OUTPUT); + /* DACREF LDO control */ + regmap_update_bits(rt5668->regmap, RT5663_DACREF_LDO, 0x3e0e, + 0x3a0a); + /* RECMIX */ + regmap_update_bits(rt5668->regmap, RT5663_RECMIX, + RT5663_RECMIX1_BST1_MASK, RT5663_RECMIX1_BST1_UNMUTE); + /* ADCDAT L/L */ + regmap_update_bits(rt5668->regmap, RT5668_TDM_1, + RT5663_DATA_SWAP_ADCDAT1_MASK, + RT5663_DATA_SWAP_ADCDAT1_LL); + } + + setup_timer(&rt5668->btn_check_timer, rt5668_btn_check_callback, + (unsigned long)rt5668); + INIT_DELAYED_WORK(&rt5668->jack_detect_work, rt5668_jack_detect_work); + + if (i2c->irq) { + ret = request_threaded_irq(i2c->irq, NULL, rt5668_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5668", rt5668); + if (ret) + dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", + __func__, ret); + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5668, + rt5668_dai, ARRAY_SIZE(rt5668_dai)); + + if (ret) { + if (i2c->irq) + free_irq(i2c->irq, rt5668); + } + + return ret; +} + +static int rt5668_i2c_remove(struct i2c_client *i2c) +{ + struct rt5668_priv *rt5668 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt5668); + + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +void rt5668_i2c_shutdown(struct i2c_client *client) +{ + struct rt5668_priv *rt5668 = i2c_get_clientdata(client); + + regmap_write(rt5668->regmap, RT5668_RESET, 0); +} + +struct i2c_driver rt5668_i2c_driver = { + .driver = { + .name = "rt5668", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5668_acpi_match), + .of_match_table = of_match_ptr(rt5668_of_match), + }, + .probe = rt5668_i2c_probe, + .remove = rt5668_i2c_remove, + .shutdown = rt5668_i2c_shutdown, + .id_table = rt5668_i2c_id, +}; +module_i2c_driver(rt5668_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5668 driver"); +MODULE_AUTHOR("John Lin <john.lin(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5668.h b/sound/soc/codecs/rt5668.h new file mode 100644 index 0000000..717d5af --- /dev/null +++ b/sound/soc/codecs/rt5668.h @@ -0,0 +1,781 @@ +/* + * rt5668.h -- RT5668 ALSA SoC audio driver + * + * Copyright 2015 Realtek Microelectronics + * Author: Jack Yu <jack.yu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5668_H__ +#define __RT5668_H__ + +/* Info */ +#define RT5668_RESET 0x0000 +#define RT5668_VENDOR_ID 0x00fd +#define RT5668_VENDOR_ID_1 0x00fe +#define RT5668_VENDOR_ID_2 0x00ff + +#define RT5668_HP_AMP_1 0x0002 +#define RT5668_HP_AMP_2 0x0003 +#define RT5668_HPL_GAIN 0x0005 +#define RT5668_HPR_GAIN 0x0006 + +#define RT5668_AEC_BST 0x000b +#define RT5668_IN1_IN2 0x000c +#define RT5668_CBJ_TYPE_1 0x0010 +#define RT5668_CBJ_TYPE_2 0x0011 +#define RT5668_CBJ_TYPE_3 0x0012 + +/* I/O - ADC/DAC/DMIC */ +#define RT5668_DAC1_DIG_VOL 0x0019 +#define RT5668_STO1_ADC_DIG_VOL 0x001c +#define RT5668_STO1_BOOST 0x001f +#define RT5668_HP_IMP_GAIN_1 0x0022 +#define RT5668_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5668_STO1_ADC_MIXER 0x0026 +#define RT5668_AD_DA_MIXER 0x0029 +#define RT5668_STO_DAC_MIXER 0x002a + +/*Mixer - Analog*/ +#define RT5668_RECMIX1L 0x003c +#define RT5668_RECMIX1R 0x003e + +/* Power */ +#define RT5668_PWR_DIG_1 0x0061 +#define RT5668_PWR_DIG_2 0x0062 +#define RT5668_PWR_ANLG_1 0x0063 +#define RT5668_PWR_ANLG_2 0x0064 +#define RT5668_PWR_ANLG_3 0x0065 +#define RT5668_PWR_MIXER 0x0066 +#define RT5668_PWR_VOL 0x0067 + +#define RT5668_MCLK_DET 0x006b +/* Format - ADC/DAC */ +#define RT5668_I2S1_SDP 0x0070 +#define RT5668_ADDA_CLK_1 0x0073 +/* Format - TDM Control */ +#define RT5668_TDM_1 0x0078 +#define RT5668_TDM_2 0x0079 +#define RT5668_TDM_3 0x007a +#define RT5668_TDM_4 0x007b +#define RT5668_TDM_5 0x007c +#define RT5668_TDM_6 0x007d +#define RT5668_TDM_7 0x007e +#define RT5668_TDM_8 0x007f + +/* Function - Analog */ +#define RT5668_GLB_CLK 0x0080 +#define RT5668_PLL_1 0x0081 +#define RT5668_PLL_2 0x0082 +#define RT5668_ASRC_1 0x0083 +#define RT5668_ASRC_2 0x0084 +#define RT5668_ASRC_3 0x0085 +#define RT5668_ASRC_8 0x008a +#define RT5668_ASRC_9 0x008b +#define RT5668_ASRC_11 0x008c +#define RT5668_DEPOP_1 0x008e +#define RT5668_DEPOP_2 0x008f +#define RT5668_HP_CHARGE_PUMP_1 0x0091 +#define RT5668_HP_CHARGE_PUMP_2 0x0092 +#define RT5668_MICBIAS_2 0x0094 +#define RT5668_AUTO_RC_CLK 0x009f + +/* Function - Digital */ +#define RT5668_ADC_EQ_1 0x00ae +#define RT5668_ADC_EQ_2 0x00af + +#define RT5668_IRQ_1 0x00b6 +#define RT5668_IRQ_2 0x00b7 +#define RT5668_IRQ_3 0x00b8 +#define RT5668_IRQ_4 0x00b9 +#define RT5668_IRQ_5 0x00ba +#define RT5668_IRQ_6 0x00bb +#define RT5668_INT_ST_1 0x00be +#define RT5668_GPIO_1 0x00c0 +#define RT5668_GPIO_2 0x00c1 +#define RT5668_GPIO_3 0x00c2 +#define RT5668_IL_CMD_1 0x00db +#define RT5668_IL_CMD_2 0x00dc +#define RT5668_IL_CMD_3 0x00dd +#define RT5668_IL_CMD_4 0x00de +#define RT5668_4BTN_IL_CMD_1 0x00df +#define RT5668_4BTN_IL_CMD_2 0x00e0 +#define RT5668_4BTN_IL_CMD_3 0x00e1 +/* General Control */ +#define RT5668_DIG_MISC 0x00fa + +#define RT5668_BIAS_CUR_8 0x0111 +#define RT5668_CHOP_DAC 0x013a +#define RT5668_CHOP_ADC 0x013b +#define RT5668_CALIB_ADC 0x013c +#define RT5668_HP_LOGIC_1 0x01da +#define RT5668_HP_LOGIC_2 0x01db +#define RT5668_HP_CALIB_1 0x01de +#define RT5668_HP_CALIB_2 0x01df +#define RT5668_HP_CALIB_3 0x01e0 +#define RT5668_HP_CALIB_4 0x01e1 +#define RT5668_HP_CALIB_5 0x01e2 +#define RT5668_HP_CALIB_6 0x01e3 +#define RT5668_HP_CALIB_7 0x01e4 +#define RT5668_HP_CALIB_8 0x01e5 +#define RT5668_HP_CALIB_9 0x01e6 +#define RT5668_HP_CALIB_10 0x01e7 + +/* Headphone Amp Control 2 (0x0003) */ +#define RT5668_EN_DAC_HPO_MASK (0x1 << 14) +#define RT5668_EN_DAC_HPO_SHIFT 14 +#define RT5668_EN_DAC_HPO_DIS (0x0 << 14) +#define RT5668_EN_DAC_HPO_EN (0x1 << 14) + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5668_GAIN_HP (0x1f << 8) +#define RT5668_GAIN_HP_SHIFT 8 + +/* AEC BST Control (0x000b) */ +#define RT5668_GAIN_CBJ_MASK (0xf << 8) +#define RT5668_GAIN_CBJ_SHIFT 8 + +/* IN1 Control / MIC GND REF (0x000c) */ +#define RT5668_IN1_DF_MASK (0x1 << 15) +#define RT5668_IN1_DF_SHIFT 15 + +/* Combo Jack and Type Detection Control 1 (0x0010) */ +#define RT5668_CBJ_DET_MASK (0x1 << 15) +#define RT5668_CBJ_DET_SHIFT 15 +#define RT5668_CBJ_DET_DIS (0x0 << 15) +#define RT5668_CBJ_DET_EN (0x1 << 15) +#define RT5668_DET_TYPE_MASK (0x1 << 12) +#define RT5668_DET_TYPE_SHIFT 12 +#define RT5668_DET_TYPE_WLCSP (0x0 << 12) +#define RT5668_DET_TYPE_QFN (0x1 << 12) +#define RT5668_VREF_BIAS_MASK (0x1 << 6) +#define RT5668_VREF_BIAS_SHIFT 6 +#define RT5668_VREF_BIAS_FSM (0x0 << 6) +#define RT5668_VREF_BIAS_REG (0x1 << 6) + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5668_RECMIX1L_BST1_CBJ (0x1 << 7) +#define RT5668_RECMIX1L_BST1_CBJ_SHIFT 7 +#define RT5668_RECMIX1L_BST2 (0x1 << 4) +#define RT5668_RECMIX1L_BST2_SHIFT 4 + +/* REC Right Mixer Control 2 (0x003e) */ +#define RT5668_RECMIX1R_BST2 (0x1 << 4) +#define RT5668_RECMIX1R_BST2_SHIFT 4 + +/* DAC1 Digital Volume (0x0019) */ +#define RT5668_DAC_L1_VOL_MASK (0xff << 8) +#define RT5668_DAC_L1_VOL_SHIFT 8 +#define RT5668_DAC_R1_VOL_MASK (0xff) +#define RT5668_DAC_R1_VOL_SHIFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5668_ADC_L_MUTE_MASK (0x1 << 15) +#define RT5668_ADC_L_MUTE_SHIFT 15 +#define RT5668_ADC_L_VOL_MASK (0x7f << 8) +#define RT5668_ADC_L_VOL_SHIFT 8 +#define RT5668_ADC_R_MUTE_MASK (0x1 << 7) +#define RT5668_ADC_R_MUTE_SHIFT 7 +#define RT5668_ADC_R_VOL_MASK (0x7f) +#define RT5668_ADC_R_VOL_SHIFT 0 + +/* Stereo ADC Mixer Control (0x0026) */ +#define RT5668_M_STO1_ADC_L1 (0x1 << 15) +#define RT5668_M_STO1_ADC_L1_SHIFT 15 +#define RT5668_M_STO1_ADC_L2 (0x1 << 14) +#define RT5668_M_STO1_ADC_L2_SHIFT 14 +#define RT5668_STO1_ADC_L1_SRC (0x1 << 13) +#define RT5668_STO1_ADC_L1_SRC_SHIFT 13 +#define RT5668_STO1_ADC_L2_SRC (0x1 << 12) +#define RT5668_STO1_ADC_L2_SRC_SHIFT 12 +#define RT5668_STO1_ADC_L_SRC (0x3 << 10) +#define RT5668_STO1_ADC_L_SRC_SHIFT 10 +#define RT5668_M_STO1_ADC_R1 (0x1 << 7) +#define RT5668_M_STO1_ADC_R1_SHIFT 7 +#define RT5668_M_STO1_ADC_R2 (0x1 << 6) +#define RT5668_M_STO1_ADC_R2_SHIFT 6 +#define RT5668_STO1_ADC_R1_SRC (0x1 << 5) +#define RT5668_STO1_ADC_R1_SRC_SHIFT 5 +#define RT5668_STO1_ADC_R2_SRC (0x1 << 4) +#define RT5668_STO1_ADC_R2_SRC_SHIFT 4 +#define RT5668_STO1_ADC_R_SRC (0x3 << 2) +#define RT5668_STO1_ADC_R_SRC_SHIFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5668_M_ADCMIX_L (0x1 << 15) +#define RT5668_M_ADCMIX_L_SHIFT 15 +#define RT5668_M_DAC1_L (0x1 << 14) +#define RT5668_M_DAC1_L_SHIFT 14 +#define RT5668_M_ADCMIX_R (0x1 << 7) +#define RT5668_M_ADCMIX_R_SHIFT 7 +#define RT5668_M_DAC1_R (0x1 << 6) +#define RT5668_M_DAC1_R_SHIFT 6 + +/* Stereo DAC Mixer Control (0x002a) */ +#define RT5668_M_DAC_L1_STO_L (0x1 << 15) +#define RT5668_M_DAC_L1_STO_L_SHIFT 15 +#define RT5668_M_DAC_R1_STO_L (0x1 << 13) +#define RT5668_M_DAC_R1_STO_L_SHIFT 13 +#define RT5668_M_DAC_L1_STO_R (0x1 << 7) +#define RT5668_M_DAC_L1_STO_R_SHIFT 7 +#define RT5668_M_DAC_R1_STO_R (0x1 << 5) +#define RT5668_M_DAC_R1_STO_R_SHIFT 5 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5668_PWR_I2S1 (0x1 << 15) +#define RT5668_PWR_I2S1_SHIFT 15 +#define RT5668_PWR_DAC_L1 (0x1 << 11) +#define RT5668_PWR_DAC_L1_SHIFT 11 +#define RT5668_PWR_DAC_R1 (0x1 << 10) +#define RT5668_PWR_DAC_R1_SHIFT 10 +#define RT5668_PWR_LDO_DACREF_MASK (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_SHIFT 8 +#define RT5668_PWR_LDO_DACREF_ON (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_DOWN (0x0 << 8) +#define RT5668_PWR_LDO_SHIFT 8 +#define RT5668_PWR_ADC_L1 (0x1 << 4) +#define RT5668_PWR_ADC_L1_SHIFT 4 +#define RT5668_PWR_ADC_R1 (0x1 << 3) +#define RT5668_PWR_ADC_R1_SHIFT 3 + +/* Power Management for Digital 2 (0x0062) */ +#define RT5668_PWR_ADC_S1F (0x1 << 15) +#define RT5668_PWR_ADC_S1F_SHIFT 15 +#define RT5668_PWR_DAC_S1F (0x1 << 10) +#define RT5668_PWR_DAC_S1F_SHIFT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5668_PWR_VREF1 (0x1 << 15) +#define RT5668_PWR_VREF1_MASK (0x1 << 15) +#define RT5668_PWR_VREF1_SHIFT 15 +#define RT5668_PWR_FV1 (0x1 << 14) +#define RT5668_PWR_FV1_MASK (0x1 << 14) +#define RT5668_PWR_FV1_SHIFT 14 +#define RT5668_PWR_VREF2 (0x1 << 13) +#define RT5668_PWR_VREF2_MASK (0x1 << 13) +#define RT5668_PWR_VREF2_SHIFT 13 +#define RT5668_PWR_FV2 (0x1 << 12) +#define RT5668_PWR_FV2_MASK (0x1 << 12) +#define RT5668_PWR_FV2_SHIFT 12 +#define RT5668_PWR_MB (0x1 << 9) +#define RT5668_PWR_MB_MASK (0x1 << 9) +#define RT5668_PWR_MB_SHIFT 9 +#define RT5668_AMP_HP_MASK (0x3 << 2) +#define RT5668_AMP_HP_SHIFT 2 +#define RT5668_AMP_HP_1X (0x0 << 2) +#define RT5668_AMP_HP_3X (0x1 << 2) +#define RT5668_AMP_HP_5X (0x3 << 2) +#define RT5668_LDO1_DVO_MASK (0x3) +#define RT5668_LDO1_DVO_SHIFT 0 +#define RT5668_LDO1_DVO_0_9V (0x0) +#define RT5668_LDO1_DVO_1_0V (0x1) +#define RT5668_LDO1_DVO_1_2V (0x2) +#define RT5668_LDO1_DVO_1_4V (0x3) + +/* Power Management for Analog 2 (0x0064) */ +#define RT5668_PWR_BST1 (0x1 << 15) +#define RT5668_PWR_BST1_MASK (0x1 << 15) +#define RT5668_PWR_BST1_SHIFT 15 +#define RT5668_PWR_BST1_OFF (0x0 << 15) +#define RT5668_PWR_BST1_ON (0x1 << 15) +#define RT5668_PWR_BST2 (0x1 << 14) +#define RT5668_PWR_BST2_MASK (0x1 << 14) +#define RT5668_PWR_BST2_SHIFT 14 +#define RT5668_PWR_MB1 (0x1 << 11) +#define RT5668_PWR_MB1_SHIFT 11 +#define RT5668_PWR_MB2 (0x1 << 10) +#define RT5668_PWR_MB2_SHIFT 10 +#define RT5668_PWR_BST2_OP (0x1 << 6) +#define RT5668_PWR_BST2_OP_MASK (0x1 << 6) +#define RT5668_PWR_BST2_OP_SHIFT 6 +#define RT5668_PWR_JD1 (0x1 << 3) +#define RT5668_PWR_JD1_MASK (0x1 << 3) +#define RT5668_PWR_JD1_SHIFT 3 +#define RT5668_PWR_JD2 (0x1 << 2) +#define RT5668_PWR_JD2_MASK (0x1 << 2) +#define RT5668_PWR_JD2_SHIFT 2 +#define RT5668_PWR_RECMIX1 (0x1 << 1) +#define RT5668_PWR_RECMIX1_SHIFT 1 +#define RT5668_PWR_RECMIX2 (0x1) +#define RT5668_PWR_RECMIX2_SHIFT 0 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5668_PWR_CBJ_MASK (0x1 << 9) +#define RT5668_PWR_CBJ_SHIFT 9 +#define RT5668_PWR_CBJ_OFF (0x0 << 9) +#define RT5668_PWR_CBJ_ON (0x1 << 9) +#define RT5668_PWR_PLL (0x1 << 6) +#define RT5668_PWR_PLL_SHIFT 6 +#define RT5668_PWR_LDO2 (0x1 << 2) +#define RT5668_PWR_LDO2_SHIFT 2 + +/* Power Management for Volume (0x0067) */ +#define RT5668_PWR_MIC_DET (0x1 << 5) +#define RT5668_PWR_MIC_DET_SHIFT 5 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5668_EN_ANA_CLK_DET_MASK (0x1 << 15) +#define RT5668_EN_ANA_CLK_DET_SHIFT 15 +#define RT5668_EN_ANA_CLK_DET_DIS (0x0 << 15) +#define RT5668_EN_ANA_CLK_DET_AUTO (0x1 << 15) +#define RT5668_PWR_CLK_DET_MASK (0x1) +#define RT5668_PWR_CLK_DET_SHIFT 0 +#define RT5668_PWR_CLK_DET_DIS (0x0) +#define RT5668_PWR_CLK_DET_EN (0x1) + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5668_I2S_MS_MASK (0x1 << 15) +#define RT5668_I2S_MS_SHIFT 15 +#define RT5668_I2S_MS_M (0x0 << 15) +#define RT5668_I2S_MS_S (0x1 << 15) +#define RT5668_I2S_BP_MASK (0x1 << 8) +#define RT5668_I2S_BP_SHIFT 8 +#define RT5668_I2S_BP_NOR (0x0 << 8) +#define RT5668_I2S_BP_INV (0x1 << 8) +#define RT5668_I2S_DL_MASK (0x3 << 4) +#define RT5668_I2S_DL_SHIFT 4 +#define RT5668_I2S_DL_16 (0x0 << 4) +#define RT5668_I2S_DL_20 (0x1 << 4) +#define RT5668_I2S_DL_24 (0x2 << 4) +#define RT5668_I2S_DL_8 (0x3 << 4) +#define RT5668_I2S_DF_MASK (0x7) +#define RT5668_I2S_DF_SHIFT 0 +#define RT5668_I2S_DF_I2S (0x0) +#define RT5668_I2S_DF_LEFT (0x1) +#define RT5668_I2S_DF_PCM_A (0x2) +#define RT5668_I2S_DF_PCM_B (0x3) +#define RT5668_I2S_DF_PCM_A_N (0x6) +#define RT5668_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5668_I2S_PD1_MASK (0x7 << 12) +#define RT5668_I2S_PD1_SHIFT 12 +#define RT5668_M_I2S_DIV_MASK (0x7 << 8) +#define RT5668_M_I2S_DIV_SHIFT 8 +#define RT5668_CLK_SRC_MASK (0x3 << 4) +#define RT5668_CLK_SRC_MCLK (0x0 << 4) +#define RT5668_CLK_SRC_PLL_OUT (0x1 << 4) +#define RT5668_CLK_SRC_DIV (0x2 << 4) +#define RT5668_CLK_SRC_RC (0x3 << 4) +#define RT5668_DAC_OSR_MASK (0x3 << 2) +#define RT5668_DAC_OSR_SHIFT 2 +#define RT5668_DAC_OSR_128 (0x0 << 2) +#define RT5668_DAC_OSR_64 (0x1 << 2) +#define RT5668_DAC_OSR_32 (0x2 << 2) +#define RT5668_ADC_OSR_MASK (0x3) +#define RT5668_ADC_OSR_SHIFT 0 +#define RT5668_ADC_OSR_128 (0x0) +#define RT5668_ADC_OSR_64 (0x1) +#define RT5668_ADC_OSR_32 (0x2) + +/* TDM1 control 1 (0x0078) */ +#define RT5668_TDM_MODE_MASK (0x1 << 15) +#define RT5668_TDM_MODE_SHIFT 15 +#define RT5668_TDM_MODE_I2S (0x0 << 15) +#define RT5668_TDM_MODE_TDM (0x1 << 15) +#define RT5668_TDM_IN_CH_MASK (0x3 << 10) +#define RT5668_TDM_IN_CH_SHIFT 10 +#define RT5668_TDM_IN_CH_2 (0x0 << 10) +#define RT5668_TDM_IN_CH_4 (0x1 << 10) +#define RT5668_TDM_IN_CH_6 (0x2 << 10) +#define RT5668_TDM_IN_CH_8 (0x3 << 10) +#define RT5668_TDM_OUT_CH_MASK (0x3 << 8) +#define RT5668_TDM_OUT_CH_SHIFT 8 +#define RT5668_TDM_OUT_CH_2 (0x0 << 8) +#define RT5668_TDM_OUT_CH_4 (0x1 << 8) +#define RT5668_TDM_OUT_CH_6 (0x2 << 8) +#define RT5668_TDM_OUT_CH_8 (0x3 << 8) +#define RT5668_TDM_IN_LEN_MASK (0x3 << 6) +#define RT5668_TDM_IN_LEN_SHIFT 6 +#define RT5668_TDM_IN_LEN_16 (0x0 << 6) +#define RT5668_TDM_IN_LEN_20 (0x1 << 6) +#define RT5668_TDM_IN_LEN_24 (0x2 << 6) +#define RT5668_TDM_IN_LEN_32 (0x3 << 6) +#define RT5668_TDM_OUT_LEN_MASK (0x3 << 4) +#define RT5668_TDM_OUT_LEN_SHIFT 4 +#define RT5668_TDM_OUT_LEN_16 (0x0 << 4) +#define RT5668_TDM_OUT_LEN_20 (0x1 << 4) +#define RT5668_TDM_OUT_LEN_24 (0x2 << 4) +#define RT5668_TDM_OUT_LEN_32 (0x3 << 4) + +/* Global Clock Control (0x0080) */ +#define RT5668_SCLK_SRC_MASK (0x3 << 14) +#define RT5668_SCLK_SRC_SHIFT 14 +#define RT5668_SCLK_SRC_MCLK (0x0 << 14) +#define RT5668_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5668_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5668_PLL1_SRC_MASK (0x7 << 8) +#define RT5668_PLL1_SRC_SHIFT 8 +#define RT5668_PLL1_SRC_MCLK (0x0 << 8) +#define RT5668_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5668_PLL1_PD_MASK (0x1 << 4) +#define RT5668_PLL1_PD_SHIFT 4 + +#define RT5668_PLL_INP_MAX 40000000 +#define RT5668_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5668_PLL_N_MAX 0x001ff +#define RT5668_PLL_N_MASK (RT5668_PLL_N_MAX << 7) +#define RT5668_PLL_N_SHIFT 7 +#define RT5668_PLL_K_MAX 0x001f +#define RT5668_PLL_K_MASK (RT5668_PLL_K_MAX) +#define RT5668_PLL_K_SHIFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5668_PLL_M_MAX 0x00f +#define RT5668_PLL_M_MASK (RT5668_PLL_M_MAX << 12) +#define RT5668_PLL_M_SHIFT 12 +#define RT5668_PLL_M_BP (0x1 << 11) +#define RT5668_PLL_M_BP_SHIFT 11 + +/* PLL tracking mode 1 (0x0083) */ +#define RT5668_I2S1_ASRC_MASK (0x1 << 13) +#define RT5668_I2S1_ASRC_SHIFT 13 +#define RT5668_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5668_DAC_STO1_ASRC_SHIFT 12 +#define RT5668_ADC_STO1_ASRC_MASK (0x1 << 4) +#define RT5668_ADC_STO1_ASRC_SHIFT 4 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5668_DA_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_DA_STO1_TRACK_SHIFT 12 +#define RT5668_DA_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_DA_STO1_TRACK_I2S1 (0x1 << 12) + +/* PLL tracking mode 3 (0x0085)*/ +#define RT5668_AD_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_AD_STO1_TRACK_SHIFT 12 +#define RT5668_AD_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_AD_STO1_TRACK_I2S1 (0x1 << 12) + +/* HPOUT Charge pump control 1 (0x0091) */ +#define RT5668_OSW_HP_L_MASK (0x1 << 11) +#define RT5668_OSW_HP_L_SHIFT 11 +#define RT5668_OSW_HP_L_EN (0x1 << 11) +#define RT5668_OSW_HP_L_DIS (0x0 << 11) +#define RT5668_OSW_HP_R_MASK (0x1 << 10) +#define RT5668_OSW_HP_R_SHIFT 10 +#define RT5668_OSW_HP_R_EN (0x1 << 10) +#define RT5668_OSW_HP_R_DIS (0x0 << 10) +#define RT5668_SEL_PM_HP_MASK (0x3 << 8) +#define RT5668_SEL_PM_HP_SHIFT 8 +#define RT5668_SEL_PM_HP_0_6 (0x0 << 8) +#define RT5668_SEL_PM_HP_0_9 (0x1 << 8) +#define RT5668_SEL_PM_HP_1_8 (0x2 << 8) +#define RT5668_SEL_PM_HP_HIGH (0x3 << 8) +#define RT5668_OVCD_HP_MASK (0x1 << 2) +#define RT5668_OVCD_HP_SHIFT 2 +#define RT5668_OVCD_HP_EN (0x1 << 2) +#define RT5668_OVCD_HP_DIS (0x0 << 2) + +/* RC Clock Control (0x0094) */ +#define RT5668_DIG_25M_CLK_MASK (0x1 << 9) +#define RT5668_DIG_25M_CLK_SHIFT 9 +#define RT5668_DIG_25M_CLK_DIS (0x0 << 9) +#define RT5668_DIG_25M_CLK_EN (0x1 << 9) +#define RT5668_DIG_1M_CLK_MASK (0x1 << 8) +#define RT5668_DIG_1M_CLK_SHIFT 8 +#define RT5668_DIG_1M_CLK_DIS (0x0 << 8) +#define RT5668_DIG_1M_CLK_EN (0x1 << 8) + +/* Auto Turn On 1M RC CLK (0x009f) */ +#define RT5668_IRQ_POW_SAV_MASK (0x1 << 15) +#define RT5668_IRQ_POW_SAV_SHIFT 15 +#define RT5668_IRQ_POW_SAV_DIS (0x0 << 15) +#define RT5668_IRQ_POW_SAV_EN (0x1 << 15) +#define RT5668_IRQ_POW_SAV_JD1_MASK (0x1 << 14) +#define RT5668_IRQ_POW_SAV_JD1_SHIFT 14 +#define RT5668_IRQ_POW_SAV_JD1_DIS (0x0 << 14) +#define RT5668_IRQ_POW_SAV_JD1_EN (0x1 << 14) + +/* IRQ Control 1 (0x00b6) */ +#define RT5668_EN_CB_JD_MASK (0x1 << 3) +#define RT5668_EN_CB_JD_SHIFT 3 +#define RT5668_EN_CB_JD_EN (0x1 << 3) +#define RT5668_EN_CB_JD_DIS (0x0 << 3) + +/* IRQ Control 3 (0x00b8) */ +#define RT5668_EN_IRQ_INLINE_MASK (0x1 << 6) +#define RT5668_EN_IRQ_INLINE_SHIFT 6 +#define RT5668_EN_IRQ_INLINE_BYP (0x0 << 6) +#define RT5668_EN_IRQ_INLINE_NOR (0x1 << 6) + +/* GPIO Control 1 (0x00c0) */ +#define RT5668_GP1_PIN_MASK (0x1 << 15) +#define RT5668_GP1_PIN_SHIFT 15 +#define RT5668_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5668_GP1_PIN_IRQ (0x1 << 15) + +/* GPIO Control 2 (0x00c1) */ +#define RT5668_GP4_PIN_CONF_MASK (0x1 << 5) +#define RT5668_GP4_PIN_CONF_SHIFT 5 +#define RT5668_GP4_PIN_CONF_INPUT (0x0 << 5) +#define RT5668_GP4_PIN_CONF_OUTPUT (0x1 << 5) + +/* GPIO Control 2 (0x00c2) */ +#define RT5668_GP8_PIN_CONF_MASK (0x1 << 13) +#define RT5668_GP8_PIN_CONF_SHIFT 13 +#define RT5668_GP8_PIN_CONF_INPUT (0x0 << 13) +#define RT5668_GP8_PIN_CONF_OUTPUT (0x1 << 13) + +/* 4 Buttons Inline Command Function 1 (0x00df) */ +#define RT5668_4BTN_CLK_DEB_MASK (0x3 << 2) +#define RT5668_4BTN_CLK_DEB_SHIFT 2 +#define RT5668_4BTN_CLK_DEB_8MS (0x0 << 2) +#define RT5668_4BTN_CLK_DEB_16MS (0x1 << 2) +#define RT5668_4BTN_CLK_DEB_32MS (0x2 << 2) +#define RT5668_4BTN_CLK_DEB_65MS (0x3 << 2) + +/* Inline Command Function 6 (0x00e0) */ +#define RT5668_EN_4BTN_INL_MASK (0x1 << 15) +#define RT5668_EN_4BTN_INL_SHIFT 15 +#define RT5668_EN_4BTN_INL_DIS (0x0 << 15) +#define RT5668_EN_4BTN_INL_EN (0x1 << 15) +#define RT5668_RESET_4BTN_INL_MASK (0x1 << 14) +#define RT5668_RESET_4BTN_INL_SHIFT 14 +#define RT5668_RESET_4BTN_INL_RESET (0x0 << 14) +#define RT5668_RESET_4BTN_INL_NOR (0x1 << 14) + +/* Digital Misc Control (0x00fa) */ +#define RT5668_DIG_GATE_CTRL_MASK 0x1 +#define RT5668_DIG_GATE_CTRL_SHIFT (0) +#define RT5668_DIG_GATE_CTRL_DIS 0x0 +#define RT5668_DIG_GATE_CTRL_EN 0x1 + +/* Chopper and Clock control for DAC L (0x013a)*/ +#define RT5668_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5668_CKXEN_DAC1_SHIFT 13 +#define RT5668_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5668_CKGEN_DAC1_SHIFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5668_CKXEN_ADCC_MASK (0x1 << 13) +#define RT5668_CKXEN_ADCC_SHIFT 13 +#define RT5668_CKGEN_ADCC_MASK (0x1 << 12) +#define RT5668_CKGEN_ADCC_SHIFT 12 + +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5668_HP_SIG_SRC1_MASK (0x3) +#define RT5668_HP_SIG_SRC1_SHIFT 0 +#define RT5668_HP_SIG_SRC1_HP_DC (0x0) +#define RT5668_HP_SIG_SRC1_HP_CALIB (0x1) +#define RT5668_HP_SIG_SRC1_REG (0x2) +#define RT5668_HP_SIG_SRC1_SILENCE (0x3) + +/* RT5663 specific register */ +#define RT5663_RECMIX 0x0010 +#define RT5663_BYPASS_STO_DAC 0x002d +#define RT5663_PRE_DIV_GATING_1 0x006e +#define RT5663_PRE_DIV_GATING_2 0x006f +#define RT5663_TDM_1 0x0077 +#define RT5663_TDM_5 0x007b +#define RT5663_EM_JACK_TYPE_1 0x00e6 +#define RT5663_EM_JACK_TYPE_2 0x00e7 +#define RT5663_ANA_BIAS_CUR_1 0x0108 +#define RT5663_DACREF_LDO 0x0112 +#define RT5663_VREF_RECMIX 0x0118 +#define RT5663_CHARGE_PUMP_1 0x0125 +#define RT5663_CHARGE_PUMP_2 0x0128 +#define RT5663_CHOP_DAC_R 0x013d +#define RT5663_STO_DRE_9 0x0168 +#define RT5663_STO_DRE_10 0x0169 +#define RT5663_HP_BIAS 0x0202 +#define RT5663_CBJ_1 0x0250 +#define RT5663_CBJ_2 0x0251 +#define RT5663_DUMMY_1 0x02fa + +/* RT5663: RECMIX Control (0x0010) */ +#define RT5663_RECMIX1_BST1_MASK (0x1) +#define RT5663_RECMIX1_BST1_SHIFT 0 +#define RT5663_RECMIX1_BST1_UNMUTE (0x0) +#define RT5663_RECMIX1_BST1_MUTE (0x1) + +/* RT5663: Bypass Stereo1 DAC Mixer Control (0x002d) */ +#define RT5663_DACL1_SRC_MASK (0x1 << 3) +#define RT5663_DACL1_SRC_SHIFT 3 +#define RT5663_DACR1_SRC_MASK (0x1 << 2) +#define RT5663_DACR1_SRC_SHIFT 2 + +/* RT5663: TDM control 2 (0x0078) */ +#define RT5663_DATA_SWAP_ADCDAT1_MASK (0x3 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_SHIFT 14 +#define RT5663_DATA_SWAP_ADCDAT1_LR (0x0 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RL (0x1 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_LL (0x2 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RR (0x3 << 14) + +/* RT5663: TDM control 5 (0x007b) */ +#define RT5663_TDM_LENGTN_MASK (0x3) +#define RT5663_TDM_LENGTN_SHIFT 0 +#define RT5663_TDM_LENGTN_16 (0x0) +#define RT5663_TDM_LENGTN_20 (0x1) +#define RT5663_TDM_LENGTN_24 (0x2) +#define RT5663_TDM_LENGTN_32 (0x3) + +/* RT5663: Global Clock Control (0x0080) */ +#define RT5663_PLL1_SRC_MASK (0x7 << 11) +#define RT5663_PLL1_SRC_SHIFT 11 +#define RT5663_PLL1_SRC_MCLK (0x0 << 11) +#define RT5663_PLL1_SRC_BCLK1 (0x1 << 11) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5663_I2S1_ASRC_MASK (0x1 << 11) +#define RT5663_I2S1_ASRC_SHIFT 11 +#define RT5663_DAC_STO1_ASRC_MASK (0x1 << 10) +#define RT5663_DAC_STO1_ASRC_SHIFT 10 +#define RT5663_ADC_STO1_ASRC_MASK (0x1 << 3) +#define RT5663_ADC_STO1_ASRC_SHIFT 3 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5663_AD_STO1_TRACK_MASK (0x7) +#define RT5663_AD_STO1_TRACK_SHIFT 0 +#define RT5663_AD_STO1_TRACK_SYSCLK (0x0) +#define RT5663_AD_STO1_TRACK_I2S1 (0x1) + +/* RT5663: HPOUT Charge pump control 1 (0x0091) */ +#define RT5663_SI_HP_MASK (0x1 << 12) +#define RT5663_SI_HP_SHIFT 12 +#define RT5663_SI_HP_EN (0x1 << 12) +#define RT5663_SI_HP_DIS (0x0 << 12) + +/* RT5663: GPIO Control 2 (0x00b6) */ +#define RT5663_GP1_PIN_CONF_MASK (0x1 << 2) +#define RT5663_GP1_PIN_CONF_SHIFT 2 +#define RT5663_GP1_PIN_CONF_OUTPUT (0x1 << 2) +#define RT5663_GP1_PIN_CONF_INPUT (0x0 << 2) + +/* RT5663: GPIO Control 2 (0x00b7) */ +#define RT5663_EN_IRQ_INLINE_MASK (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_SHIFT 3 +#define RT5663_EN_IRQ_INLINE_NOR (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_BYP (0x0 << 3) + +/* RT5663: IRQ Control 1 (0x00c1) */ +#define RT5663_EN_IRQ_JD1_MASK (0x1 << 6) +#define RT5663_EN_IRQ_JD1_SHIFT 6 +#define RT5663_EN_IRQ_JD1_EN (0x1 << 6) +#define RT5663_EN_IRQ_JD1_DIS (0x0 << 6) + +/* RT5663: Inline Command Function 2 (0x00dc) */ +#define RT5663_PWR_MIC_DET_MASK (0x1) +#define RT5663_PWR_MIC_DET_SHIFT 0 +#define RT5663_PWR_MIC_DET_ON (0x1) +#define RT5663_PWR_MIC_DET_OFF (0x0) + +/* RT5663: Embeeded Jack and Type Detection Control 1 (0x00e6)*/ +#define RT5663_CBJ_DET_MASK (0x1 << 15) +#define RT5663_CBJ_DET_SHIFT 15 +#define RT5663_CBJ_DET_DIS (0x0 << 15) +#define RT5663_CBJ_DET_EN (0x1 << 15) +#define RT5663_EXT_JD_MASK (0x1 << 11) +#define RT5663_EXT_JD_SHIFT 11 +#define RT5663_EXT_JD_EN (0x1 << 11) +#define RT5663_EXT_JD_DIS (0x0 << 11) +#define RT5663_POL_EXT_JD_MASK (0x1 << 10) +#define RT5663_POL_EXT_JD_SHIFT 10 +#define RT5663_POL_EXT_JD_EN (0x1 << 10) +#define RT5663_POL_EXT_JD_DIS (0x0 << 10) + +/* RT5663: DACREF LDO Control (0x0112)*/ +#define RT5663_PWR_LDO_DACREFL_MASK (0x1 << 9) +#define RT5663_PWR_LDO_DACREFL_SHIFT 9 +#define RT5663_PWR_LDO_DACREFR_MASK (0x1 << 1) +#define RT5663_PWR_LDO_DACREFR_SHIFT 1 + +/* RT5663: Stereo Dynamic Range Enhancement Control 9 (0x0168, 0x0169)*/ +#define RT5663_DRE_GAIN_HP_MASK (0x1f) +#define RT5663_DRE_GAIN_HP_SHIFT 0 + +/* RT5663: Combo Jack Control (0x0250) */ +#define RT5663_INBUF_CBJ_BST1_MASK (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_SHIFT 11 +#define RT5663_INBUF_CBJ_BST1_ON (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_OFF (0x0 << 11) +#define RT5663_CBJ_SENSE_BST1_MASK (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_SHIFT 10 +#define RT5663_CBJ_SENSE_BST1_L (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_R (0x0 << 10) + +/* RT5663: Combo Jack Control (0x0251) */ +#define RT5663_GAIN_BST1_MASK (0xf) +#define RT5663_GAIN_BST1_SHIFT 0 + +/* RT5663: Dummy register 1 (0x02fa) */ +#define RT5663_EMB_CLK_MASK (0x1 << 9) +#define RT5663_EMB_CLK_SHIFT 9 +#define RT5663_EMB_CLK_EN (0x1 << 9) +#define RT5663_EMB_CLK_DIS (0x0 << 9) +#define RT5663_HPA_CPL_BIAS_MASK (0x7 << 6) +#define RT5663_HPA_CPL_BIAS_SHIFT 6 +#define RT5663_HPA_CPL_BIAS_0_5 (0x0 << 6) +#define RT5663_HPA_CPL_BIAS_1 (0x1 << 6) +#define RT5663_HPA_CPL_BIAS_2 (0x2 << 6) +#define RT5663_HPA_CPL_BIAS_3 (0x3 << 6) +#define RT5663_HPA_CPL_BIAS_4_1 (0x4 << 6) +#define RT5663_HPA_CPL_BIAS_4_2 (0x5 << 6) +#define RT5663_HPA_CPL_BIAS_6 (0x6 << 6) +#define RT5663_HPA_CPL_BIAS_8 (0x7 << 6) +#define RT5663_HPA_CPR_BIAS_MASK (0x7 << 3) +#define RT5663_HPA_CPR_BIAS_SHIFT 3 +#define RT5663_HPA_CPR_BIAS_0_5 (0x0 << 3) +#define RT5663_HPA_CPR_BIAS_1 (0x1 << 3) +#define RT5663_HPA_CPR_BIAS_2 (0x2 << 3) +#define RT5663_HPA_CPR_BIAS_3 (0x3 << 3) +#define RT5663_HPA_CPR_BIAS_4_1 (0x4 << 3) +#define RT5663_HPA_CPR_BIAS_4_2 (0x5 << 3) +#define RT5663_HPA_CPR_BIAS_6 (0x6 << 3) +#define RT5663_HPA_CPR_BIAS_8 (0x7 << 3) +#define RT5663_DUMMY_BIAS_MASK (0x7) +#define RT5663_DUMMY_BIAS_SHIFT 0 +#define RT5663_DUMMY_BIAS_0_5 (0x0) +#define RT5663_DUMMY_BIAS_1 (0x1) +#define RT5663_DUMMY_BIAS_2 (0x2) +#define RT5663_DUMMY_BIAS_3 (0x3) +#define RT5663_DUMMY_BIAS_4_1 (0x4) +#define RT5663_DUMMY_BIAS_4_2 (0x5) +#define RT5663_DUMMY_BIAS_6 (0x6) +#define RT5663_DUMMY_BIAS_8 (0x7) + + +/* System Clock Source */ +enum { + RT5668_SCLK_S_MCLK, + RT5668_SCLK_S_PLL1, + RT5668_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5668_PLL1_S_MCLK, + RT5668_PLL1_S_BCLK1, +}; + +enum { + RT5668_AIF, + RT5668_AIFS, +}; + +/* asrc clock source */ +enum { + RT5668_CLK_SEL_SYS = 0x0, + RT5668_CLK_SEL_I2S1_ASRC = 0x1, +}; + +/* filter mask */ +enum { + RT5668_DA_STEREO_FILTER = 0x1, + RT5668_AD_STEREO_FILTER = 0x2, +}; + +int rt5668_headset_detect(struct snd_soc_codec *codec, int jack_insert); +int rt5668_button_detect(struct snd_soc_codec *codec); +int rt5668_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack); +int rt5668_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5668_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3 0/3] ASoC: Intel: Skylake: Add HDA link management
by Vinod Koul 13 May '16

13 May '16

HDA links are always kept On, this patch adds link management routines in hdac extended core and it's use in Skylake driver and hdmi codec (runs on HDA link) The aim of serial is to shut down the links when not in use and keep them active when a user is active. The user can activate link by taking reference to the link when required. When all links are closed, the command DMAs are turned off as well. On first link power up, the command DMAs are turned on first. First patch touches HDA core, and rest two are dependent on it so Takashi if you are okay, pls ACK it and we can merge thru ASoC tree Changes in v3: - rename the varaible for dmas and make it bool - change the logic of checking links for powering off dmas as suggested by takashi Changes in v2: - use mutex instead of spinlock Vinod Koul (3): ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link mangement ASoC: hdac_hdmi: add link management include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 66 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_hdmi.c | 32 ++++++++++++++++-- sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/intel/skylake/skl.c | 34 +++++++++++++++++++ 6 files changed, 146 insertions(+), 3 deletions(-) -- 1.9.1

3 7

[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Skylake: Add HDA link management
by Vinod Koul 13 May '16

13 May '16

HDA links are always kept On, this patch adds link management routines in hdac extended core and it's use in Skylake driver and hdmi codec (runs on HDA link) The aim of serial is to shut down the links when not in use and keep them active when a user is active. The user can activate link by taking reference to the link when required. When all links are closed, the command DMAs are turned off as well. On first link power up, the command DMAs are turned on first. First patch touches HDA core, and rest two are dependent on it so Takashi if you are okay, pls ACK it and we can merge thru ASoC tree Changes in v2: - use mutex instead of spinlock Vinod Koul (3): ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link management ASoC: hdac_hdmi: add link management include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_hdmi.c | 32 +++++++++++++++++-- sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/intel/skylake/skl.c | 34 ++++++++++++++++++++ 6 files changed, 142 insertions(+), 3 deletions(-) -- 1.9.1

3 9

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Properly implement the positive and negative pins into the mixers
by Jeremy McDermond 13 May '16

13 May '16

The TLV320AIC32x4 has a very flexible mixer on the inputs to the ADCs. Each mixer has an available set of available pins that can be connected to the ADC positive and negative pins via three different resistor values. This allows for configuration of differential inputs as well as doing level manipulation between sources going into the mixers. The current code only provides positive pins and I implemented the resistors in an earlier patch. It turns out that it appears to more accurately model what's happening to implement each of the pins as a MUX rather than on/off switches and a mixer. This way each pin can be set to its desired resistor value. Since there are no switches, the mixer is no longer necessary in the DAPM path. I set the DAPM paths such that the "off" position of any of the MUXes turns the path off. This should allow for any input confiuration available on the codec. Signed-off-by: Jeremy McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 210 ++++++++++++++++++++++++++++++--------- 1 file changed, 161 insertions(+), 49 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 0eb8acc..85d4978 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -187,36 +187,67 @@ static const char * const resistor_text[] = { "Off", "10 kOhm", "20 kOhm", "40 kOhm", }; -static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, - resistor_text); - -static const struct snd_kcontrol_new left_input_mixer_controls[] = { - SOC_DAPM_ENUM("IN1_L P Switch", in1l_lpga_p_enum), - SOC_DAPM_ENUM("IN2_L P Switch", in2l_lpga_p_enum), - SOC_DAPM_ENUM("IN3_L P Switch", in3l_lpga_p_enum), - SOC_DAPM_ENUM("IN1_R P Switch", in1r_lpga_p_enum), +/* Left mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text); + +static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum), +}; +static const struct snd_kcontrol_new cml_to_lmixer_controls[] = { + SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum), +}; +static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum), +}; +static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum), }; -static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, - resistor_text); - -static const struct snd_kcontrol_new right_input_mixer_controls[] = { - SOC_DAPM_ENUM("IN1_R P Switch", in1r_rpga_p_enum), - SOC_DAPM_ENUM("IN2_R P Switch", in2r_rpga_p_enum), - SOC_DAPM_ENUM("IN3_R P Switch", in3r_rpga_p_enum), - SOC_DAPM_ENUM("IN2_L P Switch", in2l_rpga_p_enum), +/* Right mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text); +static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum), +}; +static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = { + SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum), +}; +static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum), +}; +static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -240,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { &lor_output_mixer_controls[0], ARRAY_SIZE(lor_output_mixer_controls)), SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, - &left_input_mixer_controls[0], - ARRAY_SIZE(left_input_mixer_controls)), - SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, - &right_input_mixer_controls[0], - ARRAY_SIZE(right_input_mixer_controls)), - SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_rmixer_controls), + SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cmr_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_rmixer_controls), + + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_lmixer_controls), + SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cml_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_lmixer_controls), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), SND_SOC_DAPM_OUTPUT("HPL"), @@ -287,21 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"LOR Power", NULL, "LOR Output Mixer"}, {"LOR", NULL, "LOR Power"}, - /* Left input */ - {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, - {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, - {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, - {"Left Input Mixer", "IN1_R P Switch", "IN1_R"}, - - {"Left ADC", NULL, "Left Input Mixer"}, - /* Right Input */ - {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, - {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, - {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, - {"Right Input Mixer", "IN2_L P Switch", "IN2_L"}, - - {"Right ADC", NULL, "Right Input Mixer"}, + {"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"}, + {"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"}, + {"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"}, + + {"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"}, + {"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"}, + + {"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"}, + {"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"}, + {"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"}, + + {"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"}, + {"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"}, + + {"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"}, + {"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"}, + + /* Left Input */ + {"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"}, + {"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"}, + + {"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"}, + {"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"}, + {"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"}, + + {"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"}, + {"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"}, + {"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"}, + + {"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"}, + {"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"}, + + {"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"}, + {"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"}, }; static const struct regmap_range_cfg aic32x4_regmap_pages[] = { -- 2.5.0

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. ;) Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 63 insertions(+), 11 deletions(-) -- 1.9.1

5 11

[alsa-devel] [PATCH 0/5] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (5): ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 69 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 60 insertions(+), 9 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v4 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

Version 4: - rebase on top of v4.6-rcX + alsa-broonie/for-next. - The only change with v3 concerns the patch 6/6, since CCSR_SSI_SOR is now already included in the list of volatile registers. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped at Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 80 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 69 insertions(+), 11 deletions(-) -- 1.9.1

6 23

[alsa-devel] [PATCH v3 0/7] ARM/ASoC: OMAP3: Fix McBSP2/3 sidetone support
by Peter Ujfalusi 13 May '16

13 May '16

Hi, Based on the ongoing discussion on v2 (ARM: OMAP3: Fix McBSP2/3 hwmod setup for sidetone) I have dropped the removal of the sidetone hwmod and only corrected it. The series now includes all related changes needed to have correct sidetone support whenever we boot in legacy or in DT mode. This time the ASoC part have dependency on earlier patches so they can not be applied separately. Regards, Peter --- Peter Ujfalusi (7): ARM: OMAP3: hwmod data: Fix McBSP2/3 sidetone data ARM: dts: omap3: Add clocks to McBSP nodes ARM: OMAP3: McBSP: New callback for McBSP2/3 ICLK idle configuration ARM: OMAP3: pdata-quirks: Add support for McBSP2/3 sidetone handling ASoc: omap-mcbsp: Rename omap_mcbsp_sysfs_remove() to omap_mcbsp_cleanup() ASoC: omap-mcbsp: sidetone: Use the new callback for iclk handling ARM: OMAP2+: McBSP: Remove the old iclk allow/deny idle code arch/arm/boot/dts/omap3.dtsi | 10 ++++++++++ arch/arm/mach-omap2/mcbsp.c | 31 ++++++++++++++++------------- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 24 ++++------------------ arch/arm/mach-omap2/pdata-quirks.c | 18 +++++++++++++++++ include/linux/platform_data/asoc-ti-mcbsp.h | 4 +++- sound/soc/omap/mcbsp.c | 21 +++++++++++++------ sound/soc/omap/mcbsp.h | 3 ++- sound/soc/omap/omap-mcbsp.c | 5 ++++- 8 files changed, 73 insertions(+), 43 deletions(-) -- 2.8.1

3 13

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: note buggy I2S support when the codec masters LRCLK
by Peter Rosin 13 May '16

13 May '16

The only difference in register settings between I2S and DSP mode A, for the SND_SOC_DAIFMT_CBM_CFM case, is the start condition (which is SSC_START_FALLING_RF for I2S and SSC_START_RISING_RF for DSP mode A). As this is not the only difference between the two formats, it is an indicator that something is seriously wrong. And it is the I2S support that is broken. While the start condition is correct for the left channel word in the I2S case, it is not correct that the right channel word follows immediately after the left channel word. The start of the right channel word should be triggered by a rising edge on LRCLK in the I2S case, something which simply does not happen. The correct thing to do would be to simply remove the alleged support for I2S/CBM_CFM, but that is likely to cause regressions. So, just make a note about the buggy I2S support in the source. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 4ab1e2013238..3108c6e20ad5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -558,7 +558,15 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* I2S format, CODEC supplies BCLK and LRC clocks. */ + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * NOTE! This is not really I2S, as the SSC does not allow + * for any extra empty BCLK cycles after the left channel + * word. I.e. the SSC is only looking at the falling edge + * of LRCLK and ignores the rising edge. + * In reality, this is DSP mode A with inverted LRCLK. + * Use DSP mode A instead, if your codec supports it. + */ rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_FALLING_RF) -- 2.1.4

2 3

[alsa-devel] [PATCH 0/9] ASoC: es8328 codec improvements
by John Keeping 13 May '16

13 May '16

The first 4 patches are bug fixes, then the next 4 add support for more sample rates and formats. The final patch I'm slightly unsure about since the hardware does allow different playback and capture rates, but it only allows 8kHz and 48kHz or 8.0182kHz and 44.1kHz, all other rates have to be symmetric. All of these patches have been tested on a Radxa Rock2 Square with rockchip-i2s. John Keeping (9): ASoC: es8328: Move clock setup to hw_params ASoC: es8328: Fix ADC format setup ASoC: es8328: Fix mask for VMIDSEL ASoC: es8328: Use single R/W for regmap ASoC: es8328: Use more suitable definition for mic bias ASoC: es8328: Move sample size setup to hw_params ASoC: es8328: Support more sample formats ASoC: es8328: Support more sample rates ASoC: es8328: Set symmetric rates sound/soc/codecs/es8328.c | 198 +++++++++++++++++++++++++++++++++++----------- sound/soc/codecs/es8328.h | 23 ++++-- 2 files changed, 168 insertions(+), 53 deletions(-) -- 2.8.0.rc4.238.g874082a

2 22

[alsa-devel] [PATCH 1/2] ASoC: rt298: fix null deref on acpi driver data
by Vinod Koul 13 May '16

13 May '16

ACPI driver data can be NULL so we need to check that before dereference the driver data. Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 68cf8d5a174f..45dc5fa24961 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1179,7 +1179,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c, /* enable jack combo mode on supported devices */ acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); - if (acpiid) { + if (acpiid && acpiid->driver_data) { rt298->pdata = *(struct rt298_platform_data *) acpiid->driver_data; } -- 1.9.1

3 5

[alsa-devel] [PATCH] rcar: src: skip disabled-SRC nodes
by Sergei Shtylyov 13 May '16

13 May '16

The current device tree representation of the R-Car Sample Rate Converters (SRC) assumes that they are numbered consecutively, starting from 0. Alas, this is not the case with the R8A7794 SoC where SRC0 isn't present. In order to keep the existing device trees working, I'm suggesting to use a disabled node for SRC0. Teach the SRC probe to just skip disabled nodes. Signed-off-by: Sergei Shtylyov <sergei.shtylyov(a)cogentembedded.com> --- The patch is against the 'for-next' branch of Mark Brown's 'sound.git' repo. sound/soc/sh/rcar/src.c | 4 ++++ sound/soc/sh/rcar/src.c | 4 ++++ 1 file changed, 4 insertions(+) Index: sound/sound/soc/sh/rcar/src.c =================================================================== --- sound.orig/sound/soc/sh/rcar/src.c +++ sound/sound/soc/sh/rcar/src.c @@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *pri i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + src = rsnd_src_get(priv, i); snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d", @@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *pri if (ret) goto rsnd_src_probe_done; +skip: i++; }

3 3

[alsa-devel] [PATCH] ASoC: rsnd: count .probe/.remove for rsnd_mod_call()
by Kuninori Morimoto 13 May '16

13 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_mod_call is counting its calling count to avoid unbalanced function pair calling for error cases (ex init <-> quit). SSI parent is now controlled as "mod" on current rsnd driver. Because of this reason, SSI .remove function will be called twice if it was used as SSI parent when user tried unbind. But probe/remove pair were not counted. This patch counts probe/remove functions to avoid it. Special thans Hiep Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsnd.h | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc89a67..a8f61d7 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -276,8 +276,9 @@ struct rsnd_mod { /* * status * - * 0xH0000CB0 + * 0xH0000CBA * + * A 0: probe 1: remove * B 0: init 1: quit * C 0: start 1: stop * @@ -287,19 +288,19 @@ struct rsnd_mod { * H 0: fallback * H 0: hw_params */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 #define __rsnd_mod_shift_stop 8 -#define __rsnd_mod_shift_probe 28 /* always called */ -#define __rsnd_mod_shift_remove 28 /* always called */ #define __rsnd_mod_shift_irq 28 /* always called */ #define __rsnd_mod_shift_pcm_new 28 /* always called */ #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ -#define __rsnd_mod_add_probe 0 -#define __rsnd_mod_add_remove 0 +#define __rsnd_mod_add_probe 1 +#define __rsnd_mod_add_remove -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -310,7 +311,7 @@ struct rsnd_mod { #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_call_probe 0 -#define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_remove 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 1.9.1

3 3

[alsa-devel] [PATCH v2] ASoC: MAX9860: new driver
by Peter Rosin 12 May '16

12 May '16

This is a driver for the MAX9860 Mono Audio Voice Codec. https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf This driver does not support sidetone since the DVST register field is backwards with the mute near the maximum level instead of the minimum. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9860.c | 761 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9860.h | 162 ++++++++++ 5 files changed, 938 insertions(+) create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h I have yet to find a good spot to add scanning for a mute TLV value needed for the backwards sidetone register field (DVST). Hence, still no sidetone support... New in v2: - Add comment about fall through when Integer Clock Mode is not possible. - Drop export of max9860_probe. - Ignore clk docs and read the mclk rate w/o enabling the clock. - Manage the DVDDIO supply. Cheers, Peter diff --git a/MAINTAINERS b/MAINTAINERS index a727d9959ecd..0803396b10f4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7000,6 +7000,13 @@ F: Documentation/devicetree/bindings/i2c/max6697.txt F: drivers/hwmon/max6697.c F: include/linux/platform_data/max6697.h +MAX9860 MONO AUDIO VOICE CODEC DRIVER +M: Peter Rosin <peda(a)axentia.se> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/max9860.txt +F: sound/soc/codecs/max9860* + MAXIM MUIC CHARGER DRIVERS FOR EXYNOS BASED BOARDS M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> L: linux-pm(a)vger.kernel.org diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c16478..241a23c9aa9f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C + select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX @@ -534,6 +535,11 @@ config SND_SOC_MAX98926 config SND_SOC_MAX9850 tristate +config SND_SOC_MAX9860 + tristate "Maxim MAX9860 Mono Audio Voice Codec" + depends on I2C + select REGMAP_I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..e435f4df1787 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -78,6 +78,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o +snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o @@ -287,6 +288,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c new file mode 100644 index 000000000000..ee30832b0afb --- /dev/null +++ b/sound/soc/codecs/max9860.c @@ -0,0 +1,761 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf + * + * The driver does not support sidetone since the DVST register field is + * backwards with the mute near the maximum level instead of the minimum. + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/regulator/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "max9860.h" + +struct max9860_priv { + struct regmap *regmap; + struct regulator *dvddio; + struct notifier_block dvddio_nb; + u8 psclk; + unsigned long pclk_rate; + int fmt; +}; + +static int max9860_dvddio_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct max9860_priv *max9860 = container_of(nb, struct max9860_priv, + dvddio_nb); + if (event & REGULATOR_EVENT_DISABLE) { + regcache_mark_dirty(max9860->regmap); + regcache_cache_only(max9860->regmap, true); + } + + return 0; +} + +static const struct reg_default max9860_reg_defaults[] = { + { MAX9860_PWRMAN, 0x00 }, + { MAX9860_INTEN, 0x00 }, + { MAX9860_SYSCLK, 0x00 }, + { MAX9860_AUDIOCLKHIGH, 0x00 }, + { MAX9860_AUDIOCLKLOW, 0x00 }, + { MAX9860_IFC1A, 0x00 }, + { MAX9860_IFC1B, 0x00 }, + { MAX9860_VOICEFLTR, 0x00 }, + { MAX9860_DACATTN, 0x00 }, + { MAX9860_ADCLEVEL, 0x00 }, + { MAX9860_DACGAIN, 0x00 }, + { MAX9860_MICGAIN, 0x00 }, + { MAX9860_MICADC, 0x00 }, + { MAX9860_NOISEGATE, 0x00 }, +}; + +static bool max9860_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + case MAX9860_REVISION: + return true; + } + + return false; +} + +static bool max9860_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTEN ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + return true; + } + + return false; +} + +static bool max9860_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + case MAX9860_MICREADBACK: + return true; + } + + return false; +} + +static bool max9860_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + return true; + } + + return false; +} + +const struct regmap_config max9860_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max9860_readable, + .writeable_reg = max9860_writeable, + .volatile_reg = max9860_volatile, + .precious_reg = max9860_precious, + + .max_register = MAX9860_MAX_REGISTER, + .reg_defaults = max9860_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(max9860_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_RANGE(pam_tlv, + 0, MAX9860_PAM_MAX - 1, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + MAX9860_PAM_MAX, MAX9860_PAM_MAX, TLV_DB_SCALE_ITEM(3000, 0, 0)); +static const DECLARE_TLV_DB_SCALE(pgam_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(anth_tlv, -7600, 400, 1); +static const DECLARE_TLV_DB_SCALE(agcth_tlv, -1800, 100, 0); + +static const char * const agchld_text[] = { + "AGC Disabled", "50ms", "100ms", "400ms" +}; + +static SOC_ENUM_SINGLE_DECL(agchld_enum, MAX9860_MICADC, + MAX9860_AGCHLD_SHIFT, agchld_text); + +static const char * const agcsrc_text[] = { + "Left ADC", "Left/Right ADC" +}; + +static SOC_ENUM_SINGLE_DECL(agcsrc_enum, MAX9860_MICADC, + MAX9860_AGCSRC_SHIFT, agcsrc_text); + +static const char * const agcatk_text[] = { + "3ms", "12ms", "50ms", "200ms" +}; + +static SOC_ENUM_SINGLE_DECL(agcatk_enum, MAX9860_MICADC, + MAX9860_AGCATK_SHIFT, agcatk_text); + +static const char * const agcrls_text[] = { + "78ms", "156ms", "312ms", "625ms", + "1.25s", "2.5s", "5s", "10s" +}; + +static SOC_ENUM_SINGLE_DECL(agcrls_enum, MAX9860_MICADC, + MAX9860_AGCRLS_SHIFT, agcrls_text); + +static const char * const filter_text[] = { + "Disabled", + "Elliptical HP 217Hz notch (16kHz)", + "Butterworth HP 500Hz (16kHz)", + "Elliptical HP 217Hz notch (8kHz)", + "Butterworth HP 500Hz (8kHz)", + "Butterworth HP 200Hz (48kHz)" +}; + +static SOC_ENUM_SINGLE_DECL(avflt_enum, MAX9860_VOICEFLTR, + MAX9860_AVFLT_SHIFT, filter_text); + +static SOC_ENUM_SINGLE_DECL(dvflt_enum, MAX9860_VOICEFLTR, + MAX9860_DVFLT_SHIFT, filter_text); + +static const struct snd_kcontrol_new max9860_controls[] = { +SOC_SINGLE_TLV("Master Playback Volume", MAX9860_DACATTN, + MAX9860_DVA_SHIFT, MAX9860_DVA_MUTE, 1, dva_tlv), +SOC_SINGLE_TLV("DAC Gain Volume", MAX9860_DACGAIN, + MAX9860_DVG_SHIFT, MAX9860_DVG_MAX, 0, dvg_tlv), +SOC_DOUBLE_TLV("Line Capture Volume", MAX9860_ADCLEVEL, + MAX9860_ADCLL_SHIFT, MAX9860_ADCRL_SHIFT, MAX9860_ADCxL_MIN, 1, + adc_tlv), + +SOC_ENUM("AGC Hold Time", agchld_enum), +SOC_ENUM("AGC/Noise Gate Source", agcsrc_enum), +SOC_ENUM("AGC Attack Time", agcatk_enum), +SOC_ENUM("AGC Release Time", agcrls_enum), + +SOC_SINGLE_TLV("Noise Gate Threshold Volume", MAX9860_NOISEGATE, + MAX9860_ANTH_SHIFT, MAX9860_ANTH_MAX, 0, anth_tlv), +SOC_SINGLE_TLV("AGC Signal Threshold Volume", MAX9860_NOISEGATE, + MAX9860_AGCTH_SHIFT, MAX9860_AGCTH_MIN, 1, agcth_tlv), + +SOC_SINGLE_TLV("Mic PGA Volume", MAX9860_MICGAIN, + MAX9860_PGAM_SHIFT, MAX9860_PGAM_MIN, 1, pgam_tlv), +SOC_SINGLE_TLV("Mic Preamp Volume", MAX9860_MICGAIN, + MAX9860_PAM_SHIFT, MAX9860_PAM_MAX, 0, pam_tlv), + +SOC_ENUM("ADC Filter", avflt_enum), +SOC_ENUM("DAC Filter", dvflt_enum), +}; + +static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICL"), +SND_SOC_DAPM_INPUT("MICR"), + +SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), + +SND_SOC_DAPM_OUTPUT("OUT"), + +SND_SOC_DAPM_SUPPLY("Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DVDD", 0, 0), +SND_SOC_DAPM_CLOCK_SUPPLY("mclk"), +}; + +static const struct snd_soc_dapm_route max9860_dapm_routes[] = { + { "ADCL", NULL, "MICL" }, + { "ADCR", NULL, "MICR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, + + { "DAC", NULL, "AIFINL" }, + { "DAC", NULL, "AIFINR" }, + { "OUT", NULL, "DAC" }, + + { "Supply", NULL, "AVDD" }, + { "Supply", NULL, "DVDD" }, + { "Supply", NULL, "mclk" }, + + { "DAC", NULL, "Supply" }, + { "ADCL", NULL, "Supply" }, + { "ADCR", NULL, "Supply" }, +}; + +static int max9860_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + return 0; + + default: + return -EINVAL; + } +} + +static int max9860_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 max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + u8 master; + u8 ifc1a = 0; + u8 ifc1b = 0; + u8 sysclk = 0; + unsigned long n; + int ret; + + dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n", + params_rate(params), + params_channels(params)); + + if (params_channels(params) == 2) + ifc1b |= MAX9860_ST; + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + master = MAX9860_MASTER; + break; + default: + return -EINVAL; + } + ifc1a |= master; + + if (master) { + if (params_width(params) * params_channels(params) > 48) + ifc1b |= MAX9860_BSEL_64X; + else + ifc1b |= MAX9860_BSEL_48X; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ifc1a |= MAX9860_DDLY; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + ifc1a |= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_DSP_A: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_A works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_DDLY | MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_DSP_B: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_B works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + break; + default: + return -EINVAL; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_IB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + /* fall through */ + case SND_SOC_DAIFMT_IB_NF: + ifc1a ^= MAX9860_DBCI; + ifc1b ^= MAX9860_ABCI; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "IFC1A %02x\n", ifc1a); + ret = regmap_write(max9860->regmap, MAX9860_IFC1A, ifc1a); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1A: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "IFC1B %02x\n", ifc1b); + ret = regmap_write(max9860->regmap, MAX9860_IFC1B, ifc1b); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1B: %d\n", ret); + return ret; + } + + /* + * Check if Integer Clock Mode is possible, but avoid it in slave mode + * since we then do not know if lrclk is derived from pclk and the + * datasheet mentions that the frequencies have to match exactly in + * order for this to work. + */ + if (params_rate(params) == 8000 || params_rate(params) == 16000) { + if (master) { + switch (max9860->pclk_rate) { + case 12000000: + sysclk = MAX9860_FREQ_12MHZ; + break; + case 13000000: + sysclk = MAX9860_FREQ_13MHZ; + break; + case 19200000: + sysclk = MAX9860_FREQ_19_2MHZ; + break; + default: + /* + * Integer Clock Mode not possible. Leave + * sysclk at zero and fall through to the + * code below for PLL mode. + */ + break; + } + + if (sysclk && params_rate(params) == 16000) + sysclk |= MAX9860_16KHZ; + } + } + + /* + * Largest possible n: + * 65536 * 96 * 48kHz / 10MHz -> 30199 + * Smallest possible n: + * 65536 * 96 * 8kHz / 20MHz -> 2517 + * Both fit nicely in the available 15 bits, no need to apply any mask. + */ + n = DIV_ROUND_CLOSEST_ULL(65536ULL * 96 * params_rate(params), + max9860->pclk_rate); + + if (!sysclk) { + /* PLL mode */ + if (params_rate(params) > 24000) + sysclk |= MAX9860_16KHZ; + + if (!master) + n |= 1; /* trigger rapid pll lock mode */ + } + + sysclk |= max9860->psclk; + dev_dbg(codec->dev, "SYSCLK %02x\n", sysclk); + ret = regmap_write(max9860->regmap, + MAX9860_SYSCLK, sysclk); + if (ret) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "N %lu\n", n); + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKHIGH, n >> 8); + if (ret) { + dev_err(codec->dev, "Failed to set NHI: %d\n", ret); + return ret; + } + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKLOW, n & 0xff); + if (ret) { + dev_err(codec->dev, "Failed to set NLO: %d\n", ret); + return ret; + } + + if (!master) { + dev_dbg(codec->dev, "Enable PLL\n"); + ret = regmap_update_bits(max9860->regmap, MAX9860_AUDIOCLKHIGH, + MAX9860_PLL, MAX9860_PLL); + if (ret) { + dev_err(codec->dev, "Failed to enable PLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + max9860->fmt = fmt; + return 0; +} + +static const struct snd_soc_dai_ops max9860_dai_ops = { + .startup = max9860_dai_startup, + .hw_params = max9860_hw_params, + .set_fmt = max9860_set_fmt, +}; + +static struct snd_soc_dai_driver max9860_dai = { + .name = "max9860-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &max9860_dai_ops, + .symmetric_rates = 1, +}; + +static int max9860_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max9860_priv *max9860 = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, MAX9860_SHDN); + if (ret) { + dev_err(codec->dev, "Failed to remove SHDN: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, 0); + if (ret) { + dev_err(codec->dev, "Failed to request SHDN: %d\n", + ret); + return ret; + } + break; + } + + return 0; +} + +static struct snd_soc_codec_driver max9860_codec_driver = { + .set_bias_level = max9860_set_bias_level, + .idle_bias_off = true, + + .controls = max9860_controls, + .num_controls = ARRAY_SIZE(max9860_controls), + .dapm_widgets = max9860_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9860_dapm_widgets), + .dapm_routes = max9860_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes), +}; + +#ifdef CONFIG_PM +static int max9860_suspend(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, MAX9860_PSCLK_OFF); + if (ret) { + dev_err(dev, "Failed to disable clock: %d\n", ret); + return ret; + } + + regulator_disable(max9860->dvddio); + + return 0; +} + +static int max9860_resume(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regulator_enable(max9860->dvddio); + if (ret) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + regcache_cache_only(max9860->regmap, false); + ret = regcache_sync(max9860->regmap); + if (ret) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, max9860->psclk); + if (ret) { + dev_err(dev, "Failed to enable clock: %d\n", ret); + return ret; + } + + return 0; +} +#endif + +const struct dev_pm_ops max9860_pm_ops = { + SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) +}; + +static int max9860_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860; + int ret; + struct clk *mclk; + unsigned long mclk_rate; + int i; + int intr; + + max9860 = devm_kzalloc(dev, sizeof(struct max9860_priv), GFP_KERNEL); + if (!max9860) + return -ENOMEM; + + max9860->dvddio = devm_regulator_get(dev, "DVDDIO"); + if (IS_ERR(max9860->dvddio)) { + ret = PTR_ERR(max9860->dvddio); + dev_err(dev, "Failed to get DVDDIO supply: %d\n", ret); + return ret; + } + + max9860->dvddio_nb.notifier_call = max9860_dvddio_event; + + ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb); + if (ret) + dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret); + + ret = regulator_enable(max9860->dvddio); + if (ret != 0) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + max9860->regmap = devm_regmap_init_i2c(i2c, &max9860_regmap); + if (IS_ERR(max9860->regmap)) { + ret = PTR_ERR(max9860->regmap); + goto err_regulator; + } + + dev_set_drvdata(dev, max9860); + + /* + * mclk has to be in the 10MHz to 60MHz range. + * psclk is used to scale mclk into pclk so that + * pclk is in the 10MHz to 20MHz range. + */ + mclk = clk_get(dev, "mclk"); + + if (IS_ERR(mclk)) { + ret = PTR_ERR(mclk); + dev_err(dev, "Failed to get MCLK: %d\n", ret); + goto err_regulator; + } + + mclk_rate = clk_get_rate(mclk); + clk_put(mclk); + + if (mclk_rate > 60000000 || mclk_rate < 10000000) { + dev_err(dev, "Bad mclk %luHz (needs 10MHz - 60MHz)\n", + mclk_rate); + ret = -EINVAL; + goto err_regulator; + } + if (mclk_rate >= 40000000) + max9860->psclk = 3; + else if (mclk_rate >= 20000000) + max9860->psclk = 2; + else + max9860->psclk = 1; + max9860->pclk_rate = mclk_rate >> (max9860->psclk - 1); + max9860->psclk <<= MAX9860_PSCLK_SHIFT; + dev_dbg(dev, "mclk %lu pclk %lu\n", mclk_rate, max9860->pclk_rate); + + regcache_cache_bypass(max9860->regmap, true); + for (i = 0; i < max9860_regmap.num_reg_defaults; ++i) { + ret = regmap_write(max9860->regmap, + max9860_regmap.reg_defaults[i].reg, + max9860_regmap.reg_defaults[i].def); + if (ret) { + dev_err(dev, "Failed to initialize register %u: %d\n", + max9860_regmap.reg_defaults[i].reg, ret); + goto err_regulator; + } + } + regcache_cache_bypass(max9860->regmap, false); + + ret = regmap_read(max9860->regmap, MAX9860_INTRSTATUS, &intr); + if (ret) { + dev_err(dev, "Failed to clear INTRSTATUS: %d\n", ret); + goto err_regulator; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &max9860_codec_driver, + &max9860_dai, 1); + if (ret) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_regulator: + regulator_disable(max9860->dvddio); + return ret; +} + +static int max9860_remove(struct i2c_client *i2c) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860 = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + regulator_disable(max9860->dvddio); + return 0; +} + +static const struct i2c_device_id max9860_i2c_id[] = { + { "max9860", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9860_i2c_id); + +static const struct of_device_id max9860_of_match[] = { + { .compatible = "maxim,max9860", }, + { } +}; +MODULE_DEVICE_TABLE(of, max9860_of_match); + +static struct i2c_driver max9860_i2c_driver = { + .probe = max9860_probe, + .remove = max9860_remove, + .id_table = max9860_i2c_id, + .driver = { + .name = "max9860", + .of_match_table = max9860_of_match, + .pm = &max9860_pm_ops, + }, +}; + +module_i2c_driver(max9860_i2c_driver); + +MODULE_DESCRIPTION("ASoC MAX9860 Mono Audio Voice Codec driver"); +MODULE_AUTHOR("Peter Rosin <peda(a)axentia.se>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/max9860.h b/sound/soc/codecs/max9860.h new file mode 100644 index 000000000000..22041bd67a7d --- /dev/null +++ b/sound/soc/codecs/max9860.h @@ -0,0 +1,162 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef _SND_SOC_MAX9860 +#define _SND_SOC_MAX9860 + +#define MAX9860_INTRSTATUS 0x00 +#define MAX9860_MICREADBACK 0x01 +#define MAX9860_INTEN 0x02 +#define MAX9860_SYSCLK 0x03 +#define MAX9860_AUDIOCLKHIGH 0x04 +#define MAX9860_AUDIOCLKLOW 0x05 +#define MAX9860_IFC1A 0x06 +#define MAX9860_IFC1B 0x07 +#define MAX9860_VOICEFLTR 0x08 +#define MAX9860_DACATTN 0x09 +#define MAX9860_ADCLEVEL 0x0a +#define MAX9860_DACGAIN 0x0b +#define MAX9860_MICGAIN 0x0c +#define MAX9860_RESERVED 0x0d +#define MAX9860_MICADC 0x0e +#define MAX9860_NOISEGATE 0x0f +#define MAX9860_PWRMAN 0x10 +#define MAX9860_REVISION 0xff + +#define MAX9860_MAX_REGISTER 0xff + +/* INTRSTATUS */ +#define MAX9860_CLD 0x80 +#define MAX9860_SLD 0x40 +#define MAX9860_ULK 0x20 + +/* MICREADBACK */ +#define MAX9860_NG 0xe0 +#define MAX9860_AGC 0x1f + +/* INTEN */ +#define MAX9860_ICLD 0x80 +#define MAX9860_ISLD 0x40 +#define MAX9860_IULK 0x20 + +/* SYSCLK */ +#define MAX9860_PSCLK 0x30 +#define MAX9860_PSCLK_OFF 0x00 +#define MAX9860_PSCLK_SHIFT 4 +#define MAX9860_FREQ 0x06 +#define MAX9860_FREQ_NORMAL 0x00 +#define MAX9860_FREQ_12MHZ 0x02 +#define MAX9860_FREQ_13MHZ 0x04 +#define MAX9860_FREQ_19_2MHZ 0x06 +#define MAX9860_16KHZ 0x01 + +/* AUDIOCLKHIGH */ +#define MAX9860_PLL 0x80 +#define MAX9860_NHI 0x7f + +/* AUDIOCLKLOW */ +#define MAX9860_NLO 0xff + +/* IFC1A */ +#define MAX9860_MASTER 0x80 +#define MAX9860_WCI 0x40 +#define MAX9860_DBCI 0x20 +#define MAX9860_DDLY 0x10 +#define MAX9860_HIZ 0x08 +#define MAX9860_TDM 0x04 + +/* IFC1B */ +#define MAX9860_ABCI 0x20 +#define MAX9860_ADLY 0x10 +#define MAX9860_ST 0x08 +#define MAX9860_BSEL 0x07 +#define MAX9860_BSEL_OFF 0x00 +#define MAX9860_BSEL_64X 0x01 +#define MAX9860_BSEL_48X 0x02 +#define MAX9860_BSEL_PCLK_2 0x04 +#define MAX9860_BSEL_PCLK_4 0x05 +#define MAX9860_BSEL_PCLK_8 0x06 +#define MAX9860_BSEL_PCLK_16 0x07 + +/* VOICEFLTR */ +#define MAX9860_AVFLT 0xf0 +#define MAX9860_AVFLT_SHIFT 4 +#define MAX9860_AVFLT_COUNT 6 +#define MAX9860_DVFLT 0x0f +#define MAX9860_DVFLT_SHIFT 0 +#define MAX9860_DVFLT_COUNT 6 + +/* DACATTN */ +#define MAX9860_DVA 0xfe +#define MAX9860_DVA_SHIFT 1 +#define MAX9860_DVA_MUTE 0x5e + +/* ADCLEVEL */ +#define MAX9860_ADCRL 0xf0 +#define MAX9860_ADCRL_SHIFT 4 +#define MAX9860_ADCLL 0x0f +#define MAX9860_ADCLL_SHIFT 0 +#define MAX9860_ADCxL_MIN 15 + +/* DACGAIN */ +#define MAX9860_DVG 0x60 +#define MAX9860_DVG_SHIFT 5 +#define MAX9860_DVG_MAX 3 +#define MAX9860_DVST 0x1f +#define MAX9860_DVST_SHIFT 0 +#define MAX9860_DVST_MIN 31 + +/* MICGAIN */ +#define MAX9860_PAM 0x60 +#define MAX9860_PAM_SHIFT 5 +#define MAX9860_PAM_MAX 3 +#define MAX9860_PGAM 0x1f +#define MAX9860_PGAM_SHIFT 0 +#define MAX9860_PGAM_MIN 20 + +/* MICADC */ +#define MAX9860_AGCSRC 0x80 +#define MAX9860_AGCSRC_SHIFT 7 +#define MAX9860_AGCSRC_COUNT 2 +#define MAX9860_AGCRLS 0x70 +#define MAX9860_AGCRLS_SHIFT 4 +#define MAX9860_AGCRLS_COUNT 8 +#define MAX9860_AGCATK 0x0c +#define MAX9860_AGCATK_SHIFT 2 +#define MAX9860_AGCATK_COUNT 4 +#define MAX9860_AGCHLD 0x03 +#define MAX9860_AGCHLD_OFF 0x00 +#define MAX9860_AGCHLD_SHIFT 0 +#define MAX9860_AGCHLD_COUNT 4 + +/* NOISEGATE */ +#define MAX9860_ANTH 0xf0 +#define MAX9860_ANTH_SHIFT 4 +#define MAX9860_ANTH_MAX 15 +#define MAX9860_AGCTH 0x0f +#define MAX9860_AGCTH_SHIFT 0 +#define MAX9860_AGCTH_MIN 15 + +/* PWRMAN */ +#define MAX9860_SHDN 0x80 +#define MAX9860_DACEN 0x08 +#define MAX9860_DACEN_SHIFT 3 +#define MAX9860_ADCLEN 0x02 +#define MAX9860_ADCLEN_SHIFT 1 +#define MAX9860_ADCREN 0x01 +#define MAX9860_ADCREN_SHIFT 0 + +#endif /* _SND_SOC_MAX9860 */ -- 2.1.4

1 1

[alsa-devel] [PATCH] ALSA: au88x0: Fix zero clear of stream->resources
by Takashi Iwai 12 May '16

12 May '16

There are a few calls of memset() to stream->resources, but they all are called in a wrong size, sizeof(unsigned char) * VORTEX_RESOURCE_LAST, while this field is a u32 array. This may leave the memories not zero-cleared. Fix it by replacing them with a simpler sizeof(stream->resources) instead. Reported-by: David Binderman <dcb314(a)hotmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/au88x0/au88x0_core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4667c3232b7f..4a054d720112 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2151,8 +2151,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if (stream->type != VORTEX_PCM_A3D) { @@ -2162,7 +2161,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXIN)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } @@ -2175,8 +2174,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_A3D)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); dev_err(vortex->card->dev, "out of A3D sources. Sorry\n"); return -EBUSY; @@ -2290,8 +2288,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXOUT)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if ((src[i] = @@ -2299,8 +2296,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } -- 2.8.2

1 0

[alsa-devel] sound/pci/au88x0/au88x0_core.c: many calls to memset with not-plausible size ?
by David Binderman 12 May '16

12 May '16

Hello there, sound/pci/au88x0/au88x0_core.c:2153:6: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] Source code is memset(stream->resources, 0, sizeof(unsigned char) * VORTEX_RESOURCE_LAST); but sound/pci/au88x0/au88x0.h: u32 resources[5]; Something like memset(stream->resources, 0, sizeof(stream->resources[0]) * VORTEX_RESOURCE_LAST); Some duplicates: sound/pci/au88x0/au88x0_core.c:2163:7: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2177:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2292:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2301:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] Regards David Binderman

2 2

[alsa-devel] [PATCH] ASoC: wm8985: rework and fix the clock calculation
by Petr Kulhavy 12 May '16

12 May '16

The clock calculation has several issues: 1) if PLL is used in master mode the BCLK output runs at double the speed 2) de-facto only 44.1kHz and 48kHz sampling rates are supported, other rates like 8kHz, 12kHz, 24kHz fail to find the proper BCLK divider 3) the wm8985->sysclk variable has a misleading name and is used wrongly in the clock calculation in wm8985_hw_params() which is the root cause for (1) 4) wm8985->bclk is used only in wm8985_hw_params() and therefore no need to store it in the wm8985_priv structure Therefore the clock calculation is rewritten in more clean and proper way: - move wm8985_priv->bclk as a local variable into mw8985_hw_params() - new variable wm8985_priv->pllout holds the actual frequency that is input to the MCLKDIV post-divider - move wm8985_priv->sysclk as a local variable into mw8985_hw_params() - sysclk is now always calculated as 256 * fs - the MCLKDIV is looked up as pllout/sysclk - fs_ratios[] is replaced by simpler mclk_divs[] lookup table With this patch all rates: 8, 11.025, 12, 16, 22.05, 24, 32, 44.1 and 48kHz work properly and generate the correct BCLK. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- sound/soc/codecs/wm8985.c | 59 ++++++++++++++++++++++++----------------------- 1 file changed, 30 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 18f2babe1090..628127aa3c96 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -181,22 +181,11 @@ static const int volume_update_regs[] = { struct wm8985_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8985_NUM_SUPPLIES]; - unsigned int sysclk; - unsigned int bclk; + unsigned int pllout; /* input rate to the MCLKDIV divider */ }; -static const struct { - int div; - int ratio; -} fs_ratios[] = { - { 10, 128 }, - { 15, 192 }, - { 20, 256 }, - { 30, 384 }, - { 40, 512 }, - { 60, 768 }, - { 80, 1024 }, - { 120, 1536 } +static const int mclk_divs[] = { + 10, 15, 20, 30, 40, 60, 80, 120 }; static const int srates[] = { 48000, 32000, 24000, 16000, 12000, 8000 }; @@ -693,15 +682,18 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec; struct wm8985_priv *wm8985; u16 blen, srate_idx; - unsigned int tmp; int srate_best; + unsigned int bclk; /* bit clock matching the current params */ + unsigned int sysclk; /* the actual sysclk after post division */ codec = dai->codec; wm8985 = snd_soc_codec_get_drvdata(codec); - wm8985->bclk = snd_soc_params_to_bclk(params); - if ((int)wm8985->bclk < 0) - return wm8985->bclk; + /* always use 256 * fs in order to get best filter quality */ + sysclk = 256 * params_rate(params); + bclk = snd_soc_params_to_bclk(params); + if ((int)bclk < 0) + return bclk; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -742,29 +734,28 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, WM8985_ADDITIONAL_CONTROL, WM8985_SR_MASK, srate_idx << WM8985_SR_SHIFT); - dev_dbg(dai->dev, "Target BCLK = %uHz\n", wm8985->bclk); - dev_dbg(dai->dev, "SYSCLK = %uHz\n", wm8985->sysclk); + dev_dbg(dai->dev, "Target BCLK = %uHz\n", bclk); + dev_dbg(dai->dev, "SYSCLK = %uHz\n", sysclk); - for (i = 0; i < ARRAY_SIZE(fs_ratios); ++i) { - if (wm8985->sysclk / params_rate(params) - == fs_ratios[i].ratio) + /* select the appropriate mclk divider */ + for (i = 0; i < ARRAY_SIZE(mclk_divs); ++i) { + if (wm8985->pllout / mclk_divs[i] * 10 + == sysclk) break; } - if (i == ARRAY_SIZE(fs_ratios)) { + if (i == ARRAY_SIZE(mclk_divs)) { dev_err(dai->dev, "Unable to configure MCLK ratio %u/%u\n", - wm8985->sysclk, params_rate(params)); + wm8985->pllout, sysclk); return -EINVAL; } - dev_dbg(dai->dev, "MCLK ratio = %dfs\n", fs_ratios[i].ratio); snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, WM8985_MCLKDIV_MASK, i << WM8985_MCLKDIV_SHIFT); /* select the appropriate bclk divider */ - tmp = (wm8985->sysclk / fs_ratios[i].div) * 10; for (i = 0; i < ARRAY_SIZE(bclk_divs); ++i) { - if (wm8985->bclk == tmp / bclk_divs[i]) + if (bclk == sysclk / bclk_divs[i]) break; } @@ -786,6 +777,10 @@ struct pll_div { }; #define FIXED_PLL_SIZE ((1ULL << 24) * 10) +/* + * source = MCLK input of the chip + * target = the f2 coming out of the PLL before /4 divider + */ static int pll_factors(struct pll_div *pll_div, unsigned int target, unsigned int source) { @@ -872,17 +867,23 @@ static int wm8985_set_sysclk(struct snd_soc_dai *dai, WM8985_CLKSEL_MASK, 0); snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, WM8985_PLLEN_MASK, 0); + wm8985->pllout = freq; break; case WM8985_CLKSRC_PLL: snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, WM8985_CLKSEL_MASK, WM8985_CLKSEL); + /* + * in order to run the PLL within the recommended 90MHz + * operating range the wm8985_set_pll() configures the PLL + * to output double the required frequency + */ + wm8985->pllout = 2 * freq; break; default: dev_err(dai->dev, "Unknown clock source %d\n", clk_id); return -EINVAL; } - wm8985->sysclk = freq; return 0; } -- 1.9.1

2 4

[alsa-devel] [PATCH 1/2] ASoC: nau8825: non-clock jack detection for power saving at standby
by John Hsu 12 May '16

12 May '16

The driver changes jack type detection interruption to non-clock archi- tecture for less 1mW power saving. The architecture is called manual mode jack detection. It has no hardware debounce, no jack type detection, but only detecting jack insertion. After jack insertion, the driver will switch to auto mode jack detection with internal clock which can detect microphone, jack type and do hardware debounce. The manual architecture has these main changes including codec initiation, interruption, clock control, and power management. When codec initiation or system resume, the clock is closed as jack insertion detection at man- ual mode, and bypass debounce circuit. These configurations move to resume setup function when setup bias level after resume. When jack insertion detection happens, the manual mode turns off and make configuration about jack type detection interruption at auto mode in auto irq setup function which can detect microphone and jack type. The inter- ruption will switch to manual mode again with clock free until jack ejec- tion happens. The system clock configuration adds clock disable option which can disable internal VCO clock. Before the system clock change, there is an restric- tion added to make sure clock disabled and not config any clock when no headset connected. In power management, we involve the solution about races and jack detec- tion in resume from Ben Zhang in the following patch and list his comment. [PATCH] ASoC: nau8825: Fix jack detection across suspend "Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power." "Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly." Change SAR widget to supply type which can prevent the codec keeping at SND_SOC_BIAS_ON during suspend. The codec suspend function can just invoke normally. Before the system suspends, the driver turns off all interruptions. Keep the interruption quiet before resume setup completes. The ADC channel will be disabled which is needed for interruptions at audo mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 281 +++++++++++++++++++++++++++++++++++---------- sound/soc/codecs/nau8825.h | 9 +- 2 files changed, 226 insertions(+), 64 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f35019a..9386624 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -30,10 +30,16 @@ #include "nau8825.h" + +#define NUVOTON_CODEC_DAI "nau8825-hifi" + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 +static int nau8825_configure_sysclk(struct nau8825 *nau8825, + int clk_id, unsigned int freq); + struct nau8825_fll { int mclk_src; int ratio; @@ -368,9 +374,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -614,8 +623,10 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. + */ return 0; } @@ -642,6 +653,22 @@ static void nau8825_restart_jack_detection(struct regmap *regmap) NAU8825_JACK_DET_RESTART, 0); } +static void nau8825_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8825_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + } +} + static void nau8825_eject_jack(struct nau8825 *nau8825) { struct snd_soc_dapm_context *dapm = nau8825->dapm; @@ -656,6 +683,67 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Mask all interruptions except jack insertion interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0xfffe); + + /* Enable insertion interruption only and bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); + + /* Close clock for jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); +} + +/* Enable audo mode interruptions with internal clock. */ +static void nau8825_setup_auto_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable headset jack type detection complete interruption and + * jack ejection interruption. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interruptions */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, 0); + + /* Unmask all interruptions */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + + /* Restart the jack detection process at auto mode */ + nau8825_restart_jack_detection(regmap); } static int nau8825_button_decode(int value) @@ -753,7 +841,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to clear interruption\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -789,6 +880,26 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == + NAU8825_JACK_INSERTION_DETECTED) { + /* One more step to check GPIO status directly. Thus, the + * driver can confirm the real insertion interruption because + * the intrruption at manual mode has bypassed debounce + * circuit which can get rid of unstable status. + */ + if (nau8825_is_jack_inserted(regmap)) { + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_INSERT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_INSERT_EN, NAU8825_IRQ_INSERT_EN); + /* Enable interruption for jack type detection at audo + * mode which can detect microphone and jack type. + */ + nau8825_setup_auto_irq(nau8825); + } } if (!clear_irq) @@ -938,8 +1049,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) } static const struct regmap_config nau8825_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = NAU8825_REG_DATA_LEN, + .reg_bits = NAU8825_REG_ADDR_LEN, .max_register = NAU8825_REG_MAX, .readable_reg = nau8825_readable_reg, @@ -958,11 +1069,10 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* Unmask interruptions. Handler uses dapm object so we can enable - * interruptions only after dapm is fully initialized. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. */ - regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); - nau8825_restart_jack_detection(nau8825->regmap); return 0; } @@ -1134,7 +1244,26 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, struct regmap *regmap = nau8825->regmap; int ret; + if (!nau8825_is_jack_inserted(nau8825->regmap) && + clk_id != NAU8825_CLK_DIS) { + /* For power saving less 1mW, the jack type detection inter- + * ruption changes to non-clock architecture. Therefore, the + * clock should be disabled and not allowed to config any clock + * when no headset connected. + */ + dev_warn(nau8825->dev, "Souldn't enable any clock when no headset connected\n"); + return 0; + } + switch (clk_id) { + case NAU8825_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, 0); + + break; case NAU8825_CLK_MCLK: regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); @@ -1209,6 +1338,30 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable insertion interruption only and bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, 0); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1238,11 +1391,22 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + + /* Setup codec configuration after resume */ + nau8825_resume_setup(nau8825); } break; case SND_SOC_BIAS_OFF: + /* Turn off all interruptions before system shutdown. Keep the + * interruption quiet before resume setup completes. + */ + regmap_write(nau8825->regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff); + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); @@ -1252,12 +1416,50 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return 0; } +#ifdef CONFIG_PM_SLEEP +static int nau8825_codec_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(nau8825->regmap, true); + + return 0; +} + +static int nau8825_codec_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(nau8825->regmap, false); + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + /* Check the jack plug status directly. If the headset is un- + * plugged during S3 when the chip has no power, there will + * be no jack detection irq event after restarting jack detec- + * tion, because the chip just thinks no headset has ever + * been plugged in. + */ + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + enable_irq(nau8825->irq); + + return 0; +} +#else +#define nau8825_codec_suspend NULL +#define nau8825_codec_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_codec_suspend, + .resume = nau8825_codec_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1351,29 +1553,13 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Because resume and initiation all need + * to enable the mechanism, move these configuration to resume setup + * function. */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, @@ -1442,36 +1628,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1498,7 +1654,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..1100385 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -93,6 +93,9 @@ #define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 #define NAU8825_REG_GENERAL_STATUS 0x82 #define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8825_REG_ADDR_LEN 16 +#define NAU8825_REG_DATA_LEN 16 /* ENA_CTRL (0x1) */ #define NAU8825_ENABLE_DACR_SFT 10 @@ -145,6 +148,7 @@ /* JACK_DET_CTRL (0xd) */ #define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_DET_DB_BYPASS (1 << 8) #define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 #define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) #define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 @@ -157,6 +161,7 @@ #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) +#define NAU8825_IRQ_INSERT_EN (1 << 0) /* IRQ_STATUS (0x10) */ #define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) @@ -177,6 +182,7 @@ #define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) #define NAU8825_IRQ_EJECT_DIS (1 << 2) +#define NAU8825_IRQ_INSERT_DIS (1 << 0) /* SAR_CTRL (0x13) */ #define NAU8825_SAR_ADC_EN_SFT 12 @@ -333,7 +339,8 @@ /* System Clock Source */ enum { - NAU8825_CLK_MCLK = 0, + NAU8825_CLK_DIS = 0, + NAU8825_CLK_MCLK, NAU8825_CLK_INTERNAL, NAU8825_CLK_FLL_MCLK, NAU8825_CLK_FLL_BLK, -- 2.6.4

2 12

[alsa-devel] i.MX6 SGTL5000 hangs with EIO error
by Ross Wille 12 May '16

12 May '16

Hello, I am doing an ALSA capture on a Wandboard Quad (i.MX6 quad core with SGTL5000 codec chip) from the LINEIN1 jack. I'm using a 4.6.0-rc3-armv7-x0 kernel. Sometimes when the audio glitches (for example, when I plug/unplug the audio cable or adjust my signal generator) the snd_pcm_readi() function will start returning -5 (EIO). Once this happens, the only way I've been able to recover is to reboot the computer. Calling snd_pcm_close(), snd_pcm_prepare(), snd_pcm_start(), etc. doesn't help. When in this state, running arecord returns IO errors as well. It's interesting that, on rare occasions, I must do a power cycle in order to recover. When a reboot is not effective I've noticed that the capture device doesn't appear in /proc/asound/devices. I don't believe my specific ALSA settings are important, but I'm calling snd_pcm_readi() with ALSA set to a sample rate of 48000, format of S16_LE, channels=1, frames per period of 960 (20 mS periods), and 4 periods per buffer. This same problem happens on two different Wandboards, so I don't think it's a defective board or chip. It has happened on older kernels as well. Any ideas? Thank you! Wille

1 0

[alsa-devel] Intel Cherry Trail and NAU8824 codec
by TheKit 11 May '16

11 May '16

Hello, I'm trying to get sound working in Linux on Cube iWork8 Ultimate tablet. It has Windows/Android dualboot, but no kernel sources for Android, unfortunately. From dmesg I see that NAU8824 codec is used, which I can't find any drivers for, except proposed patch that wasn't merged: https://lkml.org/lkml/2015/3/4/765 But, as I understand, additional code (like https://github.com/kirananto/ZENFONE2/blob/master/sound/soc/intel/board/cht…) is needed to bridge it with Cherry Trail SoC. Any suggestions how to go about this? Relevant lines of dmesg from stock Android kernel (where sound works): <6>[ 8.665477] nau8824_i2c_probe enter <3>[ 8.666609] snd_intel_sst: sst_platform_get_resources_edk <3>[ 8.672489] snd_intel_sst: LPE base: 0x91600000 size:0x200000snd_intel_sst: IRAM base: 0x916c0000 <3>[ 8.682557] snd_intel_sst: DRAM base: 0x91700000snd_intel_sst: SHIM base: 0x91740000 <3>[ 8.691298] snd_intel_sst: Mailbox base: 0x91744000snd_intel_sst: DDR base: 0x20000000 <6>[ 8.700428] snd_intel_sst: Enter: sst_audio_platform_init <6>[ 8.700646] snd_intel_sst: audio:set_cht_platform_config <6>[ 8.700900] snd_soc_sst_platform: sst_platform_probe is deferred <6>[ 8.700918] platform sst-platform: Driver sst-platform requests probe deferral <3>[ 8.700927] snd_intel_sst: board_name:Cherry Trail CR <6>[ 8.706611] snd_intel_sst: Registering machine device cht_nau88l24 <3>[ 8.707127] snd_intel_sst: Requesting FW fw_sst_22a8.bin now... <4>[ 8.713815] intel_sst_acpi 808622A8:00: Direct firmware load failed with error -2 <4>[ 8.724734] intel_sst_acpi 808622A8:00: Falling back to user helper <6>[ 8.789741] platform cht_nau88l24: Driver cht_nau88l24 requests probe deferral <6>[ 8.789967] cht_cr_nau88l24: Cherrytrail Machine Driver cht_nau8824 registerd <6>[ 9.077571] snd_soc_sst_platform: sst_platform_probe is deferred <3>[ 10.856689] sst-platform sst-platform: ASoC: no sink widget found for Audio Capture <3>[ 10.856693] sst-platform sst-platform: ASoC: Failed to add route media3_out -> direct -> Audio Capture <6>[ 10.873799] nau8824_codec_probe enter <6>[ 11.062658] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Headset-cpu-dai mapping ok <6>[ 11.062754] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Deepbuffer-cpu-dai mapping ok <6>[ 11.062902] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Voip-cpu-dai mapping ok <6>[ 11.063035] cht_cr_nau88l24: cht_hs_detection:Check hs insertion after 500 msec <6>[ 11.063035] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Lowlatency-cpu-dai mapping ok <6>[ 11.063383] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Probe-cpu-dai mapping ok <6>[ 11.063596] cht_nau88l24 cht_nau88l24: nau8824-aif1 <-> ssp2-port mapping ok <6>[ 11.063638] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> snd-soc-dummy-dai mapping ok <6>[ 11.063674] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> snd-soc-dummy-dai mapping ok <3>[ 11.063742] cht_nau88l24 cht_nau88l24: ASoC: Failed to add Headset Mic Switch: -16 <6>[ 11.095628] input: cherrytrailaud Headset Jack as /devices/platform/cht_nau88l24/sound/card1/input9 <6>[ 11.096265] cht_cr_nau88l24: snd_cht_mc_probe successful

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: add context information to tracepoints
by Takashi Sakamoto 11 May '16

11 May '16

In current implementation, packet processing is done in both of software IRQ contexts of IR/IT contexts and process contexts. This is usual interrupt handling of IR/IT context for 1394 OHCI. (in hardware IRQ context) irq_handler() (drivers/firewire/ohci.c) ->tasklet_schedule() (in software IRQ context) handle_it_packet() or handle_ir_packet_per_buffer() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() However, we have another chance for packet processing. It's done in PCM frame handling via ALSA PCM interfaces. (in process context) ioctl(i.e. SNDRV_PCM_IOCTL_HWSYNC) ->snd_pcm_hwsync() (sound/core/pcm_native.c) ->snd_pcm_update_hw_ptr() (sound/core/pcm_lib.c) ->snd_pcm_update_hw_ptr0() ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() (drivers/firewire/core-iso.c) ->struct fw_card_driver.flush_iso_completions() = ohci_flush_iso_completions() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() This design is for a better granularity of PCM pointer. When ioctl(2) is executed with some commands for ALSA PCM interface, queued packets are handled at first. Then, the latest number of handled PCM frames is reported. The number can represent PCM frames transferred in most near isochronous cycle. Current tracepoints include no information to distinguish running contexts. When tracing the interval of software IRQ context, this is not good. This commit adds more information for current context. Additionally, the index of packet processed in one context is added in a case that packet processing is executed in continuous context of the same kind, As a result, the output includes 11 fields with additional two fields to commit 0c95c1d6197f ("ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data"): 17131.9186: out_packet: 07 7494 ffc0 ffc1 00 000700c0 9001a496 058 45 1 13 17131.9186: out_packet: 07 7495 ffc0 ffc1 00 000700c8 9001ba00 058 46 1 14 17131.9186: out_packet: 07 7496 ffc0 ffc1 00 000700d0 9001ffff 002 47 1 15 17131.9189: out_packet: 07 7497 ffc0 ffc1 00 000700d0 9001d36a 058 00 0 00 17131.9189: out_packet: 07 7498 ffc0 ffc1 00 000700d8 9001e8d4 058 01 0 01 17131.9189: out_packet: 07 7499 ffc0 ffc1 00 000700e0 9001023e 058 02 0 00 17131.9206: in_packet: 07 7447 ffc1 ffc0 01 3f070072 9001783d 058 32 1 00 17131.9206: in_packet: 07 7448 ffc1 ffc0 01 3f070072 90ffffff 002 33 1 01 17131.9206: in_packet: 07 7449 ffc1 ffc0 01 3f07007a 900191a8 058 34 1 02 (Here, some common fields are omitted so that a line is within 80 characters.) The legend is: - The second of cycle scheduled for the packet - The count of cycle scheduled for the packet - The ID of node as source (hex) - The ID of node as destination (hex) - The value of isochronous channel - The first quadlet of CIP header (hex) - The second quadlet of CIP header (hex) - The number of included quadlets - The index of packet in a buffer maintained by this module - 0 in process context, 1 in IRQ context - The index of packet processed in the context Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 28 ++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 14 ++++++++------ 2 files changed, 28 insertions(+), 14 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 425d1d7..1622579 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -14,8 +14,8 @@ #include <linux/tracepoint.h> TRACE_EVENT(in_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets), - TP_ARGS(s, cycles, cip_header, payload_quadlets), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_quadlets, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -25,6 +25,8 @@ TRACE_EVENT(in_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -36,10 +38,12 @@ TRACE_EVENT(in_packet, __entry->cip_header0 = cip_header[0]; __entry->cip_header1 = cip_header[1]; __entry->payload_quadlets = payload_quadlets; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -48,12 +52,14 @@ TRACE_EVENT(in_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); TRACE_EVENT(out_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length), - TP_ARGS(s, cycles, cip_header, payload_length), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -63,6 +69,8 @@ TRACE_EVENT(out_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -74,10 +82,12 @@ TRACE_EVENT(out_packet, __entry->cip_header0 = be32_to_cpu(cip_header[0]); __entry->cip_header1 = be32_to_cpu(cip_header[1]); __entry->payload_quadlets = payload_length / 4; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -86,6 +96,8 @@ TRACE_EVENT(out_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index e7163bd..db30fac4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -411,7 +411,8 @@ static inline int queue_in_packet(struct amdtp_stream *s) amdtp_stream_get_max_payload(s)); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, + unsigned int index) { __be32 *buffer; unsigned int syt; @@ -436,7 +437,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; - trace_out_packet(s, cycle, buffer, payload_length); + trace_out_packet(s, cycle, buffer, payload_length, index); if (queue_out_packet(s, payload_length) < 0) return -EIO; @@ -450,7 +451,8 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, unsigned int cycle) + unsigned int payload_quadlets, unsigned int cycle, + unsigned int index) { __be32 *buffer; u32 cip_header[2]; @@ -465,7 +467,7 @@ static int handle_in_packet(struct amdtp_stream *s, cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); - trace_in_packet(s, cycle, cip_header, payload_quadlets); + trace_in_packet(s, cycle, cip_header, payload_quadlets, index); /* * This module supports 'Two-quadlet CIP header with SYT field'. @@ -604,7 +606,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, for (i = 0; i < packets; ++i) { cycle = increment_cycle_count(cycle, 1); - if (handle_out_packet(s, cycle) < 0) { + if (handle_out_packet(s, cycle, i) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -651,7 +653,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, break; } - if (handle_in_packet(s, payload_quadlets, cycle) < 0) + if (handle_in_packet(s, payload_quadlets, cycle, i) < 0) break; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: drop skip argument from helper functions to queue a packet
by Takashi Sakamoto 11 May '16

11 May '16

On most of audio and music units on IEEE 1394 bus which ALSA firewire stack supports (or plans to support), CIP with two quadlets header is used. Thus, there's no cases to queue packets with blank payload. If such packets are going to be queued, it means that they're for skips of the cycle. This commit simplifies helper functions to queue a packet. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 830a95c..e7163bd 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -370,9 +370,8 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, - unsigned int header_length, - unsigned int payload_length, bool skip) +static int queue_packet(struct amdtp_stream *s, unsigned int header_length, + unsigned int payload_length) { struct fw_iso_packet p = {0}; int err = 0; @@ -383,8 +382,10 @@ static int queue_packet(struct amdtp_stream *s, p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); p.tag = TAG_CIP; p.header_length = header_length; - p.payload_length = (!skip) ? payload_length : 0; - p.skip = skip; + if (payload_length > 0) + p.payload_length = payload_length; + else + p.skip = true; err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer, s->buffer.packets[s->packet_index].offset); if (err < 0) { @@ -399,16 +400,15 @@ end: } static inline int queue_out_packet(struct amdtp_stream *s, - unsigned int payload_length, bool skip) + unsigned int payload_length) { - return queue_packet(s, OUT_PACKET_HEADER_SIZE, - payload_length, skip); + return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length); } static inline int queue_in_packet(struct amdtp_stream *s) { return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s), false); + amdtp_stream_get_max_payload(s)); } static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) @@ -438,7 +438,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) trace_out_packet(s, cycle, buffer, payload_length); - if (queue_out_packet(s, payload_length, false) < 0) + if (queue_out_packet(s, payload_length) < 0) return -EIO; pcm = ACCESS_ONCE(s->pcm); @@ -764,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) err = queue_in_packet(s); else - err = queue_out_packet(s, 0, true); + err = queue_out_packet(s, 0); if (err < 0) goto err_context; } while (s->packet_index > 0); -- 2.7.4

3 3

[alsa-devel] [RFC][PATCH] ALSA: firewire-lib: permit process context only to flush queued packets for better PCM period granularity
by Takashi Sakamoto 11 May '16

11 May '16

Hi Clemens, Could I request your comment to this patch? It is to solve a race condition, but the condition is quite rare. Practically, it might have a less meanings, except for better program. And I think there's another race condition against processing each packets by calling out/in_stream_callback(), but I cannot observe the race. Software IRQ contexts of IR/IT contexts and process contexts are under the race condition, however I can see no problems related to it in my several trials in multi-core machine. I have no idea about the reason that packet sequence is processed correctly between the software IRQ contexts and the process contexts without any lock primitives. Do you have some ideas about it? Regards Takashi Sakamoto ----- 8< ----- Subject: ALSA: firewire-lib: permit process context only to flush queued packets for better PCM period granularity These three commits were merged to improve PCM pointer granularity. commit 76fb87894828 ("ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call") commit e9148dddc3c7 ("ALSA: firewire-lib: flush completed packets when reading PCM position") commit 92b862c7d685 ("ALSA: firewire-lib: optimize packet flushing") The point of them is to handle queued packets not only in software IRQ context of IR/IT contexts, but also in process context. This idea introduced a cyclic call of 'struct snd_pcm_ops.pointer()'. The last commit adds 'pointer_flush' member to 'struct amdtp_stream' to avoid the situation. On the other hand, This solution is weak at race condition between the process context and the software IRQ context of IR/IT contexts. Practically, this race is not so critical because it influences process context to skip flushing queued packets and to get worse granularity of PCM pointer. This is quite rare because 'struct snd_pcm_ops.pointer()' is frequently called by process contexts while IT/IR contexts run less frequently. However, it's better to solve the race. The similar solution can be achieved by 'in_interrupt()' macro. This commit applies the macro to solve the race condition against 'pointer_flush'. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 23 +++++++++++++++++------ sound/firewire/amdtp-stream.h | 1 - 2 files changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4484242..132c7f4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -248,7 +248,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -353,7 +352,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -798,11 +796,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets wll be queued in + * the buffer. These packets should be handled immediately to keep + * better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8775704..ab697cc 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -132,7 +132,6 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; -- 2.7.4

3 7

[alsa-devel] [PATCH] ALSA: usb-audio: Yet another Phoneix Audio device quirk
by Takashi Iwai 11 May '16

11 May '16

Phoenix Audio has yet another device with another id (even a different vendor id, 0556:0014) that requires the same quirk for the sample rate. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7b6dca2d7488..220764cd60bd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,6 +1137,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ + case uSB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ -- 2.8.2

3 3

[alsa-devel] [PATCH] ASoC: twl6040: Select LPPLL during standby
by Peter Ujfalusi 11 May '16

11 May '16

When the codec is in standby we do not need to keep the HPPLL active. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bc3de2e844e6..7a744b581cc1 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, { struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int ret = 0; switch (level) { case SND_SOC_BIAS_ON: @@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (priv->codec_powered) + if (priv->codec_powered) { + /* Select low power PLL in standby */ + ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL, + 32768, 19200000); break; + } ret = twl6040_power(twl6040, 1); if (ret) - return ret; + break; priv->codec_powered = 1; @@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, break; } - return 0; + return ret; } static int twl6040_startup(struct snd_pcm_substream *substream, -- 2.8.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix printk formatting
by Joonas Lahtinen 11 May '16

11 May '16

Format number after 0x in hex. Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Joonas Lahtinen <joonas.lahtinen(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1aa819c..994256b 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n", dma_area, size, pages); for (i = 0; i < pages; i++) { -- 2.5.5

2 1

[alsa-devel] [PATCH RFT] ASoC: max98371 Remove duplicate entry in max98371_reg
by Axel Lin 11 May '16

11 May '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98371.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 21fae09..cf0a39b 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -107,7 +107,6 @@ static const struct reg_default max98371_reg[] = { { 0x39, 0x00 }, { 0x3A, 0x00 }, { 0x3B, 0x00 }, - { 0x3B, 0x00 }, { 0x3C, 0x00 }, { 0x3D, 0x00 }, { 0x3E, 0x00 }, -- 2.5.0

2 1

[alsa-devel] [PATCH v5 0/4] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 11 May '16

11 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 5 of the patch: * rebased to the current master of git://git.kernel.org/pub/scm/linux/kernel/git/nsekhar/linux-davinci.git * independent changes submitted separately (3/7 and 7/7 of the version 4) * 5/7 of the version 4 removed as already applied * corrected mcbsp0 and mcbsp1 nodes in the DTS file: using correct address and sound@ prefix

3 7

[alsa-devel] [PATCH alsa-lib] pcm: dmix: Fix doubly resume of slave PCM
by Takashi Iwai 11 May '16

11 May '16

The dmix plugin and co may trigger the resume for each instance in snd_pcm_direct_resume(). It means that the slave PCM gets resumed or re-prepared/started by each opened dmix stream, and this may end up with the doubly triggers even though the slave PCM has been already resumed by another dmix stream. For avoiding this conflicts, check the slave PCM state and resume only when it's still in the suspended state. Meanwhile we keep the shadow state updated no matter whether the slave was triggered or not. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e28738b0de96..ac082f1a73b2 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -841,6 +841,12 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) int err; snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + /* resume only when the slave PCM is still in suspended state */ + if (snd_pcm_state(dmix->spcm) != SND_PCM_STATE_SUSPENDED) { + err = 0; + goto out; + } + err = snd_pcm_resume(dmix->spcm); if (err == -ENOSYS) { /* FIXME: error handling? */ @@ -848,6 +854,7 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) snd_pcm_start(dmix->spcm); err = 0; } + out: dmix->state = snd_pcm_state(dmix->spcm); snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); return err; -- 2.8.2

1 0

[alsa-devel] ASoC: DAPM and differential inputs
by Jeremy McDermond 10 May '16

10 May '16

I’ve been working on the tlv320aic32x4 driver for a while now to add some features for some projects I’m working on using the chip. I’m trying to wrap my head around DAPM and getting it to work with the variety of inputs on the aic32x4. Basically, the codec has a set of registers that control the positive and negative inputs to each of the right and left input mixers. The only setting is what value of resistor to connect between the input pin and a particular mixer (none, 10k, 20k, 40k). For example, you can have a differential pair between IN1_R and IN2_R, or between IN1_R and IN3_R, or single ended by setting the negative pin to the common. So, there’s a lot of flexibility and not just a differential “mode”. The registers would even allow you to set up two negative pins, although I’m not sure why that would be useful. I’m currently setting up DAPM so that both the positive and negative pins are wired to a single mixer. This doesn’t seem to be working since only the negative pin seems to cause the path to be enabled. Do I have to set up another “Negative Right” and “Negative Left” mixer to handle the negative pins? Am I missing something else? -- Jeremy McDermond (NH6Z) Xenotropic Systems mcdermj(a)xenotropic.com

1 0

[alsa-devel] [PATCH] MAINTAINERS: Add myself as reviewer of FSL/NXP SoC sound drivers
by Fabio Estevam 10 May '16

10 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> I would like to help reviewing FSL/NXP SoC sound drivers. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 43b85c1..f896605 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4846,6 +4846,7 @@ FREESCALE SOC SOUND DRIVERS M: Timur Tabi <timur(a)tabi.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> +R: Fabio Estevam <fabio.estevam(a)nxp.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: simple-card: Add pm callbacks to platform driver
by Peter Ujfalusi 10 May '16

10 May '16

Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, we have moved most of the TI boards to use simple-card and I have noticed that we have quite bad regression because of it caused by simple-card not having the pm ops configured. This will render audio to not recover after suspend/resume when the audio IP is loosing context. Regards, Peter sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c6111e7a6d93..52a1e1ff3c62 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -658,6 +658,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- 2.8.1

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Do not allow multiple streams in one direction
by Peter Ujfalusi 10 May '16

10 May '16

Make sure that the user can not start multiple streams with the same direction. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c4cd8e4fff9a..50523e744483 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1252,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, int i, dir; int tdm_slots = mcasp->tdm_slots; - if (mcasp->tdm_mask[substream->stream]) - tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + /* Do not allow more then one stream per direction */ + if (mcasp->substreams[substream->stream]) + return -EBUSY; mcasp->substreams[substream->stream] = substream; + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; -- 2.8.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: davinci-mcasp: Master mode clock divider correction
by Peter Ujfalusi 10 May '16

10 May '16

Hi, If the McASP is used as clock master and the reference clock is AUXCLK we can have additional level of divider. The BCLK divider is limited to maximum 32, if the desired bclk can not be reached with this, the AUXCLK divider also needs to be used. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Use defines for clkdiv IDs ASoC: davinci-mcasp: Change __davinci_mcasp_set_clkdiv() first parameter ASoC: davinci-mcasp: Restructure the davinci_mcasp_calc_clk_div() ASoC: davinci-mcasp: Calculate AUXCLK divider when setting up master clocks sound/soc/davinci/davinci-mcasp.c | 82 +++++++++++++++++++++++++-------------- sound/soc/davinci/davinci-mcasp.h | 5 +++ 2 files changed, 57 insertions(+), 30 deletions(-) -- 2.8.2

2 8

[alsa-devel] [PATCH] ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS
by Kuninori Morimoto 10 May '16

10 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_dmapp_get_id() returns 0xFF as error code if system used strange connection. It will be used as PDMACHCRn.SRS, but 0xFF is prohibited number. In order not to use prohibited number, this patch indicates error message and returns 0x00 (same as SSI00) in error case. Special thanks to Dung-san. Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8f..6bc93cb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: Intel: Add Broxton Machine
by Vinod Koul 10 May '16

10 May '16

Broxton uses SSP5 for audio on Intel board, so add the 6 SSPs and then machine driver which uses RT298 codec Pardha Saradhi K (1): ASoC: Intel: Skylake: Add more SSP DAIs Ramesh Babu (1): ASoC: Intel: Add Broxton-P machine driver sound/soc/intel/Kconfig | 15 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bxt_rt298.c | 353 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 72 ++++++++ 4 files changed, 442 insertions(+) create mode 100644 sound/soc/intel/boards/bxt_rt298.c -- 1.9.1

2 4

[alsa-devel] [PATCH 0/5] ASoC: da7213: clocking related improvements for codec
by Adam Thomson 10 May '16

10 May '16

This is a collection of updates with regards to clocking on da7213 codec, including power efficiency & stability improvements. Patches are based on v4.6-rc7 kernel. Adam Thomson (5): ASoC: da7213: Add DAI DAPM event to control DAI clocks ASoC: da7213: Add checking of SRM lock status before enabling DAI ASoC: da7213: Default PC counter to free-running when DAI disabled ASoC: da7213: Update PLL ranges to improve locking at frequency boundary ASoC: da7213: Allow PLL disable/bypass when using 32KHz sysclk sound/soc/codecs/da7213.c | 99 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 35 ++++++++++------- 2 files changed, 102 insertions(+), 32 deletions(-) -- 1.9.3

2 12

[alsa-devel] [PATCH 0/7] ALSA: drop reuse of incoming packet parameter for outgoing packet parameter
by Takashi Sakamoto 10 May '16

10 May '16

Hi, This patchset updates my former RFC, and go for merging to Linux 4.7. [alsa-devel] [RFC][PATCH 0/6] ALSA: firewire: drop reusing incoming packet parameters for outgoing packet parameters http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107290.html The aim of this patchset is to remove a feature from firewire-lib module. I call it as 'full duplex streaming with timestamp synchronization'. The feature is based on our understanding of IEC 61883-6. In the specification, a function block should reuse some parameters of incoming packets for outgoing packets. Current ALSA firewire stack partly implement it. However, actual audio and music units on IEEE 1394 bus don't expect corresponding drivers to satisfy the feature. On the other hand, the feature brings inconveniences to this module. When this feature is enabled, a software IRQ context processes packets for both of IR/IT contexts. This consumes more time over local CPUs than handling one context. Furthermore, cycle count for outgoing packets becomes improper because they're for incoming packets. For these reasons, this patchset removes the feature from ALSA firewire stack. Changes: - The last patch is newly added. Regards Takashi Sakamoto (7): ALSA: bebob: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: fireworks: drop reuse of incoming packet parameter for ougoing packet parameter ALSA: firewire-tascam: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: firewire-lib: handle IT/IR contexts in each software interrupt context ALSA: firewire-lib: code cleanup for incoming packet handling ALSA: firewire-lib: code cleanup for outgoing packet handling ALSA: firewire-lib: enable the same feature as CIP_SKIP_INIT_DBC_CHECK flag sound/firewire/amdtp-stream.c | 98 +++++++++------------------ sound/firewire/amdtp-stream.h | 36 ++-------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/bebob/bebob_stream.c | 101 +++++++--------------------- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_stream.c | 84 +++++------------------ sound/firewire/oxfw/oxfw-stream.c | 3 +- sound/firewire/tascam/tascam-stream.c | 26 ++++--- 9 files changed, 94 insertions(+), 258 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH] ALSA: pcm: Bail out when chmap is already present
by Takashi Iwai 10 May '16

10 May '16

When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a chmap has been already assigned, it returns as an error due to the conflicting snd_ctl_add() result. However, this also clears the already assigned chmap_kctl field via pcm_chmap_ctl_private_free(), and becomes inconsistent in the later operation. This patch adds the check of the conflicting chmap kctl before actually trying to allocate / assign. The check failure is treated as a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is basically a driver bug and having the stack trace would help developers to figure out the bad code path. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0aca39762ed0..bb1261591a1f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; -- 2.8.2

1 0

[alsa-devel] [PATCH 3/4] ASoC: ab8500-codec: remove platform data based probe
by Arnd Bergmann 10 May '16

10 May '16

The ux500 platform hasn't used board files for a long time, and nothing defines a ab8500_codec_platform_data, so we can just remove the probing based on that and always use device tree properties directly. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/ab8500-codec.c | 33 ++++----------------------------- 1 file changed, 4 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 8b1d0c1a7839..2fc89155f14a 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2464,45 +2464,20 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) struct device *dev = codec->dev; struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); - struct ab8500_platform_data *pdata; + struct ab8500_codec_platform_data codec_pdata; struct filter_control *fc; int status; dev_dbg(dev, "%s: Enter.\n", __func__); - /* Setup AB8500 according to board-settings */ - pdata = dev_get_platdata(dev->parent); + ab8500_codec_of_probe(dev, np, &codec_pdata); - if (np) { - if (!pdata) - pdata = devm_kzalloc(dev, - sizeof(struct ab8500_platform_data), - GFP_KERNEL); - - if (pdata && !pdata->codec) - pdata->codec - = devm_kzalloc(dev, - sizeof(struct ab8500_codec_platform_data), - GFP_KERNEL); - - if (!(pdata && pdata->codec)) - return -ENOMEM; - - ab8500_codec_of_probe(dev, np, pdata->codec); - - } else { - if (!(pdata && pdata->codec)) { - dev_err(dev, "No codec platform data or DT found\n"); - return -EINVAL; - } - } - - status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); + status = ab8500_audio_setup_mics(codec, &codec_pdata.amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); return status; } - status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv); + status = ab8500_audio_set_ear_cmv(codec, codec_pdata.ear_cmv); if (status < 0) { pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", __func__, status); -- 2.7.0

1 0

[alsa-devel] bug: reconfig broken on kernel 4.5
by Jochen Henneberg 10 May '16

10 May '16

Hi, it looks as if 'reconfig' feature is broken since the sysfs attributes have been moved to device attributes. In hda_sysfs.c:reconfig_codec() the device is cleared with snd_hda_codec_reset() which calls snd_hdac_device_unregister() which removes the sysfs entries (including the currently in-use 'reconfig'), the re-registration fails afterwards in snd_hda_codec_build_controls(). I am not sure when things got broken, but snd_hda_codec_reset() called from reconfig_codec() unregisters the device and removes the sysfs entries where in the past it only stripped all configurations (pcms, controls ...) from the device. Regards, Jochen

2 4

[alsa-devel] [PATCH] ALSA: Fix subwoofer pin on ASUS N751 and N551
by Takashi Iwai 10 May '16

10 May '16

From: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Subwoofer does not work out of the box on ASUS N751/N551 laptops. This patch fixes it. Patch tested on N751 laptop. N551 part is not tested, but according to [1] and [2] this laptop requires similar changes, so I included them in the patch. 1. https://github.com/honsiorovskyi/asus-n551-hda-fix 2. https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/1405691 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117781 Signed-off-by: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b5cd19876fc..4918ffa5ba68 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6426,6 +6426,7 @@ enum { ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, ALC662_FIXUP_ASUS_Nx50, + ALC668_FIXUP_ASUS_Nx51, }; static const struct hda_fixup alc662_fixups[] = { @@ -6672,6 +6673,15 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_BASS_1A }, + [ALC668_FIXUP_ASUS_Nx51] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x90170151}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6700,6 +6710,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), + SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), -- 2.8.2

1 0

[alsa-devel] Kernel config issue: SND_HDA_RECONFIG depends on SND_HDA_HWDEP
by Jochen Henneberg 09 May '16

09 May '16

Hi, with kernel 4.5 (and it looks as if this has not changed since then) the SND_HDA_RECONFIG sysfs entries still show up in the hwdep devices. Enabling SND_HDA_RECONFIG without SND_HDA_HWDEP is worth nothing, but the dependency has been removed from Kconfig. The commits from 13aeaf68 / 'Manage each codec instance individually' read as if the sysfs entries should show up at the codec driver but they do not. Regards, Jochen

2 5

[alsa-devel] [PATCH 0/5] Fix poll error returns
by Charles Keepax 09 May '16

09 May '16

We return negative values from the file_operations poll callback in a few places, this callback returns an unsigned int and is expected to only return the poll specific flags. This series fixes up these issues and makes a couple of small tidy ups to the code around the edges of those functions. I noticed this issue whilst doing some additional testing on my propagation of compressed stream error series, but I decided to push these up separately as it is worth getting the fixes in now and not tying them up with that patch chain which is taking longer to get merged. Also I included the first patch of that chain (Replace complex if statement with switch) because it is a trivial tidy up and might as well get merged now as well. Thanks, Charles Charles Keepax (5): ALSA: pcm: Fix poll error return codes ALSA: compress: Use snd_compr_get_poll on error path ALSA: compress: Remove pointless NULL check ALSA: compress: Fix poll error return codes ALSA: compress: Replace complex if statement with switch sound/core/compress_offload.c | 25 ++++++++++++------------- sound/core/pcm_native.c | 4 ++-- 2 files changed, 14 insertions(+), 15 deletions(-) -- 2.1.4

5 12

[alsa-devel] [PATCH 0/3 v3] firewire-lib: support tracepoints to dump a part of packet data
by Takashi Sakamoto 09 May '16

09 May '16

Hi, This patchset updates my RFCv2, and go for merging. [alsa-devel] [RFC][PATCH 0/3 v2] firewire-lib: support Linux tracing framework to dump a part of packet data http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106950.html The motivation of this patchset is to check the sequence of packet from/to audio and music units on IEEE 1394 bus. When the units have some quirks or ALSA firewire stack includes some bugs, we can work to seek them with Linux tracing framework. Changes: - Improve commit comments. - Improve Python3 script in commit comment for NODATA/NOINFO packets. - Move the position of trace_out_acket() so that the index of packet starts 0. Takashi Sakamoto (3): ALSA: firewire-lib: compute the value of second field in cycle count for IT context ALSA: firewire-lib: compute the value of second field in cycle count for IR context ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data sound/firewire/Makefile | 3 ++ sound/firewire/amdtp-stream-trace.h | 98 +++++++++++++++++++++++++++++++++++++ sound/firewire/amdtp-stream.c | 77 ++++++++++++++++++++++------- 3 files changed, 160 insertions(+), 18 deletions(-) create mode 100644 sound/firewire/amdtp-stream-trace.h -- 2.7.4

2 4

[alsa-devel] [PATCH alsa-lib] conf/ucm: ROCKCHIP-I2S: add Rockchip I2S UCM config.
by Enric Balletbo i Serra 09 May '16

09 May '16

Taken from the ChromeOS sources, this configuration was tested on Veyron Jerry based Chromebook from Google. Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/ROCKCHIP-I2S/HiFi.conf | 94 +++++++++++++++++++++++++++++ src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf | 6 ++ 3 files changed, 101 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/ROCKCHIP-I2S/HiFi.conf create mode 100644 src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index e6a6325..d1d51e5 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 ROCKCHIP-I2S diff --git a/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf b/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf new file mode 100644 index 0000000..7bfe995 --- /dev/null +++ b/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf @@ -0,0 +1,94 @@ +SectionVerb { + Value { + OutputDspName "speaker_eq" + MinBufferLevel "512" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Headphone Left Switch' off" + cset "name='Headphone Right Switch' off" + cset "name='Digital EQ 3 Band Switch' off" + cset "name='Digital EQ 5 Band Switch' off" + cset "name='Digital EQ 7 Band Switch' off" + cset "name='Biquad Switch' off" + cset "name='Filter Mode' Music" + cset "name='ADC Oversampling Rate' 0" + + cset "name='DMIC Mux' DMIC" + cset "name='MIC2 Mux' IN34" + cset "name='Right ADC Mixer MIC2 Switch' on" + cset "name='Left ADC Mixer MIC2 Switch' on" + cset "name='MIC2 Volume' 20" + cset "name='Headset Mic Switch' off" + cset "name='Int Mic Switch' on" + + cset "name='ADCR Boost Volume' 4" + cset "name='ADCL Boost Volume' 4" + cset "name='ADCR Volume' 11" + cset "name='ADCL Volume' 11" + + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Speaker Left Mixer Volume' 2" + cset "name='Speaker Right Mixer Volume' 2" + cset "name='Record Path DC Blocking' on" + cset "name='Playback Path DC Blocking' on" + + cset "name='Speaker Left Switch' on" + cset "name='Speaker Right Switch' on" + cset "name='Speaker Switch' on" + ] + + DisableSequence [ + ] +} + +SectionDevice."Headphone".0 { + Value { + JackName "ROCKCHIP-I2S Headset Jack" + OutputDspName "" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Speaker Switch' off" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + ] + DisableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Headphone Left Switch' off" + cset "name='Headphone Right Switch' off" + cset "name='Speaker Switch' on" + ] +} + +SectionDevice."Mic".0 { + Value { + JackName "ROCKCHIP-I2S Headset Jack" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Int Mic Switch' off" + cset "name='DMIC Mux' ADC" + cset "name='Headset Mic Switch' on" + cset "name='Record Path DC Blocking' on" + ] + + DisableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Headset Mic Switch' off" + cset "name='DMIC Mux' DMIC" + cset "name='Int Mic Switch' on" + cset "name='Record Path DC Blocking' off" + ] +} diff --git a/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf b/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf new file mode 100644 index 0000000..0c399b0 --- /dev/null +++ b/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf @@ -0,0 +1,6 @@ +Comment "Rockchip card" + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} -- 2.1.0

2 1

[alsa-devel] [PATCH] pcm: softvol: fix conversion of TLVs min_db and max_dB value
by Jörg Krause 09 May '16

09 May '16

Both, min_dB and max_dB, are floating type whereas the TLV is (always) unsigned. The problem with the conversion of a negative floating-point number into an unsigned integer is, that the behavior is undefined. This may, depending on the platform, result in a wrong TLV, i.e. for the default values of min_dB (-51dB) and max_dB (0dB), alsactl generates the following state on an ARM cpu build with GCC: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } With the fix applied, alsactl stores the correct TLV: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Also tested for different combinations of min_dB and max_dB other than the default values. Replaces: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107733.html Fixes: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107628.html Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- src/pcm/pcm_softvol.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 802aa4b..061080d 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -658,8 +658,8 @@ static int add_tlv_info(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo) unsigned int tlv[4]; tlv[0] = SND_CTL_TLVT_DB_SCALE; tlv[1] = 2 * sizeof(int); - tlv[2] = svol->min_dB * 100; - tlv[3] = (svol->max_dB - svol->min_dB) * 100 / svol->max_val; + tlv[2] = (int)(svol->min_dB * 100); + tlv[3] = (int)((svol->max_dB - svol->min_dB) * 100 / svol->max_val); return snd_ctl_elem_tlv_write(svol->ctl, &cinfo->id, tlv); } -- 2.8.2

2 1

[alsa-devel] [PATCH] ASoC: pcm: allow changing the playback/capture rates for symmetric links
by Peter Rosin 09 May '16

09 May '16

The below program fails on a dai link with symmetric rates without this patch. The patch makes it work. #include <sys/soundcard.h> #include <unistd.h> #include <fcntl.h> #include <stropts.h> #include <stdio.h> int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22050; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4fa86..1e876ff23524 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -237,6 +237,14 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int rate, channels, sample_bits, symmetry, i; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (!cpu_dai->capture_active) + return 0; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!cpu_dai->playback_active) + return 0; + } + rate = params_rate(params); channels = params_channels(params); sample_bits = snd_pcm_format_physical_width(params_format(params)); -- 2.1.4

3 12

[alsa-devel] [PATCH 0/2] wm5110: fixes to ANC clock control
by Richard Fitzgerald 09 May '16

09 May '16

The register for ANC clock control has a combination of volatile and non-readback bits and these two patches fix the way we write to this register. Both patches must be taken together to ensure correct operation. Richard Fitzgerald (2): mfd: wm5110: ARIZONA_CLOCK_CONTROL should be volatile ASoC: arizona: No need to update_bits when writing AEC clock control drivers/mfd/wm5110-tables.c | 1 + sound/soc/codecs/arizona.c | 3 +-- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: Add DSP firmware manifest parsing
by Vinod Koul 09 May '16

09 May '16

From: Shreyas NC <shreyas.nc(a)intel.com> Module params like module_id and loadable flag can be changed in the DSP Firmware. These are kept in the firmware manifest and driver should read these values from this manifest. So, add support to parse the DSP firmware manifest and read these module params. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Changes in v3: add bound check for firmware file sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 118 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 107 +++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 4 ++ sound/soc/intel/skylake/skl-sst.c | 11 ++- sound/soc/intel/skylake/skl-topology.c | 5 ++ 6 files changed, 245 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index c28f5d0e1d99..e2fb5b763b21 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o + skl-sst.o skl-dsp-parse.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-dsp-parse.c b/sound/soc/intel/skylake/skl-dsp-parse.c new file mode 100644 index 000000000000..a3cd5d7ab246 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.c @@ -0,0 +1,118 @@ +/* + * skl-dsp-parse.c - Implements DSP firmware parsing + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-tplg-interface.h" +#include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" + +/* + * Get the module id for the module by checking + * the table for the UUID for the module + */ +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config) +{ + int i, num; + struct uuid_tbl *tbl; + uuid_le *uuid_mod; + + tbl = ctx->tbl; + num = ctx->num_modules; + uuid_mod = (uuid_le *)uuid; + + for (i = 0; i < num; i++) { + if (uuid_le_cmp(*uuid_mod, tbl[i].uuid) == 0) { + dfw_config->module_id = tbl[i].module_id; + dfw_config->is_loadable = tbl[i].is_loadable; + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_skl_get_module_info); + +/* + * Parse the firmware binary to get the UUID, module id + * and loadable flags + */ +int snd_skl_parse_fw_bin(struct sst_dsp *ctx) +{ + struct adsp_fw_hdr *adsp_hdr; + struct adsp_module_entry *mod_entry; + int i, num_entry; + uuid_le *uuid_bin; + const char *buf; + struct skl_sst *skl = ctx->thread_context; + struct uuid_tbl *tbl; + + /* Get the FW pointer to derive ADSP header */ + buf = ctx->fw->data; + + adsp_hdr = (struct adsp_fw_hdr *)(buf + SKL_ADSP_FW_BIN_HDR_OFFSET); + + mod_entry = (struct adsp_module_entry *) + (buf + SKL_ADSP_FW_BIN_HDR_OFFSET + adsp_hdr->header_len); + + num_entry = adsp_hdr->num_module_entries; + + tbl = devm_kzalloc(ctx->dev, + num_entry * sizeof(struct uuid_tbl), GFP_KERNEL); + + if (!tbl) + return -ENOMEM; + + /* + * Read the UUID(GUID) from FW Manifest. + * The 16 byte UUID is of the format: + * XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXX + * Populate the UUID table to store module_id + * and loadable flags for the module. + */ + + for (i = 0; i < num_entry; i++, mod_entry++) { + + /* + * we check if current pointer is larger than file size from + * base value to check excceding the file while parsing + */ + if ((const char *)mod_entry >= buf + ctx->fw->size) { + dev_err(ctx->dev, + "Exceeds file bound: Entry %d Ptr %p\n", + i, mod_entry); + + return -EIO; + } + + uuid_bin = (uuid_le *)mod_entry->uuid.id; + memcpy(&tbl[i].uuid, uuid_bin, sizeof(tbl[i].uuid)); + + tbl[i].module_id = i; + tbl[i].is_loadable = mod_entry->type.load_type; + } + + skl->tbl = tbl; + skl->num_modules = num_entry; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_skl_parse_fw_bin); diff --git a/sound/soc/intel/skylake/skl-dsp-parse.h b/sound/soc/intel/skylake/skl-dsp-parse.h new file mode 100644 index 000000000000..cfdb24c45035 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.h @@ -0,0 +1,107 @@ +/* + * skl-dsp-parse.h + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <asm/types.h> +#include <linux/types.h> +#include <linux/uuid.h> +#include "skl-tplg-interface.h" +#include "../common/sst-dsp-priv.h" + +#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 +#define UUID_STR_SIZE 37 +#define DEFAULT_HASH_SHA256_LEN 32 + +struct skl_dfw_module_mod { + char name[100]; + struct skl_dfw_module skl_dfw_mod; +}; + +struct UUID { + u8 id[16]; +}; + +union seg_flags { + u32 ul; + struct { + u32 contents : 1; + u32 alloc : 1; + u32 load : 1; + u32 read_only : 1; + u32 code : 1; + u32 data : 1; + u32 _rsvd0 : 2; + u32 type : 4; + u32 _rsvd1 : 4; + u32 length : 16; + } r; +} __packed; + +struct segment_desc { + union seg_flags flags; + u32 v_base_addr; + u32 file_offset; +}; + +struct module_type { + u32 load_type : 4; + u32 auto_start : 1; + u32 domain_ll : 1; + u32 domain_dp : 1; + u32 rsvd_ : 25; +} __packed; + +struct adsp_module_entry { + u32 struct_id; + u8 name[8]; + struct UUID uuid; + struct module_type type; + u8 hash1[DEFAULT_HASH_SHA256_LEN]; + u32 entry_point; + u16 cfg_offset; + u16 cfg_count; + u32 affinity_mask; + u16 instance_max_count; + u16 instance_bss_size; + struct segment_desc segments[3]; +} __packed; + +struct adsp_fw_hdr { + u32 header_id; + u32 header_len; + u8 name[8]; + u32 preload_page_count; + u32 fw_image_flags; + u32 feature_mask; + u16 major_version; + u16 minor_version; + u16 hotfix_version; + u16 build_version; + u32 num_module_entries; + u32 hw_buf_base_addr; + u32 hw_buf_length; + u32 load_offset; +} __packed; + +struct uuid_tbl { + uuid_le uuid; + int module_id; + int is_loadable; +}; + +int snd_skl_parse_fw_bin(struct sst_dsp *ctx); +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index d59d1ba62a43..67d56be8cb77 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -60,6 +60,10 @@ struct skl_sst { void (*enable_miscbdcge)(struct device *dev, bool enable); /*Is CGCTL.MISCBDCGE disabled*/ bool miscbdcg_disabled; + + /* Populate module information */ + struct uuid_tbl *tbl; + int num_modules; }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index bec4a7c486fd..df7964fd09c3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -25,6 +25,7 @@ #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" #include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" #define SKL_BASEFW_TIMEOUT 300 #define SKL_INIT_TIMEOUT 1000 @@ -70,7 +71,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { - int ret = 0, i; + int ret = 0, i, err; struct skl_sst *skl = ctx->thread_context; u32 reg; @@ -84,6 +85,14 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) skl_dsp_disable_core(ctx); return -EIO; } + + err = snd_skl_parse_fw_bin(ctx); + if (err < 0) { + dev_err(ctx->dev, + "Firmware parsing err: %d\n", ret); + release_firmware(ctx->fw); + return err; + } } ret = skl_dsp_boot(ctx); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a1f4478fabcb..f690b016203c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -28,6 +28,7 @@ #include "skl-tplg-interface.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "skl-dsp-parse.h" #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) @@ -1566,6 +1567,10 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, w->priv = mconfig; memcpy(&mconfig->guid, &dfw_config->uuid, 16); + ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config); + if (ret < 0) + return ret; + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; -- 1.9.1

2 2

[alsa-devel] [PATCH v2 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 09 May '16

09 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. History: v2: Fix memory leak on realloc failure. The original obj pointer is reserved and so it can be freed at last by tplg_elem_free(). Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 26 +++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 133 insertions(+), 58 deletions(-) -- 2.5.0

2 7

[alsa-devel] [PATCH v2 1/2] ALSA: hda: fix the missing ptr initialization
by Vinod Koul 09 May '16

09 May '16

ebus is a member of extended device and was never initialized, so do this at device creation. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..64de0a3d6d93 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -144,6 +144,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); -- 1.9.1

2 4

[alsa-devel] [PATCH] ALSA: dice: add support for M-Audio Profire 610 and perhaps 2626
by Takashi Sakamoto 09 May '16

09 May '16

M-Audio Profire 610 has an unexpected value in version field of its config ROM, thus ALSA dice driver is not assigned to the model due to a mismatch of modalias. This commit adds an entry to support the model. I expect the entry is also for Profire 2626. I note that Profire 610 uses TCD2220 (so-called Dice Jr.), and supports a part of Extended Application Protocol (EAP). $ cd linux-firewire-utils/src $ ./crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ------------------------------------------------------------ 400 04047689 bus_info_length 4, crc_length 4, crc 30345 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 000d6c04 company_id 000d6c | 410 04400002 device_id 0404400002 | EUI-64 000d6c0404400002 root directory ------------------------------------------------------------ 414 000695fe directory_length 6, crc 38398 418 03000d6c vendor 41c 8100000a --> descriptor leaf at 444 420 17000011 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ------------------------------------------------------------ 430 0004fb14 directory_length 4, crc 64276 434 12000d6c specifier id 438 130100d1 version 43c 17000011 model 440 8100000c --> descriptor leaf at 470 descriptor leaf at 444 ------------------------------------------------------------ 444 0004b8e4 leaf_length 4, crc 47332 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4d2d4175 "M-Au" 454 64696f00 "dio" descriptor leaf at 458 ------------------------------------------------------------ 458 00053128 leaf_length 5, crc 12584 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 50726f46 "ProF" 468 69726520 "ire " 46c 36313000 "610" descriptor leaf at 470 ------------------------------------------------------------ 470 00053128 leaf_length 5, crc 12584 474 00000000 textual descriptor 478 00000000 minimal ASCII 47c 50726f46 "ProF" 480 69726520 "ire " 484 36313000 "610" $ cat /proc/asound/card1/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc0:000100000000 notification: 00000040 nick name: FW610 clock select: internal 48000 enable: 1 status: locked 48000 ext status: 00000040 sample rate: 48000 version: 1.0.4.0 clock caps: 32000 44100 48000 88200 96000 176400 192000 aes1 aes4 aes adat tdif wc arx1 arx2 internal clock source names: SPDIF\AES34\AES56\TOS\AES_ANY\ADAT\ADAT_AUX\Word Clock\Unused\Unused\Unused\Unused\Internal\\ ... Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96fe68f4..25e9f77 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -317,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); -- 2.7.4

2 4

[alsa-devel] [PATCH] pcm: softvol: fix signedness of tlv data type
by Jörg Krause 08 May '16

08 May '16

For the softvol plugin the TLV data are written wrong, e.g. for the default values of min_dB = -51 dB and max_dB = 0 dB, alsactl generates the following state: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } As both min_dB and max_dB can be negative numbers, the tlv type must not be unsigned. This fixes the TLV and the correct state is generated: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Also tested for different combinations of min_dB and max_dB other than the default values. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- src/pcm/pcm_softvol.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 802aa4b..1748667 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -655,7 +655,7 @@ static void snd_pcm_softvol_dump(snd_pcm_t *pcm, snd_output_t *out) static int add_tlv_info(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo) { - unsigned int tlv[4]; + int tlv[4]; tlv[0] = SND_CTL_TLVT_DB_SCALE; tlv[1] = 2 * sizeof(int); tlv[2] = svol->min_dB * 100; @@ -768,7 +768,7 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, } } else if (svol->max_val > 1) { /* check TLV availability */ - unsigned int tlv[4]; + int tlv[4]; err = snd_ctl_elem_tlv_read(svol->ctl, &cinfo->id, tlv, sizeof(tlv)); if (err < 0) add_tlv_info(svol, cinfo); -- 2.8.2

2 2

[alsa-devel] [PATCH 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 08 May '16

08 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 24 ++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 131 insertions(+), 58 deletions(-) -- 2.5.0

5 11

Re: [alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Kailang 08 May '16

08 May '16

Hi Takashi, Could you receive this mail? Thanks. BR, Kailang ________________________________ From: Kailang Sent: Wednesday, May 04, 2016 4:13 PM To: Takashi Iwai (tiwai(a)suse.de) Cc: (alsa-devel(a)alsa-project.org) Subject: New codecs support for ALC234/ALC274/ALC294 Hi Takashi, We have new codecs for ALC234/ALC274/ALC294. Attach file is the support patch. BR, Kailang

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Re: [alsa-devel] ALSA: hda - Fix white noise on Asus UX501VW headset
by Takashi Iwai 08 May '16

08 May '16

On Thu, 05 May 2016 16:08:02 +0200, Kaho Ng wrote: > > For reducing the noise from the headset output on ASUS UX501VW, > call the existing fixup, alc_fixup_headset_mode_alc668(), additionally. > > Thread: https://bbs.archlinux.org/viewtopic.php?id=209554 > > Signed-off-by: Kaho Ng <ngkaho1234(a)gmail.com> > --- > diff -ruNp a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > --- a/sound/pci/hda/patch_realtek.c 2016-04-20 14:44:02.000000000 +0800 > +++ b/sound/pci/hda/patch_realtek.c 2016-05-03 16:31:30.815677146 +0800 > @@ -6424,6 +6424,7 @@ enum { > ALC668_FIXUP_DELL_DISABLE_AAMIX, > ALC668_FIXUP_DELL_XPS13, > ALC662_FIXUP_ASUS_Nx50, > + ALC668_FIXUP_ASUS_UX501VW, > }; > > static const struct hda_fixup alc662_fixups[] = { > @@ -6670,6 +6671,10 @@ static const struct hda_fixup alc662_fix > .chained = true, > .chain_id = ALC662_FIXUP_BASS_1A > }, > + [ALC668_FIXUP_ASUS_UX501VW] = { > + .type = HDA_FIXUP_FUNC, > + .v.func = alc_fixup_headset_mode_alc668, > + }, > }; There is already a fixup entry (ALC668_FIXUP_HEADSET_MODE) doing the very same thing, so you don't have to redefine it. > static const struct snd_pci_quirk alc662_fixup_tbl[] = { > @@ -6692,6 +6697,7 @@ static const struct snd_pci_quirk alc662 > SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), > + SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_ASUS_UX501VW), And just add the quirk entry like SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), Could you resubmit the fix patch after adjusting like this? thanks, Takashi

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Re: [alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Takashi Iwai 08 May '16

08 May '16

On Wed, 04 May 2016 10:12:41 +0200, Kailang wrote: > > Hi Takashi, > > We have new codecs for ALC234/ALC274/ALC294. > Attach file is the support patch. Thanks, applied now. Takashi

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[alsa-devel] [patch] ALSA: isa/wavefront: prevent some out of bound writes
by Dan Carpenter 08 May '16

08 May '16

"header->number" can be up to USHRT_MAX and it comes from the ioctl so it needs to be capped. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff..718d5e3 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics

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Re: [alsa-devel] [PATCH] infoleak fix1 in timer
by Takashi Iwai 08 May '16

08 May '16

On Tue, 03 May 2016 22:44:07 +0200, Kangjie Lu wrote: > > The stack object “tread” has a total size of 32 bytes. Its field > “event” and “val” both contain 4 bytes padding. These 8 bytes > padding bytes are sent to user without being initialized. > > Signed-off-by: Kangjie Lu <kjlu(a)gatech.edu> Thanks, I applied now all three patches. But I rephrased the subjects to more meaningful ones. Better to give a bit more informative subject line at the next time. Takashi

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[alsa-devel] [PATCH] ALSA: compress: fix some typo
by Vinod Koul 08 May '16

08 May '16

Noticed two typos in Documentation, so fix them up Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Documentation/sound/alsa/compress_offload.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 630c492c3dc2..8ba556a131c3 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -149,7 +149,7 @@ Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder delay and padding and directly move from one track content to another. The end -user can perceive this as gapless playback as we dont have silence while +user can perceive this as gapless playback as we don't have silence while switching from one track to another Also, there might be low-intensity noises due to encoding. Perfect gapless is @@ -184,7 +184,7 @@ Sequence flow for gapless would be: - Fill data of the first track - Trigger start - User-space finished sending all, -- Indicaite next track data by sending set_next_track +- Indicate next track data by sending set_next_track - Set metadata of the next track - then call partial_drain to flush most of buffer in DSP - Fill data of the next track -- 1.9.1

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[alsa-devel] sound: deadlock involving snd_hrtimer_callback
by Dmitry Vyukov 08 May '16

08 May '16

Hi Takashi, I've incorporated your hrtimer fixes (but also updated to ddce192106e4f984123884f8e878f66ace94b573) and now I am seeing lots of the following deadlock messages: [ INFO: possible circular locking dependency detected ] 4.6.0-rc4+ #351 Not tainted ------------------------------------------------------- swapper/0/0 is trying to acquire lock: (&(&timer->lock)->rlock){-.-...}, at: [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 but task is already holding lock: (&(&stime->lock)->rlock){-.....}, at: [<ffffffff85383d3f>] snd_hrtimer_callback+0x4f/0x2b0 sound/core/hrtimer.c:54 which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&(&stime->lock)->rlock){-.....}: [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff85383baa>] snd_hrtimer_start+0x4a/0xf0 sound/core/hrtimer.c:112 [<ffffffff85379c74>] snd_timer_start1+0x2b4/0x5a0 sound/core/timer.c:457 [<ffffffff8537d33d>] snd_timer_start+0x5d/0xa0 sound/core/timer.c:571 [< inline >] seq_timer_start sound/core/seq/seq_timer.c:393 [<ffffffff853fc260>] snd_seq_timer_start+0x1a0/0x2b0 sound/core/seq/seq_timer.c:405 [< inline >] snd_seq_queue_process_event sound/core/seq/seq_queue.c:687 [<ffffffff853f6f14>] snd_seq_control_queue+0x304/0x8b0 sound/core/seq/seq_queue.c:748 [<ffffffff853fcbb5>] event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 [<ffffffff853ef7d4>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:636 [<ffffffff853efc38>] snd_seq_deliver_event+0x118/0x800 sound/core/seq/seq_clientmgr.c:833 [<ffffffff853f1026>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2418 [<ffffffff85405c2b>] send_timer_event.isra.0+0x10b/0x150 sound/core/seq/oss/seq_oss_timer.c:153 [<ffffffff8540615a>] snd_seq_oss_timer_start+0x1ca/0x310 sound/core/seq/oss/seq_oss_timer.c:174 [< inline >] old_event sound/core/seq/oss/seq_oss_event.c:125 [<ffffffff854095ff>] snd_seq_oss_process_event+0xa1f/0x2ce0 sound/core/seq/oss/seq_oss_event.c:100 [< inline >] insert_queue sound/core/seq/oss/seq_oss_rw.c:179 [<ffffffff8540c201>] snd_seq_oss_write+0x321/0x810 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff85403f99>] odev_write+0x59/0xa0 sound/core/seq/oss/seq_oss.c:177 [<ffffffff817f3fc3>] __vfs_write+0x113/0x4b0 fs/read_write.c:529 [<ffffffff817f5a97>] vfs_write+0x167/0x4a0 fs/read_write.c:578 [< inline >] SYSC_write fs/read_write.c:625 [<ffffffff817f98d1>] SyS_write+0x111/0x220 fs/read_write.c:617 [<ffffffff867c0440>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 -> #0 (&(&timer->lock)->rlock){-.-...}: [< inline >] check_prev_add kernel/locking/lockdep.c:1823 [< inline >] check_prevs_add kernel/locking/lockdep.c:1933 [< inline >] validate_chain kernel/locking/lockdep.c:2238 [<ffffffff81474035>] __lock_acquire+0x3625/0x4d00 kernel/locking/lockdep.c:3298 [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [< inline >] arch_safe_halt ./arch/x86/include/asm/paravirt.h:118 [<ffffffff8120f402>] default_idle+0x52/0x370 arch/x86/kernel/process.c:307 [<ffffffff81210bba>] arch_cpu_idle+0xa/0x10 arch/x86/kernel/process.c:298 [<ffffffff81454c28>] default_idle_call+0x48/0xa0 kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call kernel/sched/idle.c:151 [< inline >] cpu_idle_loop kernel/sched/idle.c:242 [<ffffffff8145520f>] cpu_startup_entry+0x58f/0x7b0 kernel/sched/idle.c:291 [<ffffffff8679ac6d>] rest_init+0x18d/0x1a0 init/main.c:408 [<ffffffff88ce7759>] start_kernel+0x63a/0x660 init/main.c:661 [<ffffffff88ce6364>] x86_64_start_reservations+0x38/0x3a arch/x86/kernel/head64.c:195 [<ffffffff88ce64be>] x86_64_start_kernel+0x158/0x167 arch/x86/kernel/head64.c:176 other info that might help us debug this: Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&(&stime->lock)->rlock); lock(&(&timer->lock)->rlock); lock(&(&stime->lock)->rlock); lock(&(&timer->lock)->rlock); *** DEADLOCK *** 1 lock held by swapper/0/0: #0: (&(&stime->lock)->rlock){-.....}, at: [<ffffffff85383d3f>] snd_hrtimer_callback+0x4f/0x2b0 sound/core/hrtimer.c:54 stack backtrace: CPU: 0 PID: 0 Comm: swapper/0 Not tainted 4.6.0-rc4+ #351 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88003ec07a50 ffffffff82c8f92f ffffffff00000000 fffffbfff0fd64b8 ffffffff8997bad0 ffffffff8997bad0 ffffffff8998aa70 ffffffff87e27510 ffffffff87e26d00 ffff88003ec07aa0 ffffffff8146cf58 Call Trace: <IRQ> [< inline >] __dump_stack lib/dump_stack.c:15 <IRQ> [<ffffffff82c8f92f>] dump_stack+0x12e/0x18f lib/dump_stack.c:51 [<ffffffff8146cf58>] print_circular_bug+0x288/0x340 kernel/locking/lockdep.c:1196 [< inline >] check_prev_add kernel/locking/lockdep.c:1823 [< inline >] check_prevs_add kernel/locking/lockdep.c:1933 [< inline >] validate_chain kernel/locking/lockdep.c:2238 [<ffffffff81474035>] __lock_acquire+0x3625/0x4d00 kernel/locking/lockdep.c:3298 [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [< inline >] arch_safe_halt ./arch/x86/include/asm/paravirt.h:118 [<ffffffff8120f402>] default_idle+0x52/0x370 arch/x86/kernel/process.c:307 [<ffffffff81210bba>] arch_cpu_idle+0xa/0x10 arch/x86/kernel/process.c:298 [<ffffffff81454c28>] default_idle_call+0x48/0xa0 kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call kernel/sched/idle.c:151 [< inline >] cpu_idle_loop kernel/sched/idle.c:242 [<ffffffff8145520f>] cpu_startup_entry+0x58f/0x7b0 kernel/sched/idle.c:291 [<ffffffff8679ac6d>] rest_init+0x18d/0x1a0 init/main.c:408 [<ffffffff88ce7759>] start_kernel+0x63a/0x660 init/main.c:661 [<ffffffff88ce6364>] x86_64_start_reservations+0x38/0x3a arch/x86/kernel/head64.c:195 [<ffffffff88ce64be>] x86_64_start_kernel+0x158/0x167 arch/x86/kernel/head64.c:176 They are also followed by rcu stalls and actual spinlock lockups: BUG: spinlock lockup suspected on CPU#3, syz-fuzzer/7040 lock: 0xffff880032ae2bd8, .magic: dead4ead, .owner: syz-executor/11206, .owner_cpu: 0 CPU: 3 PID: 7040 Comm: syz-fuzzer Not tainted 4.6.0-rc4+ #351 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88006d507c40 ffffffff82c8f92f ffffffff2c963698 fffffbfff0fd64b8 ffff880032ae2bd8 ffff88002c963040 ffff880061238040 ffff880032ae2be0 000000009a9d0630 ffff88006d507c78 ffffffff8147f58d Call Trace: <IRQ> [< inline >] __dump_stack lib/dump_stack.c:15 <IRQ> [<ffffffff82c8f92f>] dump_stack+0x12e/0x18f lib/dump_stack.c:51 [<ffffffff8147f58d>] spin_dump+0x14d/0x280 kernel/locking/spinlock_debug.c:67 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:117 [<ffffffff8147f8ad>] do_raw_spin_lock+0x15d/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff867c0067>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 <EOI>

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[alsa-devel] Unlikely big volume range Plantronics BT300
by Dennis Kadioglu 08 May '16

08 May '16

Hi, when I connect the bluetooth BT300 of my Plantronics Headset I can see the following warning: [172855.985539] usb 2-1.3: Warning! Unlikely big volume range (=720), cval->res is probably wrong. [172855.985544] usb 2-1.3: [5] FU [PCM Playback Volume] ch = 1, val = -10240/1280/16 As far as I understand it, the problem may be solved by applying a quirk which is already applied for Logitech webcams. Can somebody help me in finding out the correct value for cval->res. I have no clue which calculations are necessary for that. Thanks in advance. Dennis Kadioglu

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Re: [alsa-devel] WM5102 - Help to make baytrail machine driver work
by Pierre-Louis Bossart 07 May '16

07 May '16

On 5/4/16 8:21 PM, Paulo Sergio wrote: > Hi list, > > I've been trying to make Lenovo Yoga 2 (Model 1051F) sound work on > Android-x86/ Linux. [1] > > Based on already discussed info from Christian Hartmann with members > (see [2] and [3] ) of this list and some efforts (basically patch pxa2xx > with Mika Westerberg changes, set the correct gpio int values {246 for > reset / 405 for ldoena and 342 for irq_gpio} and set LDO1 vcc suppliers) > we could have wm5102 initialized through spi / regmap (arizona code on > drivers/mfd) and register the DAI's. > > I also ported a machine driver code from a kernel Lenovo made available > for a similar hardware (Lenovo yoga 2 1050 - uses the same baytrail > Z3745 and wolfson wm5102 audio) [4] There are quite a few references in the machine driver to concepts related to Clovertrail+ and early versions of Baytrail. I am fairly convinced the mentions of 'BYT_COMMS_BT' or 'modem mixing' are no longer supported with the DPCM platform driver/firmware you used. To enable this codec it would be simpler to start with a newer machine driver derived from the bytcr_rt5640 one, with just a basic SSP0-aif1 link for starters (+ MCLK). Better yet, do this work with a MinnowBoard and an WM5102 eval board and test all this with more control on how the hardware is setup. If you stick to working directly on the Lenovo device, make sure you use all the DSP controls set in the UCM files used for baytrail/baytrail-CR (see github.com/plbossart/UCM). > > Although machine driver is registered, we still don't have sound. Tried > some mixer settings / routes used on raspberry pi, but none help. > Application (in this case alsa / Android players) returns a "Device or > resource busy" when trying to ioctl pcm device: > > 04-30 13:05:07.877 1784 2078 E audio_hw_primary: out_write: error -1 - cannot prepare channel: Device or resource busy > > # This is raised here (external/tinyalsa/pcm.c): > > if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_PREPARE) < 0) > > return oops(pcm, errno, "cannot prepare channel"); > > > Note: As pointed here [5] had to change the acpi_ipc_irq_index to 0 on > sound/soc/intel/atom/sst/sst_acpi.c because this is the 1st ACPI entry > listed on DSDT > I'm also trying to use a fw version that uses SSP0 [6] instead of SSP2 > used as default - but no sound with both > > > I'd like some help from you in order to make this work. I've collect > several info which can be found here [7] > - /proc/asound/cards > - /proc/interrupts > - amixer controls list > - dmesg > - DSDT file (decoded with iasl) > > > Source code with changes made so far is available on github [8] > > Any hint is appreciated > > Thanks, > Paulo Sérgio Travaglia (pstglia) > > > [1] - https://groups.google.com/forum/#!forum/android-x86 > [2] - http://www.spinics.net/lists/linux-acpi/msg58680.html > [3] - https://goo.gl/vBKOeI > [4] - http://goo.gl/zriwbw > [5] - https://bugzilla.kernel.org/show_bug.cgi?id=86581 > [6] - http://thread.gmane.org/gmane.linux.alsa.devel/134554/focus=140634 > [7] > - https://drive.google.com/folderview?id=0BxO6THtB865fWjVHLTFfV25ORjQ&usp=sha… > [8] - https://github.com/pstglia/linux

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[alsa-devel] [PATCH 0/7] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 06 May '16

06 May '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since RFC v2: add BT function refine Kconfig and Makefile add prefix 'mt8173' and 'mt2701' for relative drivers. Garlic Tseng (7): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add clock and irq control for 2701 platform driver ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 35 +- sound/soc/mediatek/Makefile | 9 +- sound/soc/mediatek/mt2701/Makefile | 20 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 344 ++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 33 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 228 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1825 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 412 +++++ sound/soc/mediatek/mt2701/mt2701-irq.c | 74 + sound/soc/mediatek/mt2701/mt2701-irq.h | 26 + sound/soc/mediatek/mt2701/mt2701-reg.h | 195 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 101 ++ .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 488 +++--- sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 23 files changed, 3740 insertions(+), 368 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-irq.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-irq.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (67%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

3 13

[alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Kailang 06 May '16

06 May '16

Hi Takashi, We have new codecs for ALC234/ALC274/ALC294. Attach file is the support patch. BR, Kailang

1 0

[alsa-devel] Errors iterating over PCM stream hints and opening based on NAME/IOID
by Trent Reed 06 May '16

06 May '16

Hey All, I'm trying to understand ALSA, having a little trouble with some errors that I can't ration over. Gist Repro: https://gist.github.com/TReed0803/353ce27d37bc1ebe965854ea88c8b97b Whenever I run the above simple code (iterate over PCM stream hints, open for Playback/Capture depending on IOID; output also provided at link above) I notice that I sometimes hit errors. I'm having trouble understanding these errors, because as I understand I'm using the ALSA library correctly. For example, dmix is telling me that it "Only supports playback streams", well then, why does the hint IOID return that it supports capture as well? (same for dsnoop). More confusing are the "ALSA lib pcm_route.c:867:(find_matching_chmap) Found no matching channel map" errors. Can someone help me understand these issues, and perhaps what I'm doing wrong here? Would this point to something wrong with an installed package which perhaps configures these hints incorrectly? (Assuming that is probably the case, but I'd like to know for sure). I'm on a fairly new install of Ubuntu 16.04 LTS. Thanks, - Trent Reed

1 0

[alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6
by Takashi Sakamoto 06 May '16

06 May '16

Hi all, In this developing cycle for Linux 4.7, I concentrate on working to check packet sequence of ALSA firewire stack, because I ensure that the stack causes Dice-based models to generate sound including periodical noise or to generate no sound in the worst case. As of the end of developing cycle for Linux 4.6, I assumed that this is a bug of ALSA firewire stack[1]. However, the assumption is denied by my analysis with added tracepoints of Linux tracing framework[2]. Then, I concluded that the issue comes from quirks of Dice in an aspect of processing presentation timestamp. I'm under further investigation and have seen some of the quirks. In this message, I describe the summary of superficial issues from the quirks, the status of ALSA firewire stack in Linux 4.6 for Dice-based models and direction of my work for ALSA firewire stack in this developing cycle. Besides, resources and my memo for this work is available in this URL. I wish to upload it to alsa-project.org, not in my server: http://sakamocchi.jp/media/20160505-dice-packet-dump.tar.xz (233MB archive file compressed by xz, expands 6.7 GB text files.) Summary - Dice ASICs/firmwares/drivers communicate with packet sequences which is partly not compliant to IEC 61883-6. - Dice ASICs/firmwares/drivers doesn't sometimes transfer data blocks as the same as sampling transfer frequency. - Calculation of presentation timestamp is not based on duration of data blocks in a packet. This way is different from implementation of ALSA firewire stack. - The value of estimated 'transfer delay' is against IEEE 1394 spec. As a result: - ALSA dice driver causes these issues. ALSA dice driver in Linux kernels since 3.13 have included this issue. - To fix this issue, ALSA firewire stack require timestamp generator and data block calculator specific for Dice-based models. At least, three functions should be changed; amdtp_stream_start(), calculate_data_blocks(), calculate_syt() in 'sound/firewire/amdtp-stream.c'. - The work requires much changes, thus it's impossible for Linux 4.6. ALSA firewire stack in Linux 4.6 will be shipped with these issues. - We have a short time till merge window of Linux 4.7 (Just one week). I'll work for merging my proposed patchset[2][3][4], without fixing the issues. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105608.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106950.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107290.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107531.html Regards Takashi Sakamoto

1 0

[alsa-devel] I-MX6 & I2s output
by nick83ola 05 May '16

05 May '16

Hello to All I have an IMX6 board and I need a simple way to capture/put an I2s master signal I'm working with yocto and kernel version 3.14.28-1.0.0 I have on my development board an sgtl5000 codec but it work only with I2S slave mode I've found the driver fsl-asoc-card.c from more recent kernel and I've modified it But no matter what I do i cannot set I2S to master mode Please Help -- P.S. Le informazioni trasmesse attraverso la presente comunicazione sono di esclusiva spettanza dell'effettivo destinatario. Nel caso in cui le stesse raggiungessero, per qualunque motivo, soggetti non interessati, questi ultimi vorranno darne immediata notizia al mittente. In ogni caso, eventuali soggetti diversi dai legittimi destinatari della presente comunicazione e dei dati contenuti negli allegati, possono essere sanzionati ai sensi del T.U. sul trattamento dei dati personali d.lgs. 196/2003, sia ai sensi dell'art. 616 del Codice Penale che disciplina la violazione del segreto sulla corrispondenza.

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Move compr_attach/attached functions
by Charles Keepax 05 May '16

05 May '16

Move wm_adsp_compr_attach and wm_adsp_compr_attached functions so they will stay logically grouped with similar functions after some additional changes. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 630ebcd..42fc469 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2452,6 +2452,25 @@ void wm_adsp2_remove(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_remove); +static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) +{ + return compr->buf != NULL; +} + +static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) +{ + /* + * Note this will be more complex once each DSP can support multiple + * streams + */ + if (!compr->dsp->buffer) + return -EINVAL; + + compr->buf = compr->dsp->buffer; + + return 0; +} + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; @@ -2810,25 +2829,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp) return 0; } -static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) -{ - return compr->buf != NULL; -} - -static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) -{ - /* - * Note this will be more complex once each DSP can support multiple - * streams - */ - if (!compr->dsp->buffer) - return -EINVAL; - - compr->buf = compr->dsp->buffer; - - return 0; -} - int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) { struct wm_adsp_compr *compr = stream->runtime->private_data; -- 2.1.4

2 3

[alsa-devel] [PATCH 1/2] Revert "ASoC: rockchip: i2s: remove unused variables"
by John Keeping 05 May '16

05 May '16

This reverts commit 5938448b99275cba95167c3f9d39ca9225fdad38. It turns out that the commit that made these variables unused is wrong so we're about to revert it. Bring back the variables in prepration. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/rockchip_i2s.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e20416bd3..34743ec61c49 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -77,7 +84,11 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START, I2S_XFER_TXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); @@ -115,7 +126,11 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_RXS_START, I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); -- 2.8.0.rc4.238.g874082a

3 5

[alsa-devel] [PATCH] ASoC: rt5645: polling jd status in all conditions
by Bard Liao 05 May '16

05 May '16

We only polling jd status when rt5645->pdata.jd_invert is true. However, it should be done at all time since there will be no interrupt for jd if we press a headset button and remove the headset at the same time. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index dff706a..3c6594d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (btn_type == 0)/* button release */ report = rt5645->jack_type; else { - if (rt5645->pdata.jd_invert) { - mod_timer(&rt5645->btn_check_timer, - msecs_to_jiffies(100)); - } + mod_timer(&rt5645->btn_check_timer, + msecs_to_jiffies(100)); } break; @@ -3816,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); - setup_timer(&rt5645->btn_check_timer, - rt5645_btn_check_callback, (unsigned long)rt5645); } + setup_timer(&rt5645->btn_check_timer, + rt5645_btn_check_callback, (unsigned long)rt5645); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3 1/2] ASoC: fsl_sai: Introduce a compatible string for MX6UL
by Fabio Estevam 05 May '16

05 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> MX6UL may need to configure the General Purpose Register 1 (GPR1), so it is better to add a new compatible string to differentiate. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v2: - Add binding doc entry for mx6ul (Nicolin) Documentation/devicetree/bindings/sound/fsl-sai.txt | 4 ++-- sound/soc/fsl/fsl_sai.c | 4 +++- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 044e5d7..777b941 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -7,8 +7,8 @@ codec/DSP interfaces. Required properties: - - compatible : Compatible list, contains "fsl,vf610-sai" or - "fsl,imx6sx-sai". + - compatible : Compatible list, contains "fsl,vf610-sai", + "fsl,imx6sx-sai" or "fsl,imx6ul-sai" - reg : Offset and length of the register set for the device. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..d8b673f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- 1.9.1

3 5

[alsa-devel] [PATCH 0/2] ASoC: Memeory leak fixes
by Vinod Koul 05 May '16

05 May '16

This series fixes two memeory leaks, first in topology core to freeup the objects in widget creation and second one on Skylake driver for NHLT table cleanup Jeeja KP (2): ASoC: topology: Fix memory leak in widget creation ASoC: Intel: Skylake: Fix memory leak in nhlt init sound/soc/intel/skylake/skl-nhlt.c | 15 +++++++++------ sound/soc/intel/skylake/skl.c | 5 ++++- sound/soc/intel/skylake/skl.h | 6 +++--- sound/soc/soc-topology.c | 2 ++ 4 files changed, 18 insertions(+), 10 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v2 1/2] ASoC: fsl_sai: Introduce a compatible string for MX6UL
by Fabio Estevam 05 May '16

05 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> MX6UL may need to configure the General Purpose Register 1 (GPR1), so it is better to add a new compatible string to differentiate. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Newly introduced in this series. sound/soc/fsl/fsl_sai.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..d8b673f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: fsl_sai: Allow the SAI driver to turn on SAI_MCLK
by Fabio Estevam 04 May '16

04 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> On mx6ul the General Purpose Register 1 (GPR1) contains the following bits for enabling the output of the SAI MCLKs: SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR Introduce "gpr" and "fsl,sai-enable-mclk" optional properties to allow the enablement of the SAI_MCLK outputs. Tested on a imx6ul-evk board. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- .../devicetree/bindings/sound/fsl-sai.txt | 8 +++++ include/linux/mfd/syscon/imx6q-iomuxc-gpr.h | 3 ++ sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++ 3 files changed, 48 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 044e5d7..86755bb 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -48,6 +48,14 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties (for mx6ul): + + - gpr : The phandle to the General Purpose Register (GPR) + node. + + - fsl,sai-enable-mclk : This is a boolean property. If present, indicates + that SAI will output the SAI MCLK clock. + Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h index 238c8db..401f97e 100644 --- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h +++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h @@ -447,5 +447,8 @@ #define IMX6UL_GPR1_ENET2_CLK_OUTPUT (0x1 << 18) #define IMX6UL_GPR1_ENET_CLK_DIR (0x3 << 17) #define IMX6UL_GPR1_ENET_CLK_OUTPUT (0x3 << 17) +#define IMX6UL_GPR1_SAI1_MCLK_DIR (0x1 << 19) +#define IMX6UL_GPR1_SAI2_MCLK_DIR (0x1 << 20) +#define IMX6UL_GPR1_SAI3_MCLK_DIR (0x1 << 21) #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */ diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..311325e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> #include "fsl_sai.h" #include "imx-pcm.h" @@ -785,11 +787,14 @@ static const struct regmap_config fsl_sai_regmap_config = { static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct device_node *gpr_np = of_parse_phandle(np, "gpr", 0); struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + u32 index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -877,6 +882,38 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-enable-mclk", NULL)) { + ret = of_property_read_u32(np, "sai-index", &index); + if (ret) { + dev_err(&pdev->dev, "could not read sai-index\n"); + return ret; + } + + gpr = syscon_node_to_regmap(gpr_np); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "could not find gpr node\n"); + return PTR_ERR(gpr); + } + + switch (index) { + case 1: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI1_MCLK_DIR, + IMX6UL_GPR1_SAI1_MCLK_DIR); + break; + case 2: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI2_MCLK_DIR, + IMX6UL_GPR1_SAI2_MCLK_DIR); + break; + case 3: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI3_MCLK_DIR, + IMX6UL_GPR1_SAI3_MCLK_DIR); + break; + } + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; -- 1.9.1

2 8

[alsa-devel] [PATCH 0/2 v6] Add I2S audio support for ARC AXS10x boards
by Jose Abreu 04 May '16

04 May '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds I2S audio for the AXS10x platform. Changes v5 -> v6 * Use SNDRV_DMA_TYPE_CONTINUOUS Changes v4 -> v5 * Resolve undefined references when compiling as module * Dropped adv7511 audio patches * Use DMA properties in I2S to check which mode to use: PIO or DMA (as suggested by Lars-Peter Clausen) Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh(a)kernel.org> Cc: Alexey Brodkin <abrodkin(a)synopsys.com> Cc: linux-snps-arc(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Jose Abreu (2): ASoC: dwc: Add custom PCM driver ASoC: dwc: Update DOCUMENTATION for I2S Driver .../devicetree/bindings/sound/designware-i2s.txt | 9 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware.h | 71 +++++++ sound/soc/dwc/designware_i2s.c | 94 ++++++--- sound/soc/dwc/designware_pcm.c | 228 +++++++++++++++++++++ 6 files changed, 383 insertions(+), 29 deletions(-) create mode 100644 sound/soc/dwc/designware.h create mode 100644 sound/soc/dwc/designware_pcm.c -- 1.9.1

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[alsa-devel] [PATCH v2 1/2] ASoC: rockchip: Revert "ASoC: rockchip: i2s: remove unused variables"
by John Keeping 04 May '16

04 May '16

This reverts commit 5938448b99275cba95167c3f9d39ca9225fdad38. It turns out that the commit that made these variables unused is wrong so we're about to revert it. Bring back the variables in prepration. Signed-off-by: John Keeping <john(a)metanate.com> --- v2: Add "ASoC: rockchip: " prefix to subject sound/soc/rockchip/rockchip_i2s.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e20416bd3..34743ec61c49 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -77,7 +84,11 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START, I2S_XFER_TXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); @@ -115,7 +126,11 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_RXS_START, I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); -- 2.8.0.rc4.238.g874082a

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[alsa-devel] [PATCH v6 0/4] Propagate errors out from compressed streams
by Charles Keepax 04 May '16

04 May '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v5: - Add a debug print to note which state we transition to in snd_compr_stop_error It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 81 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 +++++- sound/soc/soc-compress.c | 5 ++- 4 files changed, 93 insertions(+), 9 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH v2 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 04 May '16

04 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 68 +++++++++++++++++++++++++++++++++++++--------- sound/soc/soc-pcm.c | 3 +- 3 files changed, 61 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..fe5bc93 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..075f0d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,13 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); + if (custom_stop_condition) { + con = custom_stop_condition(widget, dir); + if (con >= 0) + return con; + con = 0; + } + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1117,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1130,41 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1172,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1211,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1310,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1970,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

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[alsa-devel] Get dB value using softvol plugin
by Jörg Krause 04 May '16

04 May '16

Hi, I configured my softvol plugin with a min_dB value of -6.0. However, if I store the state with alsactl store, the comment section shows a dbmin value of 0. I figured out that the softvol plugin adds the tlv info when it adds the user ctl and stores the db values in snd_pcm_softvol_t. Why differ the values of the tlv and the values of snd_pcm_softvol_t? My goal is to get information of the current dB value used by the softvol plugin, e.g. -6.0dB for a raw volume value of 255 and -51.0dB for 0. However, I only get values from +0 to +51dB using snd_ctl_convert_to_dB(). Do I miss something? Note, that I started from a clean asound.state. --- The state produced by alsactl: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } --- The output of aplay -D softvol -v: Playing WAVE '/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Soft volume PCM Control: Master min_dB: -57 max_dB: -6 resolution: 256 --- My asound.conf: pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "hw:0" } control { name "Master" card 0 } min_dB -57.0 max_dB -6.0 } ctl.!default { type hw card 0 } ctl.softvol { type hw card 0 } --- Best regards Jörg Krause

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Re: [alsa-devel] [Alsa-user] How do I run arecord on raw digital mic buffer?
by Clemens Ladisch 04 May '16

04 May '16

Rob Nertney wrote: >>> is the best method to build a sound card and a capture PCM? >> >> As compared to what? Any other method will not allow your hardware to >> be used for audio recording. > > As compared to my driver configuring the DMA to capture the data into > a known buffer, and then directly accessing the buffer to wrap it in > a WAV format. I should have rephrased my initial question as, "my goal > is to simply wrap multichannel digital mic input in WAV format; is > ALSA/arecord the most efficient way to do this?" If you bypass ALSA, you have to write all the software yourself, and you will not be able to use your hardware for anything else. > I should make my buffer at least 2x the size of a period. Yes. (You don't need periods if you don't use interrupts to wait for data being available.) > I have a buffer ready, under which function does it get passed back to > the ALSA framework? Normally, ALSA knows about the buffer, and the driver just reports the current position. > * dma completes/fires period-based IRQ > * <<question>> buffer is transferred in function X (?) to Y (?) When a period is finished, you must call snd_pcm_period_elapsed(). ALSA will then call your .pointer callback to know where in the buffer the hardware has finished writing. (Please not that .pointer can also be called at other times.) Regards, Clemens

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[alsa-devel] [patch] ASoC: hdac_hdmi: Potential NULL deref in hdac_hdmi_get_spk_alloc()
by Dan Carpenter 03 May '16

03 May '16

We intended || here instead of &&. The original code potentially leads to a NULL dereference. Fixes: 2889099eb8cd ('ASoC: hdac_hdmi: Register chmap controls and ops') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3043120..7e7ff1b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1609,7 +1609,7 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); struct hdac_hdmi_pin *pin = pcm->pin; - if (!pin && !pin->eld.eld_valid) + if (!pin || !pin->eld.eld_valid) return 0; return pin->eld.info.spk_alloc;

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Re: [alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by sugar 03 May '16

03 May '16

Hi John, On 4/30/2016 15:00, John Keeping Wrote: > Hi Enric, > > On Fri, Apr 29, 2016 at 04:59:27PM +0200, Enric Balletbo Serra wrote: >> 2015-12-09 11:32 GMT+01:00 John Keeping <john(a)metanate.com>: >>> If we only clear the tx/rx state when both are disabled it is not >>> possible to start/stop one multiple times while the other is running. >>> Since the two are independently controlled, treat them as such and >>> remove the false dependency between capture and playback. >>> >>> Signed-off-by: John Keeping <john(a)metanate.com> >>> --- >>> sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- >>> 1 file changed, 32 insertions(+), 40 deletions(-) >>> >>> diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c >>> index 83b1b9c..acc6225 100644 >>> --- a/sound/soc/rockchip/rockchip_i2s.c >>> +++ b/sound/soc/rockchip/rockchip_i2s.c >>> @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >>> I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); >>> >>> regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >>> + I2S_XFER_TXS_START, >>> + I2S_XFER_TXS_START); >>> >>> i2s->tx_start = true; >>> } else { >>> @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >>> regmap_update_bits(i2s->regmap, I2S_DMACR, >>> I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); >>> >>> - if (!i2s->rx_start) { >>> - regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | >>> - I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_STOP | >>> - I2S_XFER_RXS_STOP); >>> + regmap_update_bits(i2s->regmap, I2S_XFER, >>> + I2S_XFER_TXS_START, >>> + I2S_XFER_TXS_STOP); >>> >>> - regmap_update_bits(i2s->regmap, I2S_CLR, >>> - I2S_CLR_TXC | I2S_CLR_RXC, >>> - I2S_CLR_TXC | I2S_CLR_RXC); >>> + regmap_update_bits(i2s->regmap, I2S_CLR, >>> + I2S_CLR_TXC, >>> + I2S_CLR_TXC); >>> >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> >>> - /* Should wait for clear operation to finish */ >>> - while (val) { >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> - retry--; >>> - if (!retry) { >>> - dev_warn(i2s->dev, "fail to clear\n"); >>> - break; >>> - } >>> + /* Should wait for clear operation to finish */ >>> + while (val & I2S_CLR_TXC) { >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> + retry--; >>> + if (!retry) { >>> + dev_warn(i2s->dev, "fail to clear\n"); >>> + break; >>> } >>> } >>> } >>> @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >>> I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); >>> >>> regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >>> + I2S_XFER_RXS_START, >>> + I2S_XFER_RXS_START); >>> >>> i2s->rx_start = true; >>> } else { >>> @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >>> regmap_update_bits(i2s->regmap, I2S_DMACR, >>> I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); >>> >>> - if (!i2s->tx_start) { >>> - regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | >>> - I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_STOP | >>> - I2S_XFER_RXS_STOP); >>> + regmap_update_bits(i2s->regmap, I2S_XFER, >>> + I2S_XFER_RXS_START, >>> + I2S_XFER_RXS_STOP); >>> >>> - regmap_update_bits(i2s->regmap, I2S_CLR, >>> - I2S_CLR_TXC | I2S_CLR_RXC, >>> - I2S_CLR_TXC | I2S_CLR_RXC); >>> + regmap_update_bits(i2s->regmap, I2S_CLR, >>> + I2S_CLR_RXC, >>> + I2S_CLR_RXC); >>> >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> >>> - /* Should wait for clear operation to finish */ >>> - while (val) { >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> - retry--; >>> - if (!retry) { >>> - dev_warn(i2s->dev, "fail to clear\n"); >>> - break; >>> - } >>> + /* Should wait for clear operation to finish */ >>> + while (val & I2S_CLR_RXC) { >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> + retry--; >>> + if (!retry) { >>> + dev_warn(i2s->dev, "fail to clear\n"); >>> + break; >>> } >>> } >>> } >>> -- >>> 2.6.3.462.gbe2c914 >> >> Using my Veyron Jerry Chromebook wiht latest kernel I found that the >> speakers doesn't work. Bisect points to this patch as the offending >> commit. However your changes looks reasonable the fact is that >> reverting the patch makes the audio work again on my device. I need to >> dig a bit more into the issue, but meanwhile, any idea on what is >> happening ? Can I ask which device did you test? > > I tested this on a Radxa Rock2 square. This change fixed an issue I > found when running separate processes for playback and recording (I was > testing with arecord and speaker-test IIRC) but I don't think that > should be relevant. I just read through the patch again and I can't see > anything obviously wrong so I'm afraid I don't have any idea what's > causing the problem you're seeing. > Do you have test playback/capture at the same time? I think you may reproduce this issue if you do as follows: step 1: tinyplay test.wav -D 0 -d 0 -p 1024 -n 4 & step 2: tinycap r.wav -D 0 -d 0 -p 1024 -n 4 & you will find that there is no music now. because of this problem, we enable tx/rx at the same time to WA this issue. May I know what's the issue that you met before? > > John > > _______________________________________________ > Linux-rockchip mailing list > Linux-rockchip(a)lists.infradead.org > http://lists.infradead.org/mailman/listinfo/linux-rockchip > > >

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[alsa-devel] [PATCH v3 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 03 May '16

03 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 81 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 70 insertions(+), 11 deletions(-) -- 1.9.1

6 20

[alsa-devel] iMX6UL EVK with kernel 4.6
by Petr Kulhavy 02 May '16

02 May '16

Hi, I'm trying to run the latest mainline kernel 4.6 on the iMX6UL EVK. The board mostly works and the only problem is audio. Even though the sound card is detected and the ALSA mixer can be controlled there is no MCLK coming out of the SAI2 interface. All the related peripherals in the DTS (pinmux, SAI2, sound card, codec) seem to be set-up correctly, I also tried out various things like disabling the ASRC, but without success and I'm slowly running out of ideas. Could somebody give me a hint please, what is different to the Freescale kernel branch where the MCLK works? By the way, why aren't the Freescale patches merged to the mainline kernel? Thanks Petr

2 14

Re: [alsa-devel] bytcr_rt5640.c:(.text+0x1593b4): undefined reference to `sst_acpi_find_name_from_hid'
by Randy Dunlap 02 May '16

02 May '16

On 03/29/16 07:21, Borislav Petkov wrote: > Hi guys, > > I'm seeing a bunch of those errors today on randconfig builds: > > sound/built-in.o: In function `snd_byt_rt5640_mc_probe': > bytcr_rt5640.c:(.text+0x1593b4): undefined reference to `sst_acpi_find_name_from_hid' > make: *** [vmlinux] Error 1 > > ERROR: "sst_acpi_find_name_from_hid" [sound/soc/intel/boards/snd-soc-sst-bytcr-rt5640.ko] undefined! > make[1]: *** [__modpost] Error 1 > make: *** [modules] Error 2 > make: *** Waiting for unfinished jobs.... > > Some of the failing configs attached. > I have 2 patches for this. I'll send them right away. -- ~Randy

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Add support for TLV based binary controls
by Charles Keepax 02 May '16

02 May '16

This patch adds support for the arbitrary length TLV based binary controls. This allows users to properly access controls that are more than 512 bytes in length. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 128 ++++++++++++++++++++++++++++++++++++++------- 1 file changed, 108 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5699a81..630ebcd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -160,6 +160,8 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +#define ADSP_MAX_STD_CTRL_SIZE 512 + struct wm_adsp_buf { struct list_head list; void *buf; @@ -435,6 +437,7 @@ struct wm_coeff_ctl { size_t len; unsigned int set:1; struct snd_kcontrol *kcontrol; + struct soc_bytes_ext bytes_ext; unsigned int flags; }; @@ -711,10 +714,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } +static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) +{ + return container_of(ext, struct wm_coeff_ctl, bytes_ext); +} + static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -763,7 +773,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -780,6 +792,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, + const unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (copy_from_user(ctl->cache, bytes, size)) { + ret = -EFAULT; + } else { + ctl->set = 1; + if (ctl->enabled) + ret = wm_coeff_write_control(ctl, ctl->cache, size); + } + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, void *buf, size_t len) { @@ -822,7 +857,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -845,12 +882,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, + unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { + if (ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + else + ret = -EPERM; + } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + } + + if (!ret && copy_to_user(bytes, ctl->cache, size)) + ret = -EFAULT; + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + struct wmfw_ctl_work { struct wm_adsp *dsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; +static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) +{ + unsigned int out, rd, wr, vol; + + if (len > ADSP_MAX_STD_CTRL_SIZE) { + rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ; + wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } else { + rd = SNDRV_CTL_ELEM_ACCESS_READ; + wr = SNDRV_CTL_ELEM_ACCESS_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = 0; + } + + if (in) { + if (in & WMFW_CTL_FLAG_READABLE) + out |= rd; + if (in & WMFW_CTL_FLAG_WRITEABLE) + out |= wr; + if (in & WMFW_CTL_FLAG_VOLATILE) + out |= vol; + } else { + out |= rd | wr | vol; + } + + return out; +} + static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; @@ -868,19 +965,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; - kcontrol->private_value = (unsigned long)ctl; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kcontrol->tlv.c = snd_soc_bytes_tlv_callback; + kcontrol->private_value = (unsigned long)&ctl->bytes_ext; - if (ctl->flags) { - if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE; - if (ctl->flags & WMFW_CTL_FLAG_READABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ; - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } else { - kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE; - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + + kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len); ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1); if (ret < 0) @@ -1039,11 +1132,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->flags = flags; ctl->offset = offset; - if (len > 512) { - adsp_warn(dsp, "Truncating control %s from %d\n", - ctl->name, len); - len = 512; - } ctl->len = len; ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { -- 2.1.4

2 1

[alsa-devel] Volume control with mute in the wrong end
by Peter Rosin 02 May '16

02 May '16

Hi! I'm writing a driver for the max9860 codec https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf and it has a digital sidetone control where it can mix in the left channel ADC into the (mono) DAC output. The register that controls the sidetone level makes me wonder how I should wire up DAPM for it. You set the gain for the sidetone in the same register field as you enable it. I have a problem though, since the mute value is at the wrong and of the scale, i.e. you turn the volume up and up and up, and when you reach "max" it is off. I did try to declare a TLV range like so: /* The dvst field has its mute in the wrong end. Sigh. */ static const DECLARE_TLV_DB_RANGE(dvst_tlv, 0, 30, TLV_DB_SCALE_ITEM(-6000, 200, 0), 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); and then hook it that up with a dapm mixer: static const struct snd_kcontrol_new max9860_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("Sidetone Volume", MAX9860_DACGAIN, MAX9860_DVST_SHIFT, 31, 1, dvst_tlv), }; static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 0, max9860_mixer_controls, ARRAY_SIZE(max9860_mixer_controls)), SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), SND_SOC_DAPM_OUTPUT("OUT"), }; static const struct snd_soc_dapm_route max9860_dapm_routes[] = { { "ADCL", NULL, "MICL" }, { "ADCR", NULL, "MICR" }, { "AIFOUTL", NULL, "ADCL" }, { "AIFOUTR", NULL, "ADCR" }, { "OUT", NULL, "DAC" }, { "DAC", NULL, "Mixer" }, { "Mixer", "Sidetone Volume", "ADCL" }, { "Mixer", NULL, "AIFINL" }, { "Mixer", NULL, "AIFINR" }, }; But DAPM and some tools (e.g. alsamixer) do not properly recognize the mute value at the wrong end on the scale for the sidetone. So, the sidetone path is never off (in the dapm sense). I can use amixer to set the special mute value, and it states -99999.99dB, so I think the tlv declaration is correct, at least in theory. But as I said alsamixer is thoroughly confused by the control and behaves rather strange... If I change the tlv declaration so that DAPM thinks the mute is in the natural end of the scale, the paths are off in the dapm sense when the sidetone is muted (but the sidetone is not in fact muted, of course). Is there anything I can do to make DAPM handle this insane register field better? Cheers, Peter

1 0

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix overwriting of ahclkx
by J.D. Schroeder 02 May '16

02 May '16

From: Jim Lodes <jim.lodes(a)garmin.com> The mcasp davinci_mcasp_set_dai_fmt function was overriding ahclkx input/output status that had already been set by the davinci_mcasp_set_sysclk function. This commit removes clearing of the ahclkx input/output status from davinci_mcasp_set_dai_fmt. Signed-off-by: Jim Lodes <jim.lodes(a)garmin.com> Signed-off-by: J.D. Schroeder <jay.schroeder(a)garmin.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e132498..a1197ad 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + ACLKX | AFSX | ACLKR | AHCLKR | AFSR); mcasp->bclk_master = 0; break; default: -- 1.9.1

3 2

[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake: Wov optimizations
by Vinod Koul 02 May '16

02 May '16

This series starts with two minor fixes on driver to fix a memory leak and one NULL pointer exception and then adds power optimization to driver for WOV case by suspending the stream DMA and keeping only DSP pipe and endpoint On during that case Dharageswari.R (2): ASoC: Intel: Skylake: Fix the NULL pointer exception in dsp_clean up ASoC: Intel: Skylake: Prevent sending Set DMA Control IPC if the widget is "On" Jeeja KP (2): ASoC: Intel: Skylake: Suspend PCMs when marked as active suspend ASoC: Intel: Boards: remove ignore_suspend for WoV streams Mousumi Jana (1): ASoC: Intel: Skylake: Fix memory leak during init instance sound/soc/intel/boards/skl_nau88l25_max98357a.c | 1 - sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 1 - sound/soc/intel/boards/skl_rt286.c | 1 - sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 32 ++++++++++++++++--------- sound/soc/intel/skylake/skl-sst-dsp.c | 2 -- sound/soc/intel/skylake/skl-sst.c | 4 ++++ 7 files changed, 26 insertions(+), 17 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by Enric Balletbo Serra 02 May '16

02 May '16

Hi John, Thanks for answer. 2016-04-30 9:00 GMT+02:00 John Keeping <john(a)keeping.me.uk>: > Hi Enric, > > On Fri, Apr 29, 2016 at 04:59:27PM +0200, Enric Balletbo Serra wrote: >> 2015-12-09 11:32 GMT+01:00 John Keeping <john(a)metanate.com>: >> > If we only clear the tx/rx state when both are disabled it is not >> > possible to start/stop one multiple times while the other is running. >> > Since the two are independently controlled, treat them as such and >> > remove the false dependency between capture and playback. >> > >> > Signed-off-by: John Keeping <john(a)metanate.com> >> > --- >> > sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- >> > 1 file changed, 32 insertions(+), 40 deletions(-) >> > >> > diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c >> > index 83b1b9c..acc6225 100644 >> > --- a/sound/soc/rockchip/rockchip_i2s.c >> > +++ b/sound/soc/rockchip/rockchip_i2s.c >> > @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >> > I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); >> > >> > regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >> > + I2S_XFER_TXS_START, >> > + I2S_XFER_TXS_START); >> > >> > i2s->tx_start = true; >> > } else { >> > @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >> > regmap_update_bits(i2s->regmap, I2S_DMACR, >> > I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); >> > >> > - if (!i2s->rx_start) { >> > - regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | >> > - I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_STOP | >> > - I2S_XFER_RXS_STOP); >> > + regmap_update_bits(i2s->regmap, I2S_XFER, >> > + I2S_XFER_TXS_START, >> > + I2S_XFER_TXS_STOP); >> > >> > - regmap_update_bits(i2s->regmap, I2S_CLR, >> > - I2S_CLR_TXC | I2S_CLR_RXC, >> > - I2S_CLR_TXC | I2S_CLR_RXC); >> > + regmap_update_bits(i2s->regmap, I2S_CLR, >> > + I2S_CLR_TXC, >> > + I2S_CLR_TXC); >> > >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > >> > - /* Should wait for clear operation to finish */ >> > - while (val) { >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > - retry--; >> > - if (!retry) { >> > - dev_warn(i2s->dev, "fail to clear\n"); >> > - break; >> > - } >> > + /* Should wait for clear operation to finish */ >> > + while (val & I2S_CLR_TXC) { >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > + retry--; >> > + if (!retry) { >> > + dev_warn(i2s->dev, "fail to clear\n"); >> > + break; >> > } >> > } >> > } >> > @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >> > I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); >> > >> > regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >> > + I2S_XFER_RXS_START, >> > + I2S_XFER_RXS_START); >> > >> > i2s->rx_start = true; >> > } else { >> > @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >> > regmap_update_bits(i2s->regmap, I2S_DMACR, >> > I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); >> > >> > - if (!i2s->tx_start) { >> > - regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | >> > - I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_STOP | >> > - I2S_XFER_RXS_STOP); >> > + regmap_update_bits(i2s->regmap, I2S_XFER, >> > + I2S_XFER_RXS_START, >> > + I2S_XFER_RXS_STOP); >> > >> > - regmap_update_bits(i2s->regmap, I2S_CLR, >> > - I2S_CLR_TXC | I2S_CLR_RXC, >> > - I2S_CLR_TXC | I2S_CLR_RXC); >> > + regmap_update_bits(i2s->regmap, I2S_CLR, >> > + I2S_CLR_RXC, >> > + I2S_CLR_RXC); >> > >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > >> > - /* Should wait for clear operation to finish */ >> > - while (val) { >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > - retry--; >> > - if (!retry) { >> > - dev_warn(i2s->dev, "fail to clear\n"); >> > - break; >> > - } >> > + /* Should wait for clear operation to finish */ >> > + while (val & I2S_CLR_RXC) { >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > + retry--; >> > + if (!retry) { >> > + dev_warn(i2s->dev, "fail to clear\n"); >> > + break; >> > } >> > } >> > } >> > -- >> > 2.6.3.462.gbe2c914 >> >> Using my Veyron Jerry Chromebook wiht latest kernel I found that the >> speakers doesn't work. Bisect points to this patch as the offending >> commit. However your changes looks reasonable the fact is that >> reverting the patch makes the audio work again on my device. I need to >> dig a bit more into the issue, but meanwhile, any idea on what is >> happening ? Can I ask which device did you test? > > I tested this on a Radxa Rock2 square. This change fixed an issue I > found when running separate processes for playback and recording (I was > testing with arecord and speaker-test IIRC) but I don't think that > should be relevant. I just read through the patch again and I can't see > anything obviously wrong so I'm afraid I don't have any idea what's > causing the problem you're seeing. > By chance, do you have a git repo containing the changes required to test the sound on Radxa Rock2? Guess this is not in mainline. Thanks, Enric > > John

1 0

[alsa-devel] [PATCH 0/2] ASoC: Intel: Skylake: Add HDA link management
by Vinod Koul 02 May '16

02 May '16

HDA links are always kept On, this patch adds link management routines in hdac extended core and it's use in Skylake driver. One more patch to add support in codec will follow this The aim of serial is to shut down the links when not in use and keep them active when a user is active. The user can activate link by taking reference to the link when required. When all links are shutdown, the command DMAs are turned off as well. On first link power up, the command DMAs are turned on first. First patch touches HDA core, and second one is dependent on it so Takashi if you are okay, pls ACK it and we cna merge thru ASoC tree Vinod Koul (2): ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link mangement include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/intel/skylake/skl.c | 34 ++++++++++++++++++++ 5 files changed, 112 insertions(+), 1 deletion(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v4 0/3]: conf: topology: Add the conf for SKL-rt286
by Vinod Koul 02 May '16

02 May '16

This series adds SKL-i2c conf for simple PCM topology graph Changes in v4: - add back the pre-compiled binary blobs - remove running the binary on intsall step Changes in v3: - remove bin blobs, now they are generated in install step - hook the tool compilation to alsa-lib and run it on install - add u8 in alsa type_compat.h Shreyas NC (3): conf: topology: Add Skylake i2s conf Add u8 in type_compat.h conf: topology: Generate Private data binary blobs configure.ac | 2 + include/sound/type_compat.h | 2 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 5 + src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/data/Makefile.am | 4 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_local.h | 9 + .../topology/sklrt286/data/skl-tplg-interface.h | 232 +++ .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 28 files changed, 2502 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin create mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_local.h create mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin create mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin create mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin create mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 1.9.1

2 5

[alsa-devel] [PATCH 0/4] Decouple X86_32 dependency from the ISA Kconfig option
by William Breathitt Gray 01 May '16

01 May '16

This patchset is based on top of commit 3a3a5fece6f2 ("fs: kernfs: Replace CURRENT_TIME by current_fs_time()") of the driver-core-testing branch of the driver-core repository. The introduction of the ISA_BUS option in commit b3c1be1b789c ("base: isa: Remove X86_32 dependency") blocks the compilation of ISA drivers on non-x86 platforms. The ISA_BUS configuration option should not be necessary if the X86_32 dependency can be decoupled from the ISA configuration option. This patchset both removes the ISA_BUS configuration option entirely and decouples the X86_32 dependency from the ISA configuration option. The PNPBIOS driver requires preprocessor defines (located in include/asm/segment.h) only declared if the architecture is set to X86_32. If the architecture is set to X86_64, the PNPBIOS driver will not build properly. The X86 dependecy for the PNPBIOS configuration option is changed to an explicit X86_32 dependency in order to prevent an attempt to build for an unsupported architecture. Changes to the ISA SSCAPE and SCSI ULTRASTOR drivers are also included. The relevant patches simply fix format string identifier mismatches exposed during an attempted X86_64 compilation after the decoupling of the X86_32 dependency from the ISA configuration option. These patches fix compilation warnings rather than errors, but the solutions were so trivial that I decided to include them in this patchset. If it would be inappropriate to include them in this patchset, let me know and I will rebase to remove the relevant patches. William Breathitt Gray (4): pnp: pnpbios: Add explicit X86_32 dependency to PNPBIOS sound: isa: sscape: Use correct format identifier for size_t scsi: ultrastor: Use correct format identifier for kernel pointer isa: Remove the ISA_BUS Kconfig option arch/x86/Kconfig | 10 ++-------- drivers/base/Makefile | 2 +- drivers/pnp/pnpbios/Kconfig | 2 +- drivers/scsi/ultrastor.c | 8 ++++---- include/linux/isa.h | 2 +- sound/isa/sscape.c | 2 +- 6 files changed, 10 insertions(+), 16 deletions(-) -- 2.7.3

5 15

[alsa-devel] strange link on alsa-project.org, possibly a spam link
by humbert.olivier.1＠free.fr 30 Apr '16

30 Apr '16

Hi all just to let you know that there is a strange link in http://www.alsa-project.org/main/index.php/Main_Page See this screenshot : http://pix.toile-libre.org/upload/original/1461883624.png This "strange link" is under the white pointing-down arrow on the blue background circle It is linking to a "cheap holiday" booking webpage and someone on alsa@freenode suggested it looks to be one of those 'domain not used' redirects which I do agree. This link has been introduced back in August 2007 the 10th following this change : http://www.alsa-project.org/main/index.php?title=Main_Page&diff=698&oldid=6… I thought you might want to know. All the best to you all and thanks for the work done on ALSA. Cheers, Olivier

2 2

[alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by John Keeping 29 Apr '16

29 Apr '16

If we only clear the tx/rx state when both are disabled it is not possible to start/stop one multiple times while the other is running. Since the two are independently controlled, treat them as such and remove the false dependency between capture and playback. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- 1 file changed, 32 insertions(+), 40 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 83b1b9c..acc6225 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_TXS_START, + I2S_XFER_TXS_START); i2s->tx_start = true; } else { @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START, + I2S_XFER_TXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC, + I2S_CLR_TXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_TXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_RXS_START, + I2S_XFER_RXS_START); i2s->rx_start = true; } else { @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_RXS_START, + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_RXC, + I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_RXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } -- 2.6.3.462.gbe2c914

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: Quirk for yet another Phoenix Audio devices (v2)
by Takashi Iwai 29 Apr '16

29 Apr '16

Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the same workaround as TMX320 for avoiding the firmware bug. It fixes the frequent error about the sample rate inquiries and the slow device probe as consequence. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0adfd9537cf7..7b6dca2d7488 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1138,7 +1138,9 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ + case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } -- 2.8.1

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Quirk for yet another Phoenix Audio device
by Takashi Iwai 29 Apr '16

29 Apr '16

Phoenix Audio MT202pcs (1de7:0114) needs the same workaround as TMX320 for avoiding the firmware bug. It fixes the frequent error about the sample rate inquiries and the slow device probe as consequence. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0adfd9537cf7..c07facfe6699 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1139,6 +1139,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ + case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } -- 2.8.1

1 1

[alsa-devel] [PATCH 1/1] ASoC: fsl_ssi: add CCSR_SSI_SOR to volatile register list
by Caleb Crome 29 Apr '16

29 Apr '16

Due to caching, SOR wasn't written when it should have been. This patch simply adds SOR to the volatile list. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 216e3cb..2f3bf9c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -151,6 +151,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) case CCSR_SSI_SACDAT: case CCSR_SSI_SATAG: case CCSR_SSI_SACCST: + case CCSR_SSI_SOR: return true; default: return false; -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: Limit retrying sample rate reads
by Takashi Iwai 29 Apr '16

29 Apr '16

There are many USB audio devices with buggy firmware that don't react with the sample rate reading properly. This often results in the flood of error messages and slowing down the operation. The sample rate read back is basically only for confirming the sample rate setup, and it's not critically important. As a compromise, in this patch, we stop the sample rate read back once when the device gives errors more than tolerance (twice, as of now). This should improve most of error cases while we still can catch the firmware bugginess. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 4 ++++ sound/usb/usbaudio.h | 1 + 2 files changed, 5 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7ccbcaf6a147..26dd5f20f149 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -309,6 +309,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, * support reading */ if (snd_usb_get_sample_rate_quirk(chip)) return 0; + /* the firmware is likely buggy, don't repeat to fail too many times */ + if (chip->sample_rate_read_error > 2) + return 0; if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, @@ -316,6 +319,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data, sizeof(data))) < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); + chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a161c7c1b126..89b6853030de 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -50,6 +50,7 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + int sample_rate_read_error; struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ -- 2.8.1

1 0

[alsa-devel] [PATCH v2 1/1] ASoC: fsl_ssi: add CCSR_SSI_SOR to volatile register list
by Caleb Crome 29 Apr '16

29 Apr '16

The CCSR_SSI_SOR is a register that clears the TX and/or the RX fifo on the i.MX SSI port. The fsl_ssi_trigger writes this register in order to clear the fifo at trigger time. However, since the CCSR_SSI_SOR register is not in the volatile list, the caching mechanism prevented the register write in the trigger function. This caused the fifo to not be cleared (because the value was unchanged from the last time the register was written), and thus causes the channels in both TDM or simple I2S mode to slip and be in the wrong time slots on SSI restart. This has gone unnoticed for so long because with simple stereo mode, the consequence is that left and right are swapped, which isn't that noticeable. However, it's catestrophic in some systems that require the channels to be in the right slots. Signed-off-by: Caleb Crome <caleb(a)crome.org> Suggested-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 216e3cb..2f3bf9c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -151,6 +151,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) case CCSR_SSI_SACDAT: case CCSR_SSI_SATAG: case CCSR_SSI_SACCST: + case CCSR_SSI_SOR: return true; default: return false; -- 1.9.1

3 2

[alsa-devel] sound on asus x205ta
by Michele Curti 28 Apr '16

28 Apr '16

On Thu, Mar 03, 2016 at 10:20:55AM -0600, Pierre-Louis Bossart wrote: > On 3/3/16 5:27 AM, Vinod Koul wrote: > >On Thu, Mar 03, 2016 at 12:11:10PM +0100, Michele Curti wrote: > >>Hi Vinod, > >> > >>>>[ 7.970937] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 > >>>>[ 7.970964] intel_sst_acpi 80860F28:00: fw returned err -16 > >>>Yes this is a problem, which FW are you using > > > >Have you tried the recent kernels, Pierre did a bunch of update and got many > >things working. > > > >As for timeout, that simply means we have send a message to DSP which failed > >to get a response. Perhpas the message had some bad element of FW had a > >bug/crashed... > > The firmware timeout is due to a change in the BIOS for BYT-CR devices that > impacts the IPC interrupt number, I have a set of patches that fix this on > Asus T100TAF but I still can't get any sound out. > Hi! So, it seems I got dsp interrupts after the timeout errors. I added dev_dbg(drv->dev, "Interrupt received\n"); to the intel_sst_interrupt_mrfld() function [zio@asus ~]$ dmesg | grep "Interrupt received" | wc -l 105 [zio@asus ~]$ cat /proc/interrupts | grep sst 1: 105 0 0 0 IO-APIC 29-fasteoi intel_sst_driver [zio@asus ~]$ dmesg | grep intel_sst [ 3.003961] intel_sst_acpi 80860F28:00: LPE base: 0x90a00000 size:0x200000 [ 3.003970] intel_sst_acpi 80860F28:00: IRAM base: 0x90ac0000 [ 3.004002] intel_sst_acpi 80860F28:00: DRAM base: 0x90b00000 [ 3.004013] intel_sst_acpi 80860F28:00: SHIM base: 0x90b40000 [ 3.004024] intel_sst_acpi 80860F28:00: Mailbox base: 0x90b44000 [ 3.004033] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [ 3.004260] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [ 3.004388] intel_sst_acpi 80860F28:00: Registered IRQ 0x1 [ 3.004396] intel_sst_acpi 80860F28:00: Requesting FW intel/fw_sst_0f28_ssp0.bin now... [ 3.004792] intel_sst_acpi 80860F28:00: LPE is already in RESET state, No action [ 3.024222] intel_sst_acpi 80860F28:00: Enter [ 3.024232] intel_sst_acpi 80860F28:00: Request Fw completed [ 3.024507] intel_sst_acpi 80860F28:00: copied fw to ffff88007a900000 [ 3.024514] intel_sst_acpi 80860F28:00: phys: 7a900000 [ 3.024933] intel_sst_acpi 80860F28:00: Enter [ 3.024943] intel_sst_acpi 80860F28:00: header sign=$SST\xffffff96\xffffffb4 [ 3.024949] intel_sst_acpi 80860F28:00: module sign $SST\xffffff82\xffffffb4 [ 3.024955] intel_sst_acpi 80860F28:00: module entrypoint 0x0 [ 4.905820] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 4.905833] intel_sst_acpi 80860F28:00: fw returned err -16 [ 5.905864] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 5.905891] intel_sst_acpi 80860F28:00: fw returned err -16 [ 6.905818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 6.905838] intel_sst_acpi 80860F28:00: fw returned err -16 [ 7.905823] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 7.905849] intel_sst_acpi 80860F28:00: fw returned err -16 [ 8.905820] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 8.905847] intel_sst_acpi 80860F28:00: fw returned err -16 [ 9.905830] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 9.905857] intel_sst_acpi 80860F28:00: fw returned err -16 [ 10.905828] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 10.905855] intel_sst_acpi 80860F28:00: fw returned err -16 [ 11.905826] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 11.905853] intel_sst_acpi 80860F28:00: fw returned err -16 [ 12.905828] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 12.905854] intel_sst_acpi 80860F28:00: fw returned err -16 [ 13.905819] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 13.905843] intel_sst_acpi 80860F28:00: fw returned err -16 [ 14.925811] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 14.925836] intel_sst_acpi 80860F28:00: fw returned err -16 [ 15.925823] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 15.925850] intel_sst_acpi 80860F28:00: fw returned err -16 [ 16.925819] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 16.925844] intel_sst_acpi 80860F28:00: fw returned err -16 [ 17.925818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 17.925841] intel_sst_acpi 80860F28:00: fw returned err -16 [ 18.925821] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 18.925845] intel_sst_acpi 80860F28:00: fw returned err -16 [ 19.925810] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 19.925834] intel_sst_acpi 80860F28:00: fw returned err -16 [ 20.925788] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.925809] intel_sst_acpi 80860F28:00: fw returned err -16 [ 21.925821] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 21.925846] intel_sst_acpi 80860F28:00: fw returned err -16 [ 22.920830] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 22.920856] intel_sst_acpi 80860F28:00: fw returned err -16 [ 23.884836] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 23.884859] intel_sst_acpi 80860F28:00: fw returned err -16 [ 24.853161] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 24.853174] intel_sst_acpi 80860F28:00: fw returned err -16 [ 25.825400] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 25.825427] intel_sst_acpi 80860F28:00: fw returned err -16 [ 26.800986] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 26.801008] intel_sst_acpi 80860F28:00: fw returned err -16 [ 27.779566] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 27.779592] intel_sst_acpi 80860F28:00: fw returned err -16 [ 28.760784] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 28.760809] intel_sst_acpi 80860F28:00: fw returned err -16 [ 29.744302] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 29.744328] intel_sst_acpi 80860F28:00: fw returned err -16 [ 30.729854] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 30.729881] intel_sst_acpi 80860F28:00: fw returned err -16 [ 31.717175] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 31.717202] intel_sst_acpi 80860F28:00: fw returned err -16 [ 32.706075] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 32.706102] intel_sst_acpi 80860F28:00: fw returned err -16 [ 33.696343] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 33.696367] intel_sst_acpi 80860F28:00: fw returned err -16 [ 34.687818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 34.687844] intel_sst_acpi 80860F28:00: fw returned err -16 [ 35.680357] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 35.680383] intel_sst_acpi 80860F28:00: fw returned err -16 [ 36.673818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 36.673844] intel_sst_acpi 80860F28:00: fw returned err -16 [ 37.668111] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 37.668135] intel_sst_acpi 80860F28:00: fw returned err -16 [ 38.663091] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 38.663118] intel_sst_acpi 80860F28:00: fw returned err -16 [ 39.658680] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 39.658691] intel_sst_acpi 80860F28:00: fw returned err -16 [ 40.654812] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 40.654824] intel_sst_acpi 80860F28:00: fw returned err -16 [ 41.651466] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 41.651479] intel_sst_acpi 80860F28:00: fw returned err -16 [ 42.648513] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 42.648525] intel_sst_acpi 80860F28:00: fw returned err -16 [ 43.645942] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 43.645952] intel_sst_acpi 80860F28:00: fw returned err -16 [ 45.052845] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 45.052873] intel_sst_acpi 80860F28:00: fw returned err -16 [ 46.050948] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 46.050960] intel_sst_acpi 80860F28:00: fw returned err -16 [ 47.049320] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 47.049331] intel_sst_acpi 80860F28:00: fw returned err -16 [ 48.047885] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 48.047896] intel_sst_acpi 80860F28:00: fw returned err -16 [ 49.046627] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 49.046638] intel_sst_acpi 80860F28:00: fw returned err -16 [ 50.045542] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 50.045553] intel_sst_acpi 80860F28:00: fw returned err -16 [ 51.044603] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 51.044632] intel_sst_acpi 80860F28:00: fw returned err -16 [ 52.043787] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 52.043815] intel_sst_acpi 80860F28:00: fw returned err -16 [ 53.043030] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 53.043056] intel_sst_acpi 80860F28:00: fw returned err -16 [ 54.042669] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 54.042696] intel_sst_acpi 80860F28:00: fw returned err -16 [ 55.045310] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 55.045337] intel_sst_acpi 80860F28:00: fw returned err -16 [ 56.047789] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 56.047815] intel_sst_acpi 80860F28:00: fw returned err -16 [ 57.050159] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 57.050185] intel_sst_acpi 80860F28:00: fw returned err -16 [ 58.052350] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 58.052362] intel_sst_acpi 80860F28:00: fw returned err -16 [ 59.054494] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 59.054521] intel_sst_acpi 80860F28:00: fw returned err -16 [ 60.056485] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 60.056511] intel_sst_acpi 80860F28:00: fw returned err -16 [ 61.058360] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 61.058385] intel_sst_acpi 80860F28:00: fw returned err -16 [ 62.060160] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 62.060189] intel_sst_acpi 80860F28:00: fw returned err -16 [ 63.061829] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 63.061840] intel_sst_acpi 80860F28:00: fw returned err -16 [ 64.063443] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 64.063469] intel_sst_acpi 80860F28:00: fw returned err -16 [ 65.064970] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 65.064997] intel_sst_acpi 80860F28:00: fw returned err -16 [ 66.066390] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 66.066403] intel_sst_acpi 80860F28:00: fw returned err -16 [ 67.067803] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 67.067829] intel_sst_acpi 80860F28:00: fw returned err -16 [ 68.069108] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 68.069134] intel_sst_acpi 80860F28:00: fw returned err -16 [ 69.070353] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 69.070380] intel_sst_acpi 80860F28:00: fw returned err -16 [ 70.071530] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 70.071553] intel_sst_acpi 80860F28:00: fw returned err -16 [ 71.072665] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 71.072692] intel_sst_acpi 80860F28:00: fw returned err -16 [ 72.077040] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 72.077050] intel_sst_acpi 80860F28:00: fw returned err -16 [ 73.078102] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 73.078128] intel_sst_acpi 80860F28:00: fw returned err -16 [ 74.079073] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 74.079099] intel_sst_acpi 80860F28:00: fw returned err -16 [ 75.080034] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 75.080060] intel_sst_acpi 80860F28:00: fw returned err -16 [ 76.080922] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 76.080949] intel_sst_acpi 80860F28:00: fw returned err -16 [ 77.081768] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 77.081794] intel_sst_acpi 80860F28:00: fw returned err -16 [ 78.082592] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 78.082617] intel_sst_acpi 80860F28:00: fw returned err -16 [ 79.083395] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 79.083422] intel_sst_acpi 80860F28:00: fw returned err -16 [ 80.084155] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 80.084181] intel_sst_acpi 80860F28:00: fw returned err -16 [ 81.084883] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 81.084910] intel_sst_acpi 80860F28:00: fw returned err -16 [ 81.187135] intel_sst_acpi 80860F28:00: sst_load_fw [ 81.187162] intel_sst_acpi 80860F28:00: sst: Resetting the DSP in mrfld [ 81.187174] intel_sst_acpi 80860F28:00: value:0x1e40001 [ 81.187184] intel_sst_acpi 80860F28:00: value:0x1e40007 [ 81.187193] intel_sst_acpi 80860F28:00: value:0x1e40006 [ 81.229015] intel_sst_acpi 80860F28:00: config written to DCCM [ 81.229024] intel_sst_acpi 80860F28:00: sst: Starting the DSP in mrfld LALALALA [ 81.229030] intel_sst_acpi 80860F28:00: value:0x1e40006 [ 81.229035] intel_sst_acpi 80860F28:00: value:0x1e40007 [ 81.229041] intel_sst_acpi 80860F28:00: sst: Starting the DSP_merrifield:1e40802 [ 81.239680] intel_sst_acpi 80860F28:00: Interrupt received [ 81.239999] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240040] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240113] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240165] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240232] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240252] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240322] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240357] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240423] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240451] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240517] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240538] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240601] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240632] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240695] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240719] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240785] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240817] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240883] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240956] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241029] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241099] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241176] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241218] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241298] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241340] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241413] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241451] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241475] intel_sst_acpi 80860F28:00: FW sent error response 0x5000e [ 81.241528] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241552] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241580] intel_sst_acpi 80860F28:00: FW sent error response 0x5000e [ 81.241638] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241661] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241692] intel_sst_acpi 80860F28:00: FW sent error response 0x5000e [ 81.241734] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241999] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242069] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242096] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242145] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242298] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242340] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242400] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242431] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242550] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242570] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242689] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242807] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242828] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242888] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242919] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243012] intel_sst_acpi 80860F28:00: FW sent async error msg: [ 81.243029] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243048] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243163] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243277] intel_sst_acpi 80860F28:00: fw load successful!!! [ 81.243279] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243307] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243384] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243460] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243503] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243541] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243614] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243653] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243684] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243715] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243757] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243789] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243820] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243860] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243911] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243953] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243994] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244036] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244078] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244120] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244170] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244213] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244257] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244298] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244340] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244382] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244423] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244472] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244517] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244639] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244769] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244792] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244915] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244956] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245058] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245187] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245316] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245424] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245479] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245532] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245579] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245623] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245672] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245706] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245748] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245786] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245816] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245880] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245931] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245967] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245997] intel_sst_acpi 80860F28:00: Interrupt received [ 81.246050] intel_sst_acpi 80860F28:00: Interrupt received [ 84.180360] intel_sst_acpi 80860F28:00: Interrupt received [ 84.180479] intel_sst_acpi 80860F28:00: sst: Resetting the DSP in mrfld [ 84.180507] intel_sst_acpi 80860F28:00: value:0x1e4080a [ 84.180524] intel_sst_acpi 80860F28:00: value:0x1e40807 [ 84.180539] intel_sst_acpi 80860F28:00: value:0x1e40806 Am I doing something wrong? Any hints? Thanks, Michele My changes: Signed-off-by: Michele Curti <michele.curti(a)gmail.com> --- sound/soc/codecs/rt5645.c | 2 + sound/soc/intel/Kconfig | 13 ++ sound/soc/intel/atom/sst/sst.c | 3 + sound/soc/intel/atom/sst/sst_acpi.c | 4 +- sound/soc/intel/atom/sst/sst_loader.c | 1 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcr_rt5645.c | 332 ++++++++++++++++++++++++++++++++++ 7 files changed, 356 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/boards/bytcr_rt5645.c diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7af5e73..c48e5c8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3524,6 +3524,7 @@ static const struct regmap_config temp_regmap = { static const struct i2c_device_id rt5645_i2c_id[] = { { "rt5645", 0 }, + { "rt5648", 0 }, { "rt5650", 0 }, { } }; @@ -3532,6 +3533,7 @@ MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); #ifdef CONFIG_ACPI static const struct acpi_device_id rt5645_acpi_match[] = { { "10EC5645", 0 }, + { "10EC5648", 0 }, { "10EC5650", 0 }, {}, }; diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3e6c23..46b7f8b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -108,6 +108,19 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH Say Y if you have such a device If unsure select "N". +config SND_SOC_INTEL_BYTCR_RT5645_MACH + tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5645 codec" + depends on X86 && I2C + select SND_SOC_RT5645 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI + help + This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR + platforms with RT5645 audio codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec" depends on X86 && I2C diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..3ffdb2e 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -19,6 +19,7 @@ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#define DEBUG 1 #include <linux/module.h> #include <linux/fs.h> #include <linux/interrupt.h> @@ -58,6 +59,8 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) unsigned int size = 0; struct intel_sst_drv *drv = (struct intel_sst_drv *) context; irqreturn_t retval = IRQ_HANDLED; + + dev_dbg(drv->dev, "Interrupt received\n"); /* Interrupt arrived, check src */ isr.full = sst_shim_read64(drv->shim, SST_ISRX); diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3bc4b63..38447d2 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -109,7 +109,7 @@ static const struct sst_res_info byt_rvp_res_info = { .mbox_size = 0x1000, .acpi_lpe_res_index = 0, .acpi_ddr_index = 2, - .acpi_ipc_irq_index = 5, + .acpi_ipc_irq_index = 0, }; static struct sst_platform_info byt_rvp_platform_data = { @@ -327,6 +327,8 @@ static struct sst_acpi_mach sst_acpi_bytcr[] = { &byt_rvp_platform_data }, {"INTCCFFD", "bytcr_rt5640", "intel/fw_sst_0f28.bin", "bytcr_rt5640", NULL, &byt_rvp_platform_data }, + {"10EC5648", "bytcr_rt5645", "intel/fw_sst_0f28_ssp0.bin", "bytcr_rt5645", NULL, + &byt_rvp_platform_data }, {"10EC5651", "bytcr_rt5651", "intel/fw_sst_0f28.bin", "bytcr_rt5651", NULL, &byt_rvp_platform_data }, {}, diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 3391714..9b750ed 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -22,6 +22,7 @@ * This file contains all dsp controlling functions like firmware download, * setting/resetting dsp cores, etc */ +#define DEBUG 1 #include <linux/pci.h> #include <linux/delay.h> #include <linux/fs.h> diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 3310c0f..af24d39 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -3,6 +3,7 @@ snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o +snd-soc-sst-bytcr-rt5645-objs := bytcr_rt5645.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o +obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH) += snd-soc-sst-bytcr-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o diff --git a/sound/soc/intel/boards/bytcr_rt5645.c b/sound/soc/intel/boards/bytcr_rt5645.c new file mode 100644 index 0000000..4974d6b --- /dev/null +++ b/sound/soc/intel/boards/bytcr_rt5645.c @@ -0,0 +1,332 @@ +/* + * bytcr_rt5645.c - ASoc Machine driver for Intel Byt CR platform + * (derived from bytcr_rt5640.c) + * + * Copyright (C) 2015 Intel Corp + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/dmi.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/rt5645.h" +#include "../atom/sst-atom-controls.h" + +static const struct snd_soc_dapm_widget byt_rt5645_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route byt_rt5645_audio_map[] = { + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + + {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5645 */ + {"IN2P", NULL, "Headset Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Speaker", NULL, "LOUTL"}, + {"Speaker", NULL, "LOUTR"}, +}; + +static const struct snd_soc_dapm_route byt_rt5645_intmic_dmic1_map[] = { + {"DMIC1", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5645_intmic_dmic2_map[] = { + {"DMIC2", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5645_intmic_in1_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN1P", NULL, "Internal Mic"}, +}; + +enum { + BYT_RT5645_DMIC1_MAP, + BYT_RT5645_DMIC2_MAP, + BYT_RT5645_IN1_MAP, +}; + +#define BYT_RT5645_MAP(quirk) ((quirk) & 0xff) +#define BYT_RT5645_DMIC_EN BIT(16) + +static unsigned long byt_rt5645_quirk = BYT_RT5645_DMIC1_MAP | + BYT_RT5645_DMIC_EN; + +static const struct snd_kcontrol_new byt_rt5645_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int byt_rt5645_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + snd_soc_dai_set_bclk_ratio(codec_dai, 50); + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec clock %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_BCLK1, + params_rate(params) * 50, + params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct dmi_system_id byt_rt5645_quirk_table[] = { + {} +}; + +static int byt_rt5645_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + const struct snd_soc_dapm_route *custom_map; + int num_routes; + + card->dapm.idle_bias_off = true; + + dmi_check_system(byt_rt5645_quirk_table); + switch (BYT_RT5645_MAP(byt_rt5645_quirk)) { + case BYT_RT5645_IN1_MAP: + custom_map = byt_rt5645_intmic_in1_map; + num_routes = ARRAY_SIZE(byt_rt5645_intmic_in1_map); + break; + case BYT_RT5645_DMIC2_MAP: + custom_map = byt_rt5645_intmic_dmic2_map; + num_routes = ARRAY_SIZE(byt_rt5645_intmic_dmic2_map); + break; + default: + custom_map = byt_rt5645_intmic_dmic1_map; + num_routes = ARRAY_SIZE(byt_rt5645_intmic_dmic1_map); + } + + ret = snd_soc_add_card_controls(card, byt_rt5645_controls, + ARRAY_SIZE(byt_rt5645_controls)); + if (ret) { + dev_err(card->dev, "unable to add card controls\n"); + return ret; + } + snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + + return ret; +} + +static const struct snd_soc_pcm_stream byt_rt5645_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int byt_rt5645_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + int ret; + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + + /* + * Default mode for SSP configuration is TDM 4 slot, override config + * with explicit setting to I2S 2ch 24-bit. The word length is set with + * dai_set_tdm_slot() since there is no other API exposed + */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS + ); + + if (ret < 0) { + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); + return ret; + } + + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int byt_rt5645_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops byt_rt5645_aif1_ops = { + .startup = byt_rt5645_aif1_startup, +}; + +static struct snd_soc_ops byt_rt5645_be_ssp2_ops = { + .hw_params = byt_rt5645_aif1_hw_params, +}; + +static struct snd_soc_dai_link byt_rt5645_dais[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &byt_rt5645_aif1_ops, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &byt_rt5645_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5648:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = byt_rt5645_codec_fixup, + .ignore_suspend = 1, + .nonatomic = true, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = byt_rt5645_init, + .ops = &byt_rt5645_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card byt_rt5645_card = { + .name = "bytcr-rt5645", + .owner = THIS_MODULE, + .dai_link = byt_rt5645_dais, + .num_links = ARRAY_SIZE(byt_rt5645_dais), + .dapm_widgets = byt_rt5645_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_rt5645_widgets), + .dapm_routes = byt_rt5645_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_rt5645_audio_map), + .fully_routed = true, +}; + +static int snd_byt_rt5645_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + byt_rt5645_card.dev = &pdev->dev; + + ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5645_card); + + if (ret_val) { + dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", + ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &byt_rt5645_card); + return ret_val; +} + +static struct platform_driver snd_byt_rt5645_mc_driver = { + .driver = { + .name = "bytcr_rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_byt_rt5645_mc_probe, +}; + +module_platform_driver(snd_byt_rt5645_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver for RT5645"); +MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytcr_rt5645"); -- 2.8.0

1 0

[alsa-devel] [PATCH v2 0/9] ASoC: hdac_hdmi: Add multichannel support
by Subhransu S. Prusty 28 Apr '16

28 Apr '16

This series parses ELD to identify multichannel capability of the sync, and uses HDA framework to programs codec's channel map registers. Channel map controls are registered in patch 8, with which user can specify a specific channel order. chmap controls are registered inside jack_init callback. As PCMs need to be registered before registering chmap controls, jack_init call in the machine driver is moved to late_probe instead of init callback. Skylake driver is modified and buffer size calculation is fixed for multichannel support. Please note, patch 2 adds a small macro to read channel counts from cap bits in HDA headers. This needs Takashi's ack, so that it can be merged through ASoC. Patch 7 has dependency on commit#44fde3b ("ALSA: hda - Update chmap tlv to report sink's capability) which is merged in Takashi's tree. As Takashi suggested Mark may pull Takashi's for-next branch before applying this or may send a pull request, so that Takashi pull's Mark's tree before applying these with Mark's ack. Mark, please check how you want this to be merged. Changes in v2: * Dropped drm change and accommodate the change in driver * Fix one auto build test error Subhransu S. Prusty (9): ASoC: hdac_hdmi: parse eld for channel map capability ALSA: hda - add helper to get channels from cap bits ASoC: hdac_hdmi: Add multichannel support ASoC: skl_rt286: Fix to support hdmi channel map support ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap ASoC: hdac_hdmi: Register chmap controls and ops ASoC: Intel: Skylake: Add multichannel support for HDMI ASoC: Intel: Skylake: Update channel map based on runtime params sound/hda/local.h | 10 ++ sound/soc/codecs/hdac_hdmi.c | 165 ++++++++++++++++++++++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 74 ++++++++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 73 ++++++++++- sound/soc/intel/boards/skl_rt286.c | 48 ++++++- sound/soc/intel/skylake/skl-pcm.c | 14 +- sound/soc/intel/skylake/skl-topology.c | 21 ++- 7 files changed, 381 insertions(+), 24 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH v3 0/9] ASoC: hdac_hdmi: Add multichannel support
by Subhransu S. Prusty 28 Apr '16

28 Apr '16

This series parses ELD to identify multichannel capability of the sync, and uses HDA framework to programs codec's channel map registers. Channel map controls are registered in patch 8, with which user can specify a specific channel order. chmap controls are registered inside jack_init callback. As PCMs need to be registered before registering chmap controls, jack_init call in the machine driver is moved to late_probe instead of init callback. Skylake driver is modified and buffer size calculation is fixed for multichannel support. Please note, patch 2 adds a small macro to read channel counts from cap bits in HDA headers. This needs Takashi's ack, so that it can be merged through ASoC. Patch 7 has dependency on commit#44fde3b ("ALSA: hda - Update chmap tlv to report sink's capability) which is merged in Takashi's tree. As Takashi suggested Mark may pull Takashi's for-next branch before applying this or may send a pull request, so that Takashi pull's Mark's tree before applying these with Mark's ack. Mark, please check how you want this to be merged. Changes in v3: * As suggested, simplified channel calculation expression in get channel helper * Removed condition in max channel assignment expression Subhransu S. Prusty (9): ASoC: hdac_hdmi: parse eld for channel map capability ALSA: hda - add helper to get channels from cap bits ASoC: hdac_hdmi: Add multichannel support ASoC: skl_rt286: Fix to support hdmi channel map support ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap ASoC: hdac_hdmi: Register chmap controls and ops ASoC: Intel: Skylake: Add multichannel support for HDMI ASoC: Intel: Skylake: Update channel map based on runtime params sound/hda/local.h | 10 ++ sound/soc/codecs/hdac_hdmi.c | 163 ++++++++++++++++++++++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 74 ++++++++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 73 ++++++++++- sound/soc/intel/boards/skl_rt286.c | 48 ++++++- sound/soc/intel/skylake/skl-pcm.c | 14 +- sound/soc/intel/skylake/skl-topology.c | 21 ++- 7 files changed, 379 insertions(+), 24 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH 00/11] ASoC: hdac_hdmi: Add multichannel support
by Subhransu S. Prusty 28 Apr '16

28 Apr '16

This series parses ELD to identify multichannel capability of the sync, and uses HDA framework to programs codec's channel map registers. Channel map controls are registered in patch 8, with which user can specify a specific channel order. chmap controls are registered inside jack_init callback. As PCMs need to be registered before registering chmap controls, jack_init call in the machine driver is moved to late_probe instead of init callback. Skylake driver is modified and buffer size calculation is fixed for multichannel support. Please note, patch 2 and 4 adds small macros to read speaker allocation from ELD and channel counts from cap bits respectively in DRM and HDA headers. We have CCed DRM folks here. Please ack patch 1, so that we can have this series merged through sound trees. Patch 3 needs Takashi's ack, so that it can be merged through ASoC. Patch 9 has dependency on commit#44fde3b ("ALSA: hda - Update chmap tlv to report sink's capability) which is merged in Takashi's tree. So this may be merged in ASoC with Takashi's Ack and pulling in changes from Takashi's tree or alternatively it can be directly merged in Takashi's with Mark's Ack Takashi/Mark, please decide how you guys want this to be merged. Cc: Jani Nikula <jani.nikula(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Daniel Vetter <daniel.vetter(a)intel.com> Subhransu S. Prusty (11): ASoC: Intel: Skylake: Fix ibs/obs calc for non-integral sampling rates drm/edid: Add API to get speaker allocation ASoC: hdac_hdmi: parse eld for channel map capability ALSA: hda - add helper to get channels from cap bits ASoC: hdac_hdmi: Add multichannel support ASoC: skl_rt286: Fix to support hdmi channel map support ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap ASoC: hdac_hdmi: Register chmap controls and ops ASoC: Intel: Skylake: Add multichannel support for HDMI ASoC: Intel: Skylake: Update channel map based on runtime params include/drm/drm_edid.h | 9 ++ sound/hda/local.h | 10 ++ sound/soc/codecs/hdac_hdmi.c | 165 ++++++++++++++++++++++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 74 ++++++++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 73 ++++++++++- sound/soc/intel/boards/skl_rt286.c | 48 ++++++- sound/soc/intel/skylake/skl-pcm.c | 14 +- sound/soc/intel/skylake/skl-topology.c | 49 +++++-- 8 files changed, 409 insertions(+), 33 deletions(-) -- 1.9.1

4 24

[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 28 Apr '16

28 Apr '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changed some controls to TLV as Mark suggested. .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max98371.c | 442 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 67 ++++ 5 files changed, 531 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 913c83c..ba6c585 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -491,6 +492,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f39dee7..9f3587b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -70,6 +70,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..21fae09 --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,442 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain, + 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), + 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0), + 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0), + 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0), + 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), + 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), +); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker DHT Max Volume", MAX98371_GAIN, + 0, (1<<MAX98371_DHT_MAX_WIDTH)-1, 0, + max98371_dht_max_gain), + SOC_SINGLE_TLV("Speaker DHT Min Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_GAIN_WIDTH)-1, 0, + max98371_dht_min_gain), + SOC_SINGLE_TLV("Speaker DHT Rotation Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_ROT_WIDTH)-1, 0, + max98371_dht_rot_gain), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..9f63309 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,67 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_DHT_MAX_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_DHT_GAIN 0x31 +#define MAX98371_DHT_GAIN_WIDTH 0x4 +#define MAX98371_DHT_ROT_WIDTH 0x4 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda: fix the missing ptr initialization
by Vinod Koul 28 Apr '16

28 Apr '16

ebus is a member of extended device and was never initialized, so do this at device creation. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..64de0a3d6d93 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -144,6 +144,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: rt5651: remove unexisting widget
by Bard Liao 28 Apr '16

28 Apr '16

IN Muxes are not existing on rt5651. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 43 ------------------------------------------- 1 file changed, 43 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 7a61970..979e3e9 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -586,44 +586,6 @@ static const struct snd_kcontrol_new hpo_r_mute_control = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, RT5651_R_MUTE_SFT, 1, 1); -/* INL/R source */ -static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl_enum, RT5651_INL1_INR1_VOL, - RT5651_INL_SEL_SFT, rt5651_inl_src); - -static const struct snd_kcontrol_new rt5651_inl1_mux = - SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum); - -static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr1_enum, RT5651_INL1_INR1_VOL, - RT5651_INR_SEL_SFT, rt5651_inr1_src); - -static const struct snd_kcontrol_new rt5651_inr1_mux = - SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum); - -static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl2_enum, RT5651_INL2_INR2_VOL, - RT5651_INL_SEL_SFT, rt5651_inl2_src); - -static const struct snd_kcontrol_new rt5651_inl2_mux = - SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum); - -static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr2_enum, RT5651_INL2_INR2_VOL, - RT5651_INR_SEL_SFT, rt5651_inr2_src); - -static const struct snd_kcontrol_new rt5651_inr2_mux = - SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum); - - /* Stereo ADC source */ static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"}; @@ -955,11 +917,6 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH] ASoC: rt5651: remove unexisting widget
by Bard Liao 28 Apr '16

28 Apr '16

IN Muxes are not existing on rt5651. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 7a61970..b812c06 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -955,11 +955,6 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: nau8825: apply BIQ to reduce recording signal DC offset
by John Hsu 28 Apr '16

28 Apr '16

Enable biquad filter as HPF in the ADC path when headset detected. It helps to reduce recording signal DC offset. The biquad filter parameters are applied for the ADC path when jack is inserted. The biquad filter is cancelled when jack is ejected. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 59 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 7 ++++++ 2 files changed, 66 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f35019a..962d2f0 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -156,6 +156,20 @@ static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_CHARGE_PUMP, 0x0 }, }; +/* Biquad filter coefficients for A1, A2, B0, B1, and B3 */ +static const struct reg_default nau8825_adc_biq_reg[] = { + { NAU8825_REG_BIQ_COF1, 0x009B }, + { NAU8825_REG_BIQ_COF2, 0x0006 }, + { NAU8825_REG_BIQ_COF3, 0xff66 }, + { NAU8825_REG_BIQ_COF4, 0x0000 }, + { NAU8825_REG_BIQ_COF5, 0xffb3 }, + { NAU8825_REG_BIQ_COF6, 0x0000 }, + { NAU8825_REG_BIQ_COF7, 0x009A }, + { NAU8825_REG_BIQ_COF8, 0x0006 }, + { NAU8825_REG_BIQ_COF9, 0xffb3 }, + { NAU8825_REG_BIQ_COF10, 0x8000 }, +}; + static bool nau8825_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -217,12 +231,46 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) case NAU8825_REG_SARDOUT_RAM_STATUS: case NAU8825_REG_CHARGE_PUMP_INPUT_READ: case NAU8825_REG_GENERAL_STATUS: + case NAU8825_REG_BIQ_CTRL ... NAU8825_REG_BIQ_COF10: return true; default: return false; } } +static void nau8825_adc_biq_apply(struct nau8825 *nau8825) +{ + int i; + + /* Enable the BIQ parameter write/updated. The control is valid only + * when change from 0 to 1. The driver applies the BIQ function into + * the ADC path only. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN | NAU8825_BIQ_PATH_MASK, + NAU8825_BIQ_PATH_ADC); + for (i = 0; i < ARRAY_SIZE(nau8825_adc_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_adc_biq_reg[i].reg, + nau8825_adc_biq_reg[i].def); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, NAU8825_BIQ_WRT_EN); +} + +static void nau8825_adc_biq_cancel(struct nau8825 *nau8825) +{ + int i; + + /* Disable the BIQ parameter write/updated. The control is valid only + * when change from 0 to 1. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, 0); + for (i = 0; i < ARRAY_SIZE(nau8825_adc_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_adc_biq_reg[i].reg, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, NAU8825_BIQ_WRT_EN); +} + static int nau8825_adc_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -647,6 +695,9 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) struct snd_soc_dapm_context *dapm = nau8825->dapm; struct regmap *regmap = nau8825->regmap; + /* Cancel biquad filter in ADC path */ + nau8825_adc_biq_cancel(nau8825); + snd_soc_dapm_disable_pin(dapm, "SAR"); snd_soc_dapm_disable_pin(dapm, "MICBIAS"); /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ @@ -737,6 +788,14 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) break; } + if (type & SND_JACK_MICROPHONE) { + /* Apply biquad filter into ADC path. It helps to avoid the DC + * offset in recoding signal. The function will keep asleep + * without clock and work up automatically when clock provided. + */ + nau8825_adc_biq_apply(nau8825); + } + /* Leaving HPOL/R grounded after jack insert by default. They will be * ungrounded as part of the widget power up sequence at the beginning * of playback to reduce pop. diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..f3b4acb 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -231,6 +231,13 @@ #define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) +/* BIQ_CTRL (0x20) */ +#define NAU8825_BIQ_WRT_EN (1 << 4) +#define NAU8825_BIQ_PATH_SFT 0 +#define NAU8825_BIQ_PATH_MASK (1 << NAU8825_BIQ_PATH_SFT) +#define NAU8825_BIQ_PATH_ADC (0 << NAU8825_BIQ_PATH_SFT) +#define NAU8825_BIQ_PATH_DAC (1 << NAU8825_BIQ_PATH_SFT) + /* ADC_RATE (0x2b) */ #define NAU8825_ADC_SYNC_DOWN_SFT 0 #define NAU8825_ADC_SYNC_DOWN_MASK 0x3 -- 2.6.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: dmaengine_pcm: Add support for packed transfers
by Matthias Reichl 27 Apr '16

27 Apr '16

dmaengine_pcm currently only supports setups where FIFO reads/writes correspond to exactly one sample, eg 16-bit sample data is transferred via 16-bit FIFO accesses, 32-bit data via 32-bit accesses. This patch adds support for setups with fixed width FIFOs where multiple samples are packed into a larger word. For example setups with a 32-bit wide FIFO register that expect 16-bit sample transfers to be done with the left+right sample data packed into a 32-bit word. Support for packed transfers is controlled via the SND_DMAENGINE_PCM_DAI_FLAG_PACK flag in snd_dmaengine_dai_dma_data.flags If this flag is set dmaengine_pcm doesn't put any restriction on the supported formats and sets the DMA transfer width to undefined. This means control over the constraints is now transferred to the DAI driver and it's responsible to provide proper configuration and check for possible corner cases that aren't handled by the ALSA core. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- include/sound/dmaengine_pcm.h | 12 ++++++++ sound/core/pcm_dmaengine.c | 11 +++++-- sound/soc/soc-generic-dmaengine-pcm.c | 57 +++++++++++++++++++++-------------- 3 files changed, 55 insertions(+), 25 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f86ef5e..67be244 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -51,6 +51,16 @@ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, void *filter_data); struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); +/* + * The DAI supports packed transfers, eg 2 16-bit samples in a 32-bit word. + * If this flag is set the dmaengine driver won't put any restriction on + * the supported sample formats and set the DMA transfer size to undefined. + * The DAI driver is responsible to disable any unsupported formats in it's + * configuration and catch corner cases that are not already handled in + * the ALSA core. + */ +#define SND_DMAENGINE_PCM_DAI_FLAG_PACK BIT(0) + /** * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data * @addr: Address of the DAI data source or destination register. @@ -63,6 +73,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * requesting the DMA channel. * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes + * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -72,6 +83,7 @@ struct snd_dmaengine_dai_dma_data { void *filter_data; const char *chan_name; unsigned int fifo_size; + unsigned int flags; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 697c166..8eb58c7 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); * direction of the substream. If the substream is a playback stream the dst * fields will be initialized, if it is a capture stream the src fields will be * initialized. The {dst,src}_addr_width field will only be initialized if the - * addr_width field of the DAI DMA data struct is not equal to - * DMA_SLAVE_BUSWIDTH_UNDEFINED. + * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of + * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If + * both conditions are met the latter takes priority. */ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_pcm_substream *substream, @@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data( if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->dst_addr = dma_data->addr; slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->dst_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->dst_addr_width = dma_data->addr_width; } else { slave_config->src_addr = dma_data->addr; slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->src_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->src_addr_width = dma_data->addr_width; } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6fd1906..6cef397 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea } /* - * Prepare formats mask for valid/allowed sample types. If the dma does - * not have support for the given physical word size, it needs to be - * masked out so user space can not use the format which produces - * corrupted audio. - * In case the dma driver does not implement the slave_caps the default - * assumption is that it supports 1, 2 and 4 bytes widths. + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. */ - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* Enable only samples with DMA supported physical widths */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); - break; - default: - /* Unsupported types */ - break; + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } - } return snd_soc_set_runtime_hwparams(substream, &hw); } -- 2.1.4

5 7

[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 27 Apr '16

27 Apr '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max98371.c | 419 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 64 ++++ 5 files changed, 505 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 913c83c..ba6c585 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -491,6 +492,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f39dee7..9f3587b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -70,6 +70,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..ce735db --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,419 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const voltage_level_text[] = { + "9.5dB", "10.5dB", "11.5dB", "11.5dB", "12.5dB", + "13.5dB", "14.5dB", "15.5dB", "16.5dB", "17.5dB", + "18.5dB", "19.5dB", "20.5dB" +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_voltage, MAX98371_GAIN, 0, + voltage_level_text); + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker No-load Volume", MAX98371_GAIN, + 0, (1<<MAX98371_NO_LOAD_WIDTH)-1, 0, + max98371_noload_gain_tlv), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Voltage Level", max98371_voltage), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..c75b259 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,64 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_NO_LOAD_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

2 7

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: read DSP mode A data on rising edges of bclk
by Peter Rosin 27 Apr '16

27 Apr '16

Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6dca95aa8a82..4ab1e2013238 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm_adsp: free memory when unloaded or closed
by Richard Fitzgerald 27 Apr '16

27 Apr '16

The patch adds a wm_adsp2_remove() function to ensure that memory is freed when the driver is unloaded or shut down. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 20 ++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 21 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b6470de..c350a52 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1039,6 +1039,13 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } +static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) +{ + kfree(ctl->cache); + kfree(ctl->name); + kfree(ctl); +} + static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region, unsigned int offset, unsigned int len, @@ -2434,6 +2441,19 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +void wm_adsp2_remove(struct wm_adsp *dsp) +{ + struct wm_coeff_ctl *ctl; + + while (!list_empty(&dsp->ctl_list)) { + ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl, + list); + list_del(&ctl->list); + wm_adsp_free_ctl_blk(ctl); + } +} +EXPORT_SYMBOL_GPL(wm_adsp2_remove); + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index b61cb57..feb61e2 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +void wm_adsp2_remove(struct wm_adsp *dsp); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, -- 1.9.1

2 5

[alsa-devel] [PATCH v4 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 27 Apr '16

27 Apr '16

Minor update to driver for TI's TAS5720L/M digital audio amplifiers incorporating previous feedback (thanks Mark Brown and Andrew F Davis), see below change log for details. Also re-tested different aspects of the driver to make sure it still builds/works. -- Andreas Dannenberg Texas Instruments Inc Changes since v3: - Rework fault condition reporting to use dev_crit() to pay justice to the severity of the issues, also shorten the error string for more clarity/brevity - Fix Kconfig help entry to no longer mention the interrupt signal that was used previously - Simplify hw_params checks as the driver in its current form only supports left-justified audio formats and the codec is tolerant regarding the number of bits that are actually received. - Clarify DT bindings to make the power supply properties required - Remove .owner assignment from I2C driver struct as it is no longer needed (taken care off by the Kernel) Changes since v2: - Switched fault handling from using interrupts to polling mode - Remove interrupt related description from DT bindings doc - Remove unlikely() to simplify an expression - Remove unnesseary tas5720_set_dai_sysclk() function stub Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 25 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 620 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 745 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

2 3

[alsa-devel] [PATCH 1/2] ASoC: topology: Check failure to create a widget
by mengdong.lin＠linux.intel.com 27 Apr '16

27 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Stop loading topology info if error happens when creating a widget. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index bde79f7..e38664e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1500,9 +1500,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; ret = soc_tplg_dapm_widget_create(tplg, widget); - if (ret < 0) + if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", widget->name); + return ret; + } } return 0; -- 2.5.0

2 1

[alsa-devel] [PATCH 2/2] ASoC: topology: Check size mismatch of ABI objects before parsing
by mengdong.lin＠linux.intel.com 27 Apr '16

27 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> If size mismatch of manifest, ABI headers or elements is found, stop parsing topology info and return the error. New fields may be append to the tail of ABI objects which will cause object size to increase. If user space and kernel use different versions of ABI, size mismatch will be detected here. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index e38664e..2261585 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + if (control_hdr->size != sizeof(*control_hdr)) { + dev_err(tplg->dev, "ASoC: invalid control size\n"); + return -EINVAL; + } + switch (control_hdr->ops.info) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: @@ -1499,6 +1504,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; + if (widget->size != sizeof(*widget)) { + dev_err(tplg->dev, "ASoC: invalid widget size\n"); + return -EINVAL; + } + ret = soc_tplg_dapm_widget_create(tplg, widget); if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", @@ -1651,8 +1661,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; - pcm = (struct snd_soc_tplg_pcm *)tplg->pos; - if (soc_tplg_check_elem_count(tplg, sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { @@ -1662,7 +1670,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ + pcm = (struct snd_soc_tplg_pcm *)tplg->pos; for (i = 0; i < count; i++) { + if (pcm->size != sizeof(*pcm)) { + dev_err(tplg->dev, "ASoC: invalid pcm size\n"); + return -EINVAL; + } + soc_tplg_pcm_create(tplg, pcm); pcm++; } @@ -1682,6 +1696,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, return 0; manifest = (struct snd_soc_tplg_manifest *)tplg->pos; + if (manifest->size != sizeof(*manifest)) { + dev_err(tplg->dev, "ASoC: invalid manifest size\n"); + return -EINVAL; + } + tplg->pos += sizeof(struct snd_soc_tplg_manifest); if (tplg->comp && tplg->ops && tplg->ops->manifest) @@ -1698,6 +1717,14 @@ static int soc_valid_header(struct soc_tplg *tplg, if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) return 0; + if (hdr->size != sizeof(*hdr)) { + dev_err(tplg->dev, + "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n", + hdr->type, soc_tplg_get_hdr_offset(tplg), + tplg->fw->size); + return -EINVAL; + } + /* big endian firmware objects not supported atm */ if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { dev_err(tplg->dev, -- 2.5.0

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.6
by Mark Brown 27 Apr '16

27 Apr '16

The following changes since commit 02da2d72174c61988eb4456b53f405e3ebdebce4: Linux 4.6-rc5 (2016-04-24 16:17:05 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.6-rc5 for you to fetch changes up to 8c0f55100478649e3c0133684aa4aebab77ce10f: Merge remote-tracking branches 'asoc/fix/rt5640' and 'asoc/fix/wm8962' into asoc-linus (2016-04-26 19:25:18 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.6 This is a fairly large collection of fixes but almost all driver specific ones, especially to the new Intel drivers which have had a lot of recent development. The one core fix is a change to the debugfs code to avoid crashes in some relatively unusual configurations. ---------------------------------------------------------------- Arnd Bergmann (2): ASoC: cs35l32: avoid uninitialized variable access ASoC: rt5616: add I2C dependency Ben Zhang (1): ASoC: nau8825: Fix jack detection across suspend Charles Keepax (3): ASoC: wm8962: Correct typo when setting DSPCLK rate ASoC: arizona: Free speaker thermal IRQs in CODEC remove ASoC: wm5102: Free compressed IRQ in CODEC remove Dan Carpenter (1): ASoC: Intel: sst: fix a loop timeout in sst_hsw_stream_reset() Jeeja KP (2): ASoC: Intel: Skylake: Fix for unloading module only when it is loaded ASoC: Intel: Skylake: Fix DSP resource de-allocation Mark Brown (5): ASoC: dapm: Make sure we have a card when displaying component widgets Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/cs35l32', 'asoc/fix/hdac', 'asoc/fix/nau8825' and 'asoc/fix/rt5616' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5640' and 'asoc/fix/wm8962' into asoc-linus Subhransu S. Prusty (4): ASoC: hdac_hdmi: Fix to use dev_pm ops instead soc pm ASoC: hdac_hdmi: Fix codec power state in S3 during playback ASoC: Intel: Skylake: Fix to turn OFF codec power when entering S3 ASoC: Intel: Skylake: Fix ibs/obs calc for non-integral sampling rates Sugar Zhang (1): ASoC: rt5640: Correct the digital interface data select Vinod Koul (7): ALSA: hda: use list macro for parsing on cleanup ASoC: Intel: Skylake: free codec objects on removal ASoC: Intel: Skylake: Freeup properly on skl_dsp_free ASoC: Intel: Skylake: Unmap the address last ASoC: Intel: Skylake: Call i915 exit last ASoC: Intel: Skylake: remove call to pci_dev_put ASoC: Intel: Skylake: Fix kbuild dependency sound/hda/ext/hdac_ext_stream.c | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.c | 12 +++ sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/cs35l32.c | 17 +++- sound/soc/codecs/cs47l24.c | 3 + sound/soc/codecs/hdac_hdmi.c | 94 ++++++++++------------ sound/soc/codecs/nau8825.c | 126 +++++++++++++++++------------- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5640.h | 36 ++++----- sound/soc/codecs/wm5102.c | 5 ++ sound/soc/codecs/wm5110.c | 2 + sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8997.c | 2 + sound/soc/codecs/wm8998.c | 2 + sound/soc/intel/Kconfig | 1 - sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- sound/soc/intel/skylake/skl-sst-dsp.c | 5 ++ sound/soc/intel/skylake/skl-topology.c | 42 ++++++---- sound/soc/intel/skylake/skl-topology.h | 8 +- sound/soc/intel/skylake/skl.c | 32 +++++--- sound/soc/soc-dapm.c | 7 ++ 22 files changed, 245 insertions(+), 163 deletions(-)

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[alsa-devel] [PATCH v7 0/9] ASoC: MT8173 HDMI codec support
by Philipp Zabel 27 Apr '16

27 Apr '16

Since the hdmi-codec patches are now applied [1], I have merged the hdmi-codec interface into the initial MT8173 HDMI [2] driver submission. This series now only enables the HDMI dai-links in the mt8173-rt5650-rt5676 and mt8173-rt5650 machine drivers and adds hdmi-codec ELD support and the still up in the air hdmi notifications between DRM encoder driver and hdmi-codec. Changes since v6: - Dropped "drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver", patch now merged into MT8173 HDMI encoder patch. - Fixed jack detection in case HDMI is already connected at boot and the hdmi-codec driver registers the notifier block after the HDMI_CONNECTED notification is issued. [1] commit 09184118a8ab ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/hdmi The mediatek drm patches apply on top of [2] https://patchwork.kernel.org/patch/8887051/ ("drm/mediatek: Add HDMI support") and need this patch: [3] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (7): ASoC: hdmi-codec: Add ELD control video: rmk's HDMI notification prototype ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver drm/mediatek: hdmi: issue notifications drm/mediatek: hdmi: use helper function for N and CTS calculation .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 88 ++++----------- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 ++++++++++ include/linux/hdmi-notifier.h | 44 ++++++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 124 +++++++++++++++++++-- sound/soc/mediatek/Kconfig | 2 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 48 ++++++++ sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++++ 14 files changed, 370 insertions(+), 76 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.8.0.rc3

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[alsa-devel] [PATCH RESEND] ASoC: arizona: Do not create OUT4R widget for CS47L24/WM1831
by Richard Fitzgerald 27 Apr '16

27 Apr '16

The CS47L24 and WM1831 codecs only use the OUT4L widget so we can skip creation of the OUT4R widget. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index da61c62..0cd27e0 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -223,6 +223,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); -- 1.9.1

1 1

[alsa-devel] [PATCH 1/3] ASoC: bcm2835: add 24bit support
by kernel＠martin.sperl.org 26 Apr '16

26 Apr '16

From: Matthias Reichl <hias(a)horus.com> This adds 24 bit support to the I2S driver of the BCM2835 Code ported from bcm2708-i2s driver in Raspberry Pi tree. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> Signed-off-by: Matthias Reichl <hias(a)horus.com> Signed-off-by: Martin Sperl <kernel(a)martin.sperl.org> --- sound/soc/bcm/bcm2835-i2s.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 1c1f221..d2663e7 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: data_length = 16; break; + case SNDRV_PCM_FORMAT_S24_LE: + data_length = 24; + break; case SNDRV_PCM_FORMAT_S32_LE: data_length = 32; break; @@ -279,7 +282,7 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* Setup the frame format */ format = BCM2835_I2S_CHEN; - if (data_length > 24) + if (data_length >= 24) format |= BCM2835_I2S_CHWEX; format |= BCM2835_I2S_CHWID((data_length-8)&0xf); @@ -570,6 +573,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .capture = { @@ -577,6 +581,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .ops = &bcm2835_i2s_dai_ops, -- 2.1.4

3 11

[alsa-devel] [PATCH] ASoC: omap-pcm: Initialize DMA configuration
by J.D. Schroeder 26 Apr '16

26 Apr '16

From: Jim Lodes <jim.lodes(a)garmin.com> Initialize the dma_slave_config for PCM DMA transfers, instead of leaving it uninitialized. Keeps previous data on the stack from giving us invalid values in uninitialized members of the config structure. Signed-off-by: Jim Lodes <jim.lodes(a)garmin.com> Signed-off-by: J.D. Schroeder <jay.schroeder(a)garmin.com> --- sound/soc/omap/omap-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 99381a2..a84f677 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_chan *chan; int err = 0; + memset(&config, 0x00, sizeof(config)); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* return if this is a bufferless transfer e.g. -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: cs42l56: Use IS_ENABLED() instead of checking for built-in or module
by Javier Martinez Canillas 26 Apr '16

26 Apr '16

The IS_ENABLED() macro checks if a Kconfig symbol has been enabled either built-in or as a module, use that macro instead of open coding the same. Signed-off-by: Javier Martinez Canillas <javier(a)osg.samsung.com> --- sound/soc/codecs/cs42l56.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 7cd5f769bb61..eec1ff853b98 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -56,7 +56,7 @@ struct cs42l56_private { u8 iface; u8 iface_fmt; u8 iface_inv; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; -- 2.5.5

2 1

[alsa-devel] [PATCH] ALSA: intel_hda - Add dock support for ThinkPad X260
by Conrad Kostecki 26 Apr '16

26 Apr '16

Hi! My shiny new ThinkPad X260 is the same way affected, as older ThinkPad (X*40, X*50) generations. That means, I am unable to get sound output, when I am using the Lenovo CES 2013 docking station series (basic, pro, ultra). It can be fixed the same way, as it was already done for X240 and X250, as the X260 uses the same docking connector. I am attaching my patch, which works for me. Cheers Conrad ------------------------------ Fixes audio output on a ThinkPad X260, when using Lenovo CES 2013 docking station series (basic, pro, ultra). Signed-off-by: Conrad Kostecki <ck+linuxkernel(a)bl4ckb0x.de> diff -uprN -X linux-4.6-rc4-vanilla/Documentation/dontdiff linux-4.6-rc4-vanilla/sound/pci/hda/patch_realtek.c linux-4.6-rc4/sound/pci/hda/patch_realtek.c --- linux-4.6-rc4-vanilla/sound/pci/hda/patch_realtek.c 2016-04-24 03:26:36.330983586 +0200 +++ linux-4.6-rc4/sound/pci/hda/patch_realtek.c 2016-04-24 03:27:13.737981843 +0200 @@ -5570,6 +5570,7 @@ static const struct snd_pci_quirk alc269 SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), ------------------------------ -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

2 3

[alsa-devel] [PATCH 1/2] ASoC: rt298: reset AD dilter is there is no MCLK
by Bard Liao 26 Apr '16

26 Apr '16

rt298 need to reset AD filter and the ADC settings will take effort. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt298.h | 2 ++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f0e6c06..178b1cc 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -481,6 +481,26 @@ static int rt298_adc_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), 0x7080, 0x7000); + /* If MCLK doesn't exist, reset AD filter */ + if (!(snd_soc_read(codec, RT298_VAD_CTRL) & 0x200)) { + pr_info("NO MCLK\n"); + switch (nid) { + case RT298_ADC_IN1: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x2); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x0); + break; + case RT298_ADC_IN2: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x4); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x0); + break; + } + } break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(codec, diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index d66f884..3638f3d 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -137,6 +137,7 @@ #define RT298_A_BIAS_CTRL2 0x02 #define RT298_POWER_CTRL1 0x03 #define RT298_A_BIAS_CTRL3 0x04 +#define RT298_D_FILTER_CTRL 0x05 #define RT298_POWER_CTRL2 0x08 #define RT298_I2S_CTRL1 0x09 #define RT298_I2S_CTRL2 0x0a @@ -148,6 +149,7 @@ #define RT298_IRQ_CTRL 0x33 #define RT298_WIND_FILTER_CTRL 0x46 #define RT298_PLL_CTRL1 0x49 +#define RT298_VAD_CTRL 0x4e #define RT298_CBJ_CTRL1 0x4f #define RT298_CBJ_CTRL2 0x50 #define RT298_PLL_CTRL 0x63 -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: mediatek: HDMI audio LR channel swapped
by PC Liao 26 Apr '16

26 Apr '16

Because LRCK of TDM use High to Low as default setting, this patch changes the TDM setting to inverse LRCK. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index f1c58a2..2b5df2e 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -123,6 +123,7 @@ #define AFE_TDM_CON1_WLEN_32BIT (0x2 << 8) #define AFE_TDM_CON1_MSB_ALIGNED (0x1 << 4) #define AFE_TDM_CON1_1_BCK_DELAY (0x1 << 3) +#define AFE_TDM_CON1_LRCK_INV (0x1 << 2) #define AFE_TDM_CON1_BCK_INV (0x1 << 1) #define AFE_TDM_CON1_EN (0x1 << 0) @@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, runtime->rate * runtime->channels * 32); val = AFE_TDM_CON1_BCK_INV | + AFE_TDM_CON1_LRCK_INV | AFE_TDM_CON1_1_BCK_DELAY | AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */ AFE_TDM_CON1_WLEN_32BIT | -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: intel_hda - Add dock support for ThinkPad T460s
by Conrad Kostecki 26 Apr '16

26 Apr '16

Hi! The new ThinkPad T460s is the same way affected, as other current ThinkPads (e.G. X260) generations. That means, you won't be able to get sound output, when using the Lenovo CES 2013 docking station series (basic, pro, ultra). It can be fixed the same way, as it was already done for X260, as the T460s uses the same docking connector. I am attaching a patch, which works for me. Cheers Conrad ------------------------------ Fixes audio output on a ThinkPad T460s, when using Lenovo CES 2013 docking station series (basic, pro, ultra). Signed-off-by: Conrad Kostecki <ck+linuxkernel(a)bl4ckb0x.de> --- linux-4.6-rc5-vanilla/sound/pci/hda/patch_realtek.c 2016-04-26 11:32:09.124832372 +0200 +++ linux-4.6-rc5/sound/pci/hda/patch_realtek.c 2016-04-26 11:32:52.123834807 +0200 @@ -5585,6 +5585,7 @@ static const struct snd_pci_quirk alc269 SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T460p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), ------------------------------ -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

1 1

[alsa-devel] [PATCH V2] ASoC: bcm2835: Register as custom PCM device to enable 16bit pcm
by kernel＠martin.sperl.org 26 Apr '16

26 Apr '16

From: Matthias Reichl <hias(a)horus.com> Register as a custom pcm device with the correct settings supporting also 16 bit. The automatic pcm discovery does not enable 16 bit modes because of a percived mismatch between the dma controller only supporting 32 bit transfers and the requirement by the HW to only support exactly 2 channels, which effectively results in a minimum size of 32 bits, which matches the dma controller transfer width. Code ported from bcm2708-i2s driver in Raspberry Pi tree. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> Signed-off-by: Matthias Reichl <hias(a)horus.com> Signed-off-by: Martin Sperl <kernel(a)martin.sperl.org> Changelog: V1 -> V2: use of SZ_* defines instead of multiple of page size better description of commit message --- sound/soc/bcm/bcm2835-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index a0026e2..66680f1 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -632,6 +632,27 @@ static const struct snd_soc_component_driver bcm2835_i2s_component = { .name = "bcm2835-i2s-comp", }; +static const struct snd_pcm_hardware bcm2835_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = 32, + .period_bytes_max = SZ_256K, + .periods_min = 2, + .periods_max = 255, + .buffer_bytes_max = SZ_512K, +}; + +static const struct snd_dmaengine_pcm_config bcm2835_dmaengine_pcm_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .pcm_hardware = &bcm2835_pcm_hardware, + .prealloc_buffer_size = SZ_1M, +}; + static int bcm2835_i2s_probe(struct platform_device *pdev) { struct bcm2835_i2s_dev *dev; @@ -704,7 +725,9 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register( + &pdev->dev, &bcm2835_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); return ret; -- 2.1.4

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Update BCLK also at hotplug for i915 HSW/BDW
by Takashi Iwai 26 Apr '16

26 Apr '16

The recent bug report suggests that BCLK setup for i915 HSW/BDW needs to be updated at each HDMI hotplug, not only at initialization and resume. That is, we need to update HSW_EM4 and HSW_EM5 registers at ELD notification, too. Otherwise the HDMI audio may be out of sync and played in a wrong pitch. However, the HDA codec driver has no access to the controller registers, and currently the code managing these registers is in hda_intel.c, i.e. local to the controller driver. For allowing the explicit BCLK update from the codec driver, as in this patch, the former haswell_set_bclk() in hda_intel.c is moved to hdac_i915.c and exposed as snd_hdac_i915_set_bclk(). This is called from both the HDA controller driver and intel_pin_eld_notify() in HDMI codec driver. Along with this change, snd_hdac_get_display_clk() gets dropped as it's no longer used. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=91410 Cc: <stable(a)vger.kernel.org> # v4.5+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_i915.h | 5 ++-- sound/hda/hdac_i915.c | 62 ++++++++++++++++++++++++++++++++++++++-------- sound/pci/hda/hda_intel.c | 56 +++-------------------------------------- sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 58 insertions(+), 66 deletions(-) diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index fa341fcb5829..f5842bcd9c94 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -9,7 +9,7 @@ #ifdef CONFIG_SND_HDA_I915 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); -int snd_hdac_get_display_clk(struct hdac_bus *bus); +void snd_hdac_i915_set_bclk(struct hdac_bus *bus); int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate); int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes); @@ -25,9 +25,8 @@ static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { return 0; } -static inline int snd_hdac_get_display_clk(struct hdac_bus *bus) +static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { - return 0; } static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 54babe1c0b16..607bbeaebddf 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,7 @@ #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> +#include <sound/hda_register.h> static struct i915_audio_component *hdac_acomp; @@ -97,26 +98,65 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) } EXPORT_SYMBOL_GPL(snd_hdac_display_power); +#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ + ((pci)->device == 0x0c0c) || \ + ((pci)->device == 0x0d0c) || \ + ((pci)->device == 0x160c)) + /** - * snd_hdac_get_display_clk - Get CDCLK in kHz + * snd_hdac_i915_set_bclk - Reprogram BCLK for HSW/BDW * @bus: HDA core bus * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. + * Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. * - * This function queries CDCLK value in kHz from the graphics driver and - * returns the value. A negative code is returned in error. + * Call this function at initializing and changing power well, as well as + * at ELD notifier for the hotplug. */ -int snd_hdac_get_display_clk(struct hdac_bus *bus) +void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { struct i915_audio_component *acomp = bus->audio_component; + struct pci_dev *pci = to_pci_dev(bus->dev); + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq) + return; /* only for i915 binding */ + if (!CONTROLLER_IN_GPU(pci)) + return; /* only HSW/BDW */ + + cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } - if (!acomp || !acomp->ops) - return -ENODEV; - - return acomp->ops->get_cdclk_freq(acomp->dev); + snd_hdac_chip_writew(bus, HSW_EM4, bclk_m); + snd_hdac_chip_writew(bus, HSW_EM5, bclk_n); } -EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); +EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); /* There is a fixed mapping between audio pin node and display port * on current Intel platforms: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 637b8a0e2a91..9a0d1445ca5c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -857,50 +857,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK - * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) - * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: - * BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled and need to - * be restored to avoid abnormal playback speed. - */ -static void haswell_set_bclk(struct hda_intel *hda) -{ - struct azx *chip = &hda->chip; - int cdclk_freq; - unsigned int bclk_m, bclk_n; - - if (!hda->need_i915_power) - return; - - cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip)); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(chip, HSW_EM4, bclk_m); - azx_writew(chip, HSW_EM5, bclk_n); -} - #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -958,7 +914,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { snd_hdac_display_power(azx_bus(chip), true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(azx_bus(chip)); } if (chip->msi) if (pci_enable_msi(pci) < 0) @@ -1058,7 +1014,7 @@ static int azx_runtime_resume(struct device *dev) bus = azx_bus(chip); if (hda->need_i915_power) { snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(bus); } else { /* toggle codec wakeup bit for STATESTS read */ snd_hdac_set_codec_wakeup(bus, true); @@ -1796,12 +1752,8 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - struct hda_intel *hda; - - hda = container_of(chip, struct hda_intel, chip); - haswell_set_bclk(hda); - } + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + snd_hdac_i915_set_bclk(bus); hda_intel_init_chip(chip, (probe_only[dev] & 2) == 0); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 40933aa33afe..1483f85999ec 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2232,6 +2232,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) if (atomic_read(&(codec)->core.in_pm)) return; + snd_hdac_i915_set_bclk(&codec->bus->core); check_presence_and_report(codec, pin_nid); } -- 2.8.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Update BCLK also at hotplug for i915 HSW/BDW
by Takashi Iwai 26 Apr '16

26 Apr '16

The recent bug report suggests that BCLK setup for i915 HSW/BDW needs to be updated at each HDMI hotplug, not only at initialization and resume. That is, we need to update HSW_EM4 and HSW_EM5 registers at ELD notification, too. Otherwise the HDMI audio may be out of sync and played in a wrong pitch. However, the HDA codec driver has no access to the controller registers, and currently the code managing these registers is in hda_intel.c, i.e. local to the controller driver. For allowing the explicit BCLK update from the codec driver, as in this patch, the former haswell_set_bclk() in hda_intel.c is moved to hdac_i915.c and exposed as snd_hdac_i915_set_bclk(). This is called from both the HDA controller driver and intel_pin_eld_notify() in HDMI codec driver. Along with this change, snd_hdac_get_display_clk() gets dropped as it's no longer used. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=91410 Cc: <stable(a)vger.kernel.org> # v4.5+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_i915.h | 5 ++-- sound/hda/hdac_i915.c | 62 +++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_intel.c | 56 +++-------------------------------------- sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 68 insertions(+), 56 deletions(-) diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index fa341fcb5829..f5842bcd9c94 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -9,7 +9,7 @@ #ifdef CONFIG_SND_HDA_I915 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); -int snd_hdac_get_display_clk(struct hdac_bus *bus); +void snd_hdac_i915_set_bclk(struct hdac_bus *bus); int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate); int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes); @@ -25,9 +25,8 @@ static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { return 0; } -static inline int snd_hdac_get_display_clk(struct hdac_bus *bus) +static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { - return 0; } static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 54babe1c0b16..181777b43aa1 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,7 @@ #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> +#include <sound/hda_register.h> static struct i915_audio_component *hdac_acomp; @@ -116,7 +117,66 @@ int snd_hdac_get_display_clk(struct hdac_bus *bus) return acomp->ops->get_cdclk_freq(acomp->dev); } -EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); + +#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ + ((pci)->device == 0x0c0c) || \ + ((pci)->device == 0x0d0c) || \ + ((pci)->device == 0x160c)) + +/** + * snd_hdac_i915_set_bclk - Reprogram BCLK for HSW/BDW + * @bus: HDA core bus + * + * Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. + * + * Call this function at initializing and changing power well, as well as + * at ELD notifier for the hotplug. + */ +void snd_hdac_i915_set_bclk(struct hdac_bus *bus) +{ + struct i915_audio_component *acomp = bus->audio_component; + struct pci_dev *pci = to_pci_dev(bus->dev); + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq) + return; /* only for i915 binding */ + if (!CONTROLLER_IN_GPU(pci)) + return; /* only HSW/BDW */ + + cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + snd_hdac_chip_writew(bus, HSW_EM4, bclk_m); + snd_hdac_chip_writew(bus, HSW_EM5, bclk_n); +} +EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); /* There is a fixed mapping between audio pin node and display port * on current Intel platforms: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 637b8a0e2a91..9a0d1445ca5c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -857,50 +857,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK - * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) - * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: - * BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled and need to - * be restored to avoid abnormal playback speed. - */ -static void haswell_set_bclk(struct hda_intel *hda) -{ - struct azx *chip = &hda->chip; - int cdclk_freq; - unsigned int bclk_m, bclk_n; - - if (!hda->need_i915_power) - return; - - cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip)); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(chip, HSW_EM4, bclk_m); - azx_writew(chip, HSW_EM5, bclk_n); -} - #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -958,7 +914,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { snd_hdac_display_power(azx_bus(chip), true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(azx_bus(chip)); } if (chip->msi) if (pci_enable_msi(pci) < 0) @@ -1058,7 +1014,7 @@ static int azx_runtime_resume(struct device *dev) bus = azx_bus(chip); if (hda->need_i915_power) { snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(bus); } else { /* toggle codec wakeup bit for STATESTS read */ snd_hdac_set_codec_wakeup(bus, true); @@ -1796,12 +1752,8 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - struct hda_intel *hda; - - hda = container_of(chip, struct hda_intel, chip); - haswell_set_bclk(hda); - } + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + snd_hdac_i915_set_bclk(bus); hda_intel_init_chip(chip, (probe_only[dev] & 2) == 0); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 40933aa33afe..1483f85999ec 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2232,6 +2232,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) if (atomic_read(&(codec)->core.in_pm)) return; + snd_hdac_i915_set_bclk(&codec->bus->core); check_presence_and_report(codec, pin_nid); } -- 2.8.1

1 1

[alsa-devel] [PATCH] Asoc: Intel: common: Removed unused sst_acpi_mach fields
by Joel Holdsworth 26 Apr '16

26 Apr '16

The fields in question are currently unused, and are unlikely to be useful in the future: - "board" is a very vague name not related to a concrete matching criteria such as a DMI match or ACPI OEM IDs. - "machine_quirk" lacks the pointer arguments to be able to manipulate any sst-acpi structures. For it to be useful these would need to be added. Furthermore, a quirk function is much more likely to be relevant to a whole chipset, rather than a single board. - Without "machine_quirk", "pdata" serves no purpose and should be removed. --- sound/soc/intel/common/sst-acpi.c | 8 ++++---- sound/soc/intel/common/sst-acpi.h | 5 ----- 2 files changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c index 2c5eda1..aba6575 100644 --- a/sound/soc/intel/common/sst-acpi.c +++ b/sound/soc/intel/common/sst-acpi.c @@ -180,7 +180,7 @@ static int sst_acpi_remove(struct platform_device *pdev) } static struct sst_acpi_mach haswell_machines[] = { - { "INT33CA", "haswell-audio", "intel/IntcSST1.bin", NULL, NULL, NULL }, + { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" }, {} }; @@ -198,7 +198,7 @@ static struct sst_acpi_desc sst_acpi_haswell_desc = { }; static struct sst_acpi_mach broadwell_machines[] = { - { "INT343A", "broadwell-audio", "intel/IntcSST2.bin", NULL, NULL, NULL }, + { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" }, {} }; @@ -217,8 +217,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { #if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) static struct sst_acpi_mach baytrail_machines[] = { - { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, - { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, {} }; diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index 4dcfb7e..9eb8ade 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -28,9 +28,4 @@ struct sst_acpi_mach { const char *drv_name; /* firmware file name */ const char *fw_filename; - - /* board name */ - const char *board; - void (*machine_quirk)(void); - void *pdata; }; -- 2.7.3

2 2

[alsa-devel] [PATCH v4] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 25 Apr '16

25 Apr '16

Provide *our* view of what the rules are for the different DAI formats, so that we do not have to trust external interpretations for this crucial bit of interoperability. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- Documentation/sound/alsa/soc/clocking.txt | 195 ++++++++++++++++++++++++++++++ 1 file changed, 195 insertions(+) Use rising edge and falling edge (instead of flanks). Also differentiate how transmitters and receivers behave and explain at what edges transmitters change data and at what edges receivers read data. Don't speak about left and right channels for the DSP formats where there are just a number of channels. Cheers, Peter diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index b1300162e01c..7c2bea8124b0 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -47,5 +47,200 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. +The below diagrams all have BCLK as the first signal, LRC as the second signal +and DATA as the third. Below that is an indication about which DATA bits belong +in what channel. + +A "..." marking as DATA indicates that there may be more bits that are not +shown. Also, all DATA bits marked X may or may not be present. DAI +transmitters must add them should LRC not match the word size exactly and DAI +receivers must be prepared to ignore them. DAI transmitters must insert zeros +for I2S and Left Justified modes, and preferably not drive DATA for the DSP +modes for these extra X bits. + + +I2S +--- + +The LRC master should put the edges of LRC so that the LRC slaves can reliably +sample LRC at the rising edges of BCLK, preferably by changing LRC in response +to falling edges of BCLK. + +Transmitters wait for a falling edge of LRC, then skip a rising edge of BCLK, +and start the left word (output MSB) at the following falling edge of BCLK. +Transmitters continue with the rest of the word by changing DATA at the falling +edges to BCLK. Transmitters must pad words with zeros if LRC is not matching +the word size, so that receivers expecting wider words do not get noise. +Transmission of the right word works the same, but is initiated by a rising +edge of LRC instead. + +Receivers wait for a falling edge of LRC, then skip a rising edge of BCLK and +read the MSB of the left word at the following rising edge of BCLK. Receivers +then continue with the rest of the word at the following rising edges of BCLK. +Receivers must ignore the tail if more bits per word are transmitted than they +can use. Reception of the right word works the same, but is initiated by a +rising edge of LRC instead. + +See the official I2S specification +https://www.sparkfun.com/datasheets/BreakoutBoards/I2SBUS.pdf + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified (aka MSB) +------------------------ + +The LRC master should synchronize the LRC clock edges with the falling edges of +BCLK. + +Transmitters start the left word (output MSB) at the rising edge of LRC and +hold MSB over the first rising edge of BCLK. The transmitters then change DATA +at the falling edges of BCLK. Transmitters must pad words with zeros if LRC is +not matching the word size, so that receivers expecting wider words do not get +noise. Transmission of the right word works the same, but is initiated by a +falling edge of LRC instead. + +Receivers wait for a rising edge of LRC and reads the MSB of the left word at +the first rising edge of BCLK after that. Receivers then continue with the rest +of the word at the following rising edges of BCLK. Receivers must ignore the +tail if more bits per word are transmitted than they can use. Reception of the +right word works the same, but is initiated by a falling edge of LRC instead. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Right Justified (aka LSB) +------------------------- + +The LRC master should synchronize the LRC clock edges with the falling edges of +BCLK. + +Transmitters wait for a rising edge of LRC, and then skip the correct number of +rising edges of BCLK. If zero rising edges are skipped, transmitters start the +left word (output MSB) immediately at the rising edge of LRC. If at least one +rising edge of BCLK is skipped, they start the left word at the following +falling edge of BCLK. The correct number of rising edges of BCLK to skip after +the initial rising edge of LRC is such that the LSB is ending at the following +edge of LRC. After the MSB, transmitters continue with the rest of the word at +the falling edges of BCLK. Transmission of the right word works the same, but +is initiated by a falling edge of LRC instead. + +Receivers wait for a rising edge of LRC, and then skip the same number of rising +edges of BCLK as the transmitters, and read the MSB of the left word on the +first rising edge of BCLK after that. Receivers then continue with the rest of +the word at the following rising edges of BCLK. Reception of the right word +works the same, but is initiated by a falling edge of LRC instead. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | X |...| X |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +DSP mode A +---------- +Transmitters wait for a rising edge of LRC, then skip one rising edge of BCLK +and start the first word (output MSB) at the following falling edge of BCLK. +They then change DATA at the falling edges of BCLK. When the first word is +finished, transmitters follow up immediately with further words. + +Receivers wait for a rising edge of LRC, then skip one rising edge of BCLK and +read the MSB of the first word at the following rising edge of BCLK. Receivers +then continue with the rest of the word at the rising edges of BCLK. Further +words are following immediately. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | 1st channel | 2nd channel | | + +LRC examples for DSP mode A + .---. .---. +---' '-------------------------------------------------------' '----- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +DSP mode B +---------- +Transmitters start the first word (output MSB) at the rising edge of LRC, hold +MSB over the first rising edge of BCLK, and continue with the rest of the word +at the following falling edges of BCLK. When the first word is finished, +transmitters follow up immediately with further words. + +Receivers wait for a rising edge of LRC, then read the MSB of the first word at +the following rising edge of BCLK. Receivers then continue with the rest of the +word at the rising edges of BCLK. Further words are following immediately. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | 1st channel | 2nd channel | | + +LRC examples for DSP mode B + .---. .---. +---' '-------------------------------------------------------' '----- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +Inverted Clocks +--------------- + +The above diagrams show normal BCLK and LRC clocking where DAI transmitters +change DATA on the falling edge of BCLK and DAI receivers read DATA on the +rising edge. For inverted BCLK it is naturally the other way around. Inverted +LRC is just that. + +Left Justified, inverted BCLK +----------------------------- + +-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .- + '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified, inverted LRC +---------------------------- + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | -- 2.1.4

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Re: [alsa-devel] [PATCH 2/3] ASoC: simple-card: Add support jack detection via codec
by Mark Brown 25 Apr '16

25 Apr '16

On Fri, Apr 22, 2016 at 09:37:39AM -0700, Jeremy McDermond wrote: > components fairly well. I’m just trying to wrap my head around > where this all would fit in. Should the codec driver just register > a bunch of clock components in its initialization somewhere? > I’m thinking that it can’t have its own CCF driver separate from > the codec driver because you won’t have proper access to the > registers over I2C or SPI. I don’t find a lot of examples in the > codec drivers, and there aren’t very many other drivers in the tree > that seem to register clocks with CCF that I can find. If the clock code is worth splitting off into a separate driver that's what drivers/mfd is for.

2 1

[alsa-devel] Intel SST baytrail audio on PIPO W6
by Bharath Ghanta 25 Apr '16

25 Apr '16

Hi, I am trying to get audio working on PIPO W6 tablet. I was following this thread [1] in my efforts to get audio working. Initially i was not getting any interrupts. I changed the interrupt order as described in the thread and was able to get interrupts and successfully load the firmware. I also ran the amixer commands described in [2]. Now I am just getting a clicking sound whenever something plays. Pierre-Louis said something about playing with the DAI links and swapping the codec connections here [3]. Can anybody explain to me how its done ? I tried to enable other switches in alsamixer using the below commands but nothing worked. amixer -c0 sset 'codec_out1 mix 0 pcm0_in' on I mailed the manufacturer asking for the schematics of the audio chip and the android source code for this device but got no response. I also tried the ssp0 firmware but i got no sound instead of the clicking sound i was getting with the default firmware. I have the exact problem as the guy in this thread. [4] Thanks, Bharath [1] http://thread.gmane.org/gmane.linux.alsa.devel/134554 [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/0940 80.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106 262.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106 178.html

1 0

Re: [alsa-devel] [snd_ali5451] snd_ali5451 0000:00:06.0: BUG: , pos = 16384, buffer size = 16384, period size = 1024
by Takashi Iwai 25 Apr '16

25 Apr '16

On Mon, 25 Apr 2016 16:50:09 +0200, Enrico Mioso wrote: > > Thank you a lot Takashi. > > there are two problems: > 1 - The warning doesn't manifest always: in some cases (instances of mplayer?) this warning manifests itself from time to time, some minutes. Yes, this is likely some XRUN by either hardware boundary condition or a system load. > 2 - I think the audio stops a little bit (less of a second I think), then things go ahead. Still I don't think any message is printed by alsa libraries. > I don't know if this per se indicates there is no overrun, but still my sensation is this. > But this system isn't exactly "extremely fast": so the audio might stop even for other reasons. I'll recompile the kernel to apply this patch and let you know. > The timing will be slow, but I will try. My patch won't improve in this regard, I'm afraid. The PCM core already corrects to the position 0 if such an invalid position is detected, and the patch would do it just implicitly. But, it's still interesting to see whether it has any impact on such behavior. Takashi

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[alsa-devel] [PATCH] ASoC: Make aux_dev more like a generic component
by mengdong.lin＠linux.intel.com 25 Apr '16

25 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> aux_dev is mainly used by the machine driver to specify analog devices, which are registered as codecs. Making it more like a generic component can help the machine driver to use it to specify any component with topology info by name. Details: - Remove the stub 'rtd_aux' array from the soc card. - Add a list 'aux_comp_list' to store the components of aux_devs. And add a list head 'list_aux' to struct snd_soc_component, for adding such components to the above list. - Add a 'init' ops to a component for machine specific init. soc_bind_aux_dev() will set it to be aux_dev's init. And it will be called when probing the component. - soc_bind_aux_dev() will also search components by name of an aux_dev, since it may not be a codec. - Move probing of aux_devs before checking new DAI links brought by topology. - Move removal of aux_devs later than removal of links. Because topology of aux components may register DAIs and the DAI drivers will go with removal of the aux components, we want soc_remove_link_dais() to remove the DAIs at first. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 49011c3..c03e07a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -798,6 +798,7 @@ struct snd_soc_component { unsigned int registered_as_component:1; struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct snd_soc_dai_driver *dai_drv; int num_dai; @@ -841,6 +842,9 @@ struct snd_soc_component { int (*probe)(struct snd_soc_component *); void (*remove)(struct snd_soc_component *); + /* machine specific init */ + int (*init)(struct snd_soc_component *component); + #ifdef CONFIG_DEBUG_FS void (*init_debugfs)(struct snd_soc_component *component); const char *debugfs_prefix; @@ -1144,8 +1148,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; - struct snd_soc_pcm_runtime *rtd_aux; - int num_aux_rtd; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1553,6 +1556,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e03e82..790ee2b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1413,6 +1413,16 @@ static int soc_probe_component(struct snd_soc_card *card, component->name); } + /* machine specific init */ + if (component->init) { + ret = component->init(component); + if (ret < 0) { + dev_err(component->dev, + "Failed to do machine specific init %d\n", ret); + goto err_probe; + } + } + if (component->controls) snd_soc_add_component_controls(component, component->controls, component->num_controls); @@ -1657,65 +1667,81 @@ static int soc_probe_link_dais(struct snd_soc_card *card, static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *name = aux_dev->codec_name; - - rtd->component = soc_find_component(aux_dev->codec_of_node, name); - if (!rtd->component) { - if (aux_dev->codec_of_node) - name = of_node_full_name(aux_dev->codec_of_node); - - dev_err(card->dev, "ASoC: %s not registered\n", name); - return -EPROBE_DEFER; + struct snd_soc_component *component; + const char *name; + struct device_node *codec_of_node; + + if (aux_dev->codec_of_node || aux_dev->codec_name) { + /* codecs, usually analog devices */ + name = aux_dev->codec_name; + codec_of_node = aux_dev->codec_of_node; + component = soc_find_component(codec_of_node, name); + if (!component) { + if (codec_of_node) + name = of_node_full_name(codec_of_node); + goto err_defer; + } + } else if (aux_dev->name) { + /* generic components */ + name = aux_dev->name; + component = soc_find_component(NULL, name); + if (!component) + goto err_defer; + } else { + dev_err(card->dev, "ASoC: Invalid auxiliary device\n"); + return -EINVAL; } - /* - * Some places still reference rtd->codec, so we have to keep that - * initialized if the component is a CODEC. Once all those references - * have been removed, this code can be removed as well. - */ - rtd->codec = rtd->component->codec; - + component->init = aux_dev->init; + list_add(&component->list_aux, &card->aux_comp_list); return 0; + +err_defer: + dev_err(card->dev, "ASoC: %s not registered\n", name); + return -EPROBE_DEFER; } -static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_component *comp; + int order; int ret; - ret = soc_probe_component(card, rtd->component); - if (ret < 0) - return ret; - - /* do machine specific initialization */ - if (aux_dev->init) { - ret = aux_dev->init(rtd->component); - if (ret < 0) { - dev_err(card->dev, "ASoC: failed to init %s: %d\n", - aux_dev->name, ret); - return ret; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { + if (comp->driver->probe_order == order) { + ret = soc_probe_component(card, comp); + if (ret < 0) { + dev_err(card->dev, + "ASoC: failed to probe aux component %s %d\n", + comp->name, ret); + return ret; + } + } } } - return soc_post_component_init(rtd, aux_dev->name); + return 0; } -static void soc_remove_aux_dev(struct snd_soc_card *card, int num) +static void soc_remove_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_component *component = rtd->component; + struct snd_soc_component *comp, *_comp; + int order; - /* unregister the rtd device */ - if (rtd->dev_registered) { - device_unregister(rtd->dev); - rtd->dev_registered = 0; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry_safe(comp, _comp, + &card->aux_comp_list, list_aux) { + if (comp->driver->remove_order == order) { + soc_remove_component(comp); + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); + } + } } - - if (component) - soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -1894,6 +1920,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* probe auxiliary components */ + ret = soc_probe_aux_devices(card); + if (ret < 0) + goto probe_dai_err; + /* Find new DAI links added during probing components and bind them. * Components with topology may bring new DAIs and DAI links. */ @@ -1923,16 +1954,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } - for (i = 0; i < card->num_aux_devs; i++) { - ret = soc_probe_aux_dev(card, i); - if (ret < 0) { - dev_err(card->dev, - "ASoC: failed to add auxiliary devices %d\n", - ret); - goto probe_aux_dev_err; - } - } - snd_soc_dapm_link_dai_widgets(card); snd_soc_dapm_connect_dai_link_widgets(card); @@ -1992,8 +2013,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) return 0; probe_aux_dev_err: - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); + soc_remove_aux_devices(card); probe_dai_err: soc_remove_dai_links(card); @@ -2039,20 +2059,18 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; - int i; /* make sure any delayed work runs */ list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - /* remove auxiliary devices */ - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); - /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + soc_cleanup_card_debugfs(card); /* remove the card */ @@ -2608,16 +2626,6 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; - card->rtd_aux = devm_kzalloc(card->dev, - sizeof(struct snd_soc_pcm_runtime) * - card->num_aux_devs, - GFP_KERNEL); - if (card->rtd_aux == NULL) - return -ENOMEM; - - for (i = 0; i < card->num_aux_devs; i++) - card->rtd_aux[i].card = card; - INIT_LIST_HEAD(&card->dapm_dirty); INIT_LIST_HEAD(&card->dobj_list); card->instantiated = 0; -- 2.5.0

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[alsa-devel] Broken IRQs on BayTrail devices
by Joel Holdsworth 25 Apr '16

25 Apr '16

Dear All, I've been working on a kernel patch for a firmware IRQ quirk issue that affects multiple Intel Atom Bay-Trail boards: https://bugzilla.kernel.org/show_bug.cgi?id=117141 If you have a Bay Trail device with non-functioning sound, can you attach the DSDT and dmidecode output to the bug so I can investigate whether the quirk affect your device. Best Regards Joel Holdsworth

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[alsa-devel] [PATCH 0/3] More robust i915 / audio binding
by Takashi Iwai 25 Apr '16

25 Apr '16

Hi, here is a patchset to make the i915 / audio binding more robustly. It's based on Imre's work to implement the stub component ops for notifying the i915 disablement. Takashi === Imre Deak (1): drm/i915/hda: Add audio component stub Takashi Iwai (2): ALSA: hda - Immediately fail if the i915 audio component is disabled ALSA: hda - Support deferred i915 audio component binding drivers/gpu/drm/i915/i915_drv.c | 34 +++++++++++++-------- drivers/gpu/drm/i915/intel_audio.c | 61 ++++++++++++++++++++++++++++++++++++++ drivers/gpu/drm/i915/intel_drv.h | 2 ++ include/drm/i915_component.h | 5 ++++ sound/hda/hdac_i915.c | 28 ++++++++++++----- 5 files changed, 110 insertions(+), 20 deletions(-) -- 2.8.1

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[alsa-devel] [RFC][PATCH 0/6] ALSA: firewire: drop reusing incoming packet parameters for outgoing packet parameters
by Takashi Sakamoto 25 Apr '16

25 Apr '16

Hi, The purpose of this patchset is to remove a feature of firewire-lib module. I call it as 'full duplex streaming with timestamp synchronization'. This patchset should accumulated into my previous RFC patchset below: [alsa-devel] [RFC][PATCH 0/3 v2] firewire-lib: support Linux tracing framework to dump a part of packet data http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106950.html The feature is based on a specification of IEC 61883-6. In the specification, an audio function block should reuse some parameters of incoming packets for outgoing packets. Current ALSA firewire stack partly implement it. However, actual audio and music units on IEEE 1394 bus don't expect drivers to perform the specification. Additionally to the fact, the feature brings inconveniences to handle payloads of isochronous packet in software interrupts. For these reasons, this patchset removes the feature from ALSA firewire stack. This decision may be a help for future improvement about dynamic interrupt interval of isochronous context, about which Clemens have mentioned[1]. [1][alsa-devel] For firewire drivers, reduce period time down to 2msec? http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075696.html Regards Takashi Sakamoto (6): ALSA: bebob: drop reusing incoming packet parameter for outgoing packet parameter ALSA: fireworks: drop reusing incoming packet parameter for ougoing packet parameter ALSA: firewire-tascam: drop reusing incoming packet parameter for outgoing packet parameter ALSA: firewire-lib: handle IT/IR contexts in each software context ALSA: firewire-lib: code cleanup for incoming packet handling ALSA: firewire-lib: code cleanup for outgoing packet handling sound/firewire/amdtp-stream.c | 84 ++++++++---------------- sound/firewire/amdtp-stream.h | 35 ++-------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/bebob/bebob_stream.c | 99 ++++++++--------------------- sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_stream.c | 84 +++++------------------- sound/firewire/tascam/tascam-stream.c | 26 ++++---- 7 files changed, 87 insertions(+), 243 deletions(-) -- 2.7.4

1 6

[alsa-devel] [PATCH 0/7] ALSA: firewire: apply deleyed registration to all drivers
by Takashi Sakamoto 25 Apr '16

25 Apr '16

Hi, Commit b59fb1900b4fee('ALSA: dice: postpone card registration') applied delayed registration[1] of sound card instance to ALSA dice driver. This idea is also nice for drivers in ALSA firewire stack because in IEEE 1394 bus adding units is corresponding to bus reset and the actual units tend not to work well during bus-reset state. This patchset adds a new function to firewire-lib module to schedule a delayed work for sound card registration. Currently, the work is commonly scheduled 2 seconds after time of the last bus-reset recorded by Linux firewire core. Then, take all drivers to use it as Dice driver does. Related to this patchset, I'd like to cancel a discussion about my proposed patchset to remove unpractical module parameters from ALSA bebob/fireworks[2]. I'm sorry for reviewers but my work priority to this patchset is prior to the former one. [1]: [alsa-devel] [PATCH 0/4 v5] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102562.h… [2]: [alsa-devel] [PATCH 0/2] ALSA: bebob/fireworks: remove module parameters to reduce maintenance work http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106219.html Regards Takashi Sakamoto (8): ALSA: firewire-lib: suppress kernel warnings when releasing uninitialized stream data ALSA: dice: simplify unit probe processing ALSA: firewire-lib: add new function to schedule a work for sound card registration ALSA: bebob: delayed registration of sound card ALSA: fireworks: delayed registration of sound card ALSA: oxfw: delayed registration of sound card ALSA: firewire-digi00x: delayed registration of sound card ALSA: firewire-tascam: deleyed registration of sound card sound/firewire/amdtp-stream.c | 4 + sound/firewire/bebob/bebob.c | 215 ++++++++++++++++----------- sound/firewire/bebob/bebob.h | 5 +- sound/firewire/dice/dice.c | 34 +---- sound/firewire/digi00x/digi00x-transaction.c | 7 +- sound/firewire/digi00x/digi00x.c | 107 +++++++++---- sound/firewire/digi00x/digi00x.h | 3 + sound/firewire/fireworks/fireworks.c | 147 ++++++++++++------ sound/firewire/fireworks/fireworks.h | 3 + sound/firewire/lib.c | 32 ++++ sound/firewire/lib.h | 3 + sound/firewire/oxfw/oxfw.c | 151 ++++++++++++------- sound/firewire/oxfw/oxfw.h | 3 + sound/firewire/tascam/tascam.c | 118 ++++++++++----- sound/firewire/tascam/tascam.h | 2 + 15 files changed, 551 insertions(+), 283 deletions(-) -- 2.7.4

2 12

[alsa-devel] [PATCH 36/41] Documentation: sound: fix spelling mistakes
by Eric Engestrom 25 Apr '16

25 Apr '16

Signed-off-by: Eric Engestrom <eric(a)engestrom.ch> --- Documentation/sound/alsa/compress_offload.txt | 2 +- Documentation/sound/alsa/soc/dapm.txt | 2 +- Documentation/sound/alsa/soc/overview.txt | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 630c492..81476b4 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -149,7 +149,7 @@ Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder delay and padding and directly move from one track content to another. The end -user can perceive this as gapless playback as we dont have silence while +user can perceive this as gapless playback as we don't have silence while switching from one track to another Also, there might be low-intensity noises due to encoding. Perfect gapless is diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 6faab48..c45bd79 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, ARRAY_SIZE(wm8731_output_mixer_controls)), -If you dont want the mixer elements prefixed with the name of the mixer widget, +If you don't want the mixer elements prefixed with the name of the mixer widget, you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same as for SND_SOC_DAPM_MIXER. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index ff88f52..f3f28b7 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -63,7 +63,7 @@ multiple re-usable component drivers :- and any audio DSP drivers for that platform. * Machine class driver: The machine driver class acts as the glue that - decribes and binds the other component drivers together to form an ALSA + describes and binds the other component drivers together to form an ALSA "sound card device". It handles any machine specific controls and machine level audio events (e.g. turning on an amp at start of playback). -- 2.8.0

3 2

[alsa-devel] [PATCH] ALSA: timer: remove legacy rtctimer
by Alexandre Belloni 25 Apr '16

25 Apr '16

There are no users of rtctimer left. Remove its code as this is the in-kernel user of the legacy PC RTC driver that will hopefully be removed at some point. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Note that I've kept the definition of SNDRV_TIMER_GLOBAL_RTC as it is exposed to userspace. USerspace will still properly get -ENODEV when trying to use it (as if it was compiled out of the kernel anyway). include/uapi/sound/asound.h | 2 +- sound/core/Kconfig | 29 ------- sound/core/Makefile | 1 - sound/core/rtctimer.c | 187 -------------------------------------------- sound/core/seq/seq.c | 2 - sound/core/timer.c | 2 - 6 files changed, 1 insertion(+), 222 deletions(-) delete mode 100644 sound/core/rtctimer.c diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 67bf49d8c944..609cadb8739d 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -672,7 +672,7 @@ enum { /* global timers (device member) */ #define SNDRV_TIMER_GLOBAL_SYSTEM 0 -#define SNDRV_TIMER_GLOBAL_RTC 1 +#define SNDRV_TIMER_GLOBAL_RTC 1 /* unused */ #define SNDRV_TIMER_GLOBAL_HPET 2 #define SNDRV_TIMER_GLOBAL_HRTIMER 3 diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6d12ca9bcb80..9749f9e8b45c 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT Say Y here to use the HR-timer backend as the default sequencer timer. -config SND_RTCTIMER - tristate "RTC Timer support" - depends on RTC - select SND_TIMER - help - Say Y here to enable RTC timer support for ALSA. ALSA uses - the RTC timer as a precise timing source and maps the RTC - timer to ALSA's timer interface. The ALSA sequencer code also - can use this timing source. - - To compile this driver as a module, choose M here: the module - will be called snd-rtctimer. - - Note that this option is exclusive with the new RTC drivers - (CONFIG_RTC_CLASS) since this requires the old API. - -config SND_SEQ_RTCTIMER_DEFAULT - bool "Use RTC as default sequencer timer" - depends on SND_RTCTIMER && SND_SEQUENCER - depends on !SND_SEQ_HRTIMER_DEFAULT - default y - help - Say Y here to use the RTC timer as the default sequencer - timer. This is strongly recommended because it ensures - precise MIDI timing even when the system timer runs at less - than 1000 Hz. - - If in doubt, say Y. - config SND_DYNAMIC_MINORS bool "Dynamic device file minor numbers" help diff --git a/sound/core/Makefile b/sound/core/Makefile index 48ab4b8f8279..e85d9dd12c2d 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -37,7 +37,6 @@ obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o -obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c deleted file mode 100644 index f3420d11a12f..000000000000 --- a/sound/core/rtctimer.c +++ /dev/null @@ -1,187 +0,0 @@ -/* - * RTC based high-frequency timer - * - * Copyright (C) 2000 Takashi Iwai - * based on rtctimer.c by Steve Ratcliffe - * - * 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. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/log2.h> -#include <sound/core.h> -#include <sound/timer.h> - -#if IS_ENABLED(CONFIG_RTC) - -#include <linux/mc146818rtc.h> - -#define RTC_FREQ 1024 /* default frequency */ -#define NANO_SEC 1000000000L /* 10^9 in sec */ - -/* - * prototypes - */ -static int rtctimer_open(struct snd_timer *t); -static int rtctimer_close(struct snd_timer *t); -static int rtctimer_start(struct snd_timer *t); -static int rtctimer_stop(struct snd_timer *t); - - -/* - * The hardware dependent description for this timer. - */ -static struct snd_timer_hardware rtc_hw = { - .flags = SNDRV_TIMER_HW_AUTO | - SNDRV_TIMER_HW_FIRST | - SNDRV_TIMER_HW_TASKLET, - .ticks = 100000000L, /* FIXME: XXX */ - .open = rtctimer_open, - .close = rtctimer_close, - .start = rtctimer_start, - .stop = rtctimer_stop, -}; - -static int rtctimer_freq = RTC_FREQ; /* frequency */ -static struct snd_timer *rtctimer; -static struct tasklet_struct rtc_tasklet; -static rtc_task_t rtc_task; - - -static int -rtctimer_open(struct snd_timer *t) -{ - int err; - - err = rtc_register(&rtc_task); - if (err < 0) - return err; - t->private_data = &rtc_task; - return 0; -} - -static int -rtctimer_close(struct snd_timer *t) -{ - rtc_task_t *rtc = t->private_data; - if (rtc) { - rtc_unregister(rtc); - tasklet_kill(&rtc_tasklet); - t->private_data = NULL; - } - return 0; -} - -static int -rtctimer_start(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq); - rtc_control(rtc, RTC_PIE_ON, 0); - return 0; -} - -static int -rtctimer_stop(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_PIE_OFF, 0); - return 0; -} - -static void rtctimer_tasklet(unsigned long data) -{ - snd_timer_interrupt((struct snd_timer *)data, 1); -} - -/* - * interrupt - */ -static void rtctimer_interrupt(void *private_data) -{ - tasklet_schedule(private_data); -} - - -/* - * ENTRY functions - */ -static int __init rtctimer_init(void) -{ - int err; - struct snd_timer *timer; - - if (rtctimer_freq < 2 || rtctimer_freq > 8192 || - !is_power_of_2(rtctimer_freq)) { - pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq); - return -EINVAL; - } - - /* Create a new timer and set up the fields */ - err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer); - if (err < 0) - return err; - - timer->module = THIS_MODULE; - strcpy(timer->name, "RTC timer"); - timer->hw = rtc_hw; - timer->hw.resolution = NANO_SEC / rtctimer_freq; - - tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer); - - /* set up RTC callback */ - rtc_task.func = rtctimer_interrupt; - rtc_task.private_data = &rtc_tasklet; - - err = snd_timer_global_register(timer); - if (err < 0) { - snd_timer_global_free(timer); - return err; - } - rtctimer = timer; /* remember this */ - - return 0; -} - -static void __exit rtctimer_exit(void) -{ - if (rtctimer) { - snd_timer_global_free(rtctimer); - rtctimer = NULL; - } -} - - -/* - * exported stuff - */ -module_init(rtctimer_init) -module_exit(rtctimer_exit) - -module_param(rtctimer_freq, int, 0444); -MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz"); - -MODULE_LICENSE("GPL"); - -MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); - -#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 7e0aabb808a6..639544b4fb04 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -47,8 +47,6 @@ int seq_default_timer_card = -1; int seq_default_timer_device = #ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT SNDRV_TIMER_GLOBAL_HRTIMER -#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT) - SNDRV_TIMER_GLOBAL_RTC #else SNDRV_TIMER_GLOBAL_SYSTEM #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6469bedda2f3..0cfc028c1193 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -37,8 +37,6 @@ #if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif IS_ENABLED(CONFIG_SND_RTCTIMER) -#define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 #endif -- 2.8.0.rc3

2 1

[alsa-devel] [PATCH 0/3] Add support simple-card parse jack detection via external codec
by Xing Zheng 22 Apr '16

22 Apr '16

Hi, In most cases, many codecs already supports jack detection, previouslly, we need to create a customized machine driver every time. Hence, the simple-card need to support use them dynamically via parse dts file for better flexibility. Xing Zheng (3): ASoC: jack: Add a jack detect callback via codec ASoC: simple-card: Add support jack detection via codec ASoC: da7219: Add detect_jack callback in the snd_soc_codec_driver .../devicetree/bindings/sound/simple-card.txt | 17 +++ include/sound/soc.h | 6 + sound/soc/codecs/da7219.c | 2 + sound/soc/generic/simple-card.c | 126 +++++++++++++++++++- sound/soc/soc-jack.c | 8 ++ 5 files changed, 156 insertions(+), 3 deletions(-) -- 1.7.9.5

5 14

[alsa-devel] [PATCH] ASoC: da7218: Update PLL ranges and dividers to improve locking
by Adam Thomson 22 Apr '16

22 Apr '16

The expected MCLK frequency ranges and the associated dividers are updated to improve PLL locking in a corner scenario, with low MCLK frequency near an input divider change boundary. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on v4.6-rc1 kernel version sound/soc/codecs/da7218.c | 32 ++++++++++++++++---------------- sound/soc/codecs/da7218.h | 21 ++++++++++++--------- 2 files changed, 28 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 93575f2..99ce23e 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ if (da7218->mclk_rate == 32768) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; } else if (da7218->mclk_rate < 2000000) { dev_err(codec->dev, "PLL input clock %d below valid range\n", da7218->mclk_rate); return -EINVAL; - } else if (da7218->mclk_rate <= 5000000) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 10000000) { - indiv_bits = DA7218_PLL_INDIV_5_10_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 20000000) { - indiv_bits = DA7218_PLL_INDIV_10_20_MHZ; - indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7218->mclk_rate <= 40000000) { - indiv_bits = DA7218_PLL_INDIV_20_40_MHZ; - indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7218->mclk_rate <= 4500000) { + indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7218->mclk_rate <= 9000000) { + indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7218->mclk_rate <= 18000000) { + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7218->mclk_rate <= 36000000) { + indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7218->mclk_rate <= 54000000) { - indiv_bits = DA7218_PLL_INDIV_40_54_MHZ; - indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7218->mclk_rate); diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h index c2c5904..477cd37 100644 --- a/sound/soc/codecs/da7218.h +++ b/sound/soc/codecs/da7218.h @@ -876,15 +876,11 @@ /* DA7218_PLL_CTRL = 0x91 */ #define DA7218_PLL_INDIV_SHIFT 0 #define DA7218_PLL_INDIV_MASK (0x7 << 0) -#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0) -#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0) -#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0) -#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0) -#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0) -#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2 -#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 0) +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 0) +#define DA7218_PLL_INDIV_9_TO_18_MHZ (0x2 << 0) +#define DA7218_PLL_INDIV_18_TO_36_MHZ (0x3 << 0) +#define DA7218_PLL_INDIV_36_TO_54_MHZ (0x4 << 0) #define DA7218_PLL_MCLK_SQR_EN_SHIFT 4 #define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4) #define DA7218_PLL_MODE_SHIFT 6 @@ -1336,6 +1332,13 @@ #define DA7218_PLL_FREQ_OUT_90316 90316800 #define DA7218_PLL_FREQ_OUT_98304 98304000 +/* PLL Frequency Dividers */ +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL 16 + /* ALC Calibration */ #define DA7218_ALC_CALIB_DELAY_MIN 2500 #define DA7218_ALC_CALIB_DELAY_MAX 5000 -- 1.9.3

2 1

[alsa-devel] [PATCH 0/4] ASoC: simple-card/davinci-mcasp: System clock configuration support
by Peter Ujfalusi 22 Apr '16

22 Apr '16

Hi, Currently davinci-mcasp is hardwired for two configuration regarding to reference clock: McASP is master: AHCLKX is selected as source of the reference clock used to generate the FS and BCLK McASP is slave: AHCLKX is selected as output and the default AUXCLK is routed there. This works in most of the boards, but not all configuration can be supported with this limitation. McASP uses it's internal AUXCLK as a reference clock. AUXCLK can be sourced from the external AHCLKX pin or from internal functional clock. If the board uses McASP as master and AHCLKX is not receiving clock, we need to be able to select the internal clock source to be able to generate the FS and BCLK. Since we try to move all of our boards to simple-card instead of the davinci-evm custom machine driver, I have included Jyri's two patch to add the needed support for it to be able to select clocks and directions. I remember the discussion regarding to clock IDs, and to try to move to common clock framework. I have taken a look at that path and I'm not convinced that by trying to add ASoC level common clock handling will simplify or makes things cleaner/simpler. Regards, Peter --- Jyri Sarha (2): ASoC: simple-card: Add system-clock-direction DT parameter to dai nodes ASoC: simple-card: Support for selecting system clocks by ID Peter Ujfalusi (2): ASoC: davinci-mcasp: Use defines for clkdiv IDs via DT binding header ASoC: davinci-mcasp: Improve the sysclk selection .../devicetree/bindings/sound/simple-card.txt | 3 ++ include/dt-bindings/sound/ti-mcasp.h | 13 +++++++ include/sound/simple_card.h | 2 ++ sound/soc/davinci/davinci-mcasp.c | 42 ++++++++++++++++------ sound/soc/generic/simple-card.c | 17 ++++++++- 5 files changed, 66 insertions(+), 11 deletions(-) create mode 100644 include/dt-bindings/sound/ti-mcasp.h -- 2.7.1

7 20

[alsa-devel] [PATCH] ASoC: Intel: Atom: fix boot warning
by Vinod Koul 22 Apr '16

22 Apr '16

Users have reported seeing this false warning on atom driver [ 5.647469] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.661612] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.661646] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.661681] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.661708] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.661738] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.661771] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.661807] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! This is caused when check for control is not being associated with a dapm widget, but the check is wrong as the else case maybe triggered when widget is not powered up, so we should check if widget is associated before printing this message. Tested-by: Sandeep Tayal <sandeepx.tayal(a)intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index b97e6adcf1b2..98720a93de8a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol if (e->w && e->w->power) ret = sst_send_slot_map(drv); - else + else if (!e->w) dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n", kcontrol->id.name); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tidyup alphabetical order for SND_SOC_Bxx
by Kuninori Morimoto 22 Apr '16

22 Apr '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 907c804..7d2d994 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C + select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C select SND_SOC_CS42L51_I2C if I2C @@ -64,7 +65,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC - select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_GTM601 @@ -365,6 +365,9 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate +config SND_SOC_BT_SCO + tristate + config SND_SOC_CQ0093VC tristate @@ -471,9 +474,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_BT_SCO - tristate - config SND_SOC_DMIC tristate -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: Add kerneldoc comments for snd_soc_find_dai
by mengdong.lin＠linux.intel.com 22 Apr '16

22 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> snd_soc_find_dai() has been exported and so add the kerneldoc comments for it. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 07663de..16369ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -930,6 +930,17 @@ static struct snd_soc_component *soc_find_component( return NULL; } +/** + * snd_soc_find_dai - Find a registered DAI + * + * @dlc: name of the DAI and optional component info to match + * + * This function will search all regsitered components and their DAIs to + * find the DAI of the same name. The component's of_node and name + * should also match if being specified. + * + * Return: pointer of DAI, or NULL if not found. + */ struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc) { -- 2.5.0

2 1

[alsa-devel] [PATCH v3 0/7] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 22 Apr '16

22 Apr '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 3 of the patch: * remove the index note from the "mpu" property in the binding * describe exactly the supported platforms in compatible string * remove unused platform data from davinci-i2s driver and the related EVK board drivers

3 10

[alsa-devel] [PATCH v3 0/3] conf: topology: Add the conf for SKL-rt286
by Vinod Koul 22 Apr '16

22 Apr '16

This series adds SKL-i2c conf for simple PCM topology graph Changes in v3: - remove bin blobs, now they are generated in install step - hook the tool compilation to alsa-lib and run it on install - add u8 in alsa type_compat.h Shreyas NC (3): conf: topology: Add Skylake i2s conf Add u8 in type_compat.h conf: topology: Generate Private data binary blobs configure.ac | 2 + include/sound/type_compat.h | 2 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 11 + src/conf/topology/sklrt286/data/Makefile.am | 6 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_local.h | 9 + .../topology/sklrt286/data/skl-tplg-interface.h | 232 +++ src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 10 files changed, 2510 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_local.h create mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 1.9.1

2 13

[alsa-devel] [PATCH v2 0/2] ASoC: Intel: Skylake: Add support for firmware manifest
by Vinod Koul 22 Apr '16

22 Apr '16

In the DSP firmware, module params like module_id and loadable flags can be modified. So, it is required to read these parameters in the driver from the firmware manifest. Changes in v2: - remove strings and use UUID in bin format Shreyas NC (2): ASoC: Intel: Skylake: Use UUID in binary format ASoC: Intel: Skylake: Add DSP firmware manifest parsing sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 105 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 107 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 2 +- sound/soc/intel/skylake/skl-sst-ipc.h | 4 + sound/soc/intel/skylake/skl-sst.c | 20 ++++- sound/soc/intel/skylake/skl-topology.c | 11 ++- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl-tplg-interface.h | 2 +- 9 files changed, 243 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h -- 1.9.1

2 5

[alsa-devel] A&H Zed R16 not completely working (DICE Jr)
by Allan Klinbail 21 Apr '16

21 Apr '16

Hi, I thought I would try the DICE chipset driver from ALSA, as it appears that this will eventually take over from ffado.. It appears that it doesn't work very well for my Allen & Heath Zed -R16 which works very well under FFADO Using kernel 4.4.0 from Ubuntu The ALSA driver limited the device to only 16 inputs and outputs, however it has a total capability of 26 This is comprises of 16 inputs and outputs on the desk channels, a stereo main pair of outputs and inputs for the master bus 8 additional ins and outs to support the ADAT ports on the back (availoable in 44.1 and 48 khz modes, but not 88.2 or 96 Khz Similiarly I was not able to set the device to run at a latency lower than 256 frames in jack, however in FFADO I could go as low as 64 frames. Please let me know if I can help with any further testing to get this to the same level as the FFADO driver. Regards Allan

2 11

[alsa-devel] [PATCH v6 00/10] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 21 Apr '16

21 Apr '16

these patches add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. This version fixes the hdmi-codec notification patch and splits the jack support out of the ASoC machine driver patches. Changes since v5: - select SND_SOC_HDMI_CODEC and HDMI_NOTIFIERS where appropriate - split out jack support from machine driver patches - fix include of hdmi-notifier.h in hdmi-codec notification support - Drop applied patch: commit 05f5afd3f0d9 ("ASoC: mediatek: address dai link array entries by enum") The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/8534251/ ("drm/mediatek: Add HDMI support"), and need this patch: [2] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") The asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (8): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: hdmi-codec: Add ELD control drm/mediatek: hdmi: use helper function for N and CTS calculation ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 148 +++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 89 +++++----- drivers/gpu/drm/mediatek/mtk_hdmi.h | 10 +- drivers/gpu/drm/mediatek/mtk_hdmi_hw.c | 179 ++++----------------- drivers/gpu/drm/mediatek/mtk_hdmi_hw.h | 4 +- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 +++++++ include/linux/hdmi-notifier.h | 44 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 123 +++++++++++++- sound/soc/mediatek/Kconfig | 2 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 48 ++++++ sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++ 19 files changed, 576 insertions(+), 212 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.7.0

2 11

[alsa-devel] [PATCH 0/2] ASoC: tlv320aic32x4: Better model the input mixers
by Jeremy McDermond 21 Apr '16

21 Apr '16

The input mixers have resistors on all the input lines as well as an additional line that crosses between right and left pins. This patch enables those routings within the driver to provide more flexibility for different board configurations. These settings are documented on pages 135 and 136 of TI's SLAA557 document available at http://www.ti.com/lit/ml/slaa557/slaa557.pdf. This patch might introduce some breakage for current users because it changes some mixer routing settings from a straight switch to an enum for the different resistor values instead. I'm not sure there's anything we can do in code to mitigate these effects, but am open to requests for changes if there's something I'm not aware of. Jeremy McDermond (2): ASoC: tlv320aic32x4: Implement resistors on input pins ASoC: tlv320aic32x4: Add additional input pins sound/soc/codecs/tlv320aic32x4.c | 38 ++++++++++++++++++++++++++++++++------ 1 file changed, 32 insertions(+), 6 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH] ASoC: arizona: Prefer lower FRATIO in pseudo-fractional mode
by Richard Fitzgerald 21 Apr '16

21 Apr '16

When setting up an FLL in pseudo-fractional mode it is preferred to use a lower FRATIO if possible to give a higher reference clock frequency. This patch swaps the two loops in arizona_calc_fratio() so that lower FRATIOs are tried first. The decrementing loop is also changed to start from init_ratio because the original settings might already give a fractional value for N.K Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0cd27e0..d1e1127 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2039,7 +2039,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll, init_ratio, Fref, refdiv); while (div <= ARIZONA_FLL_MAX_REFDIV) { - for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + /* start from init_ratio because this may already give a + * fractional N.K + */ + for (ratio = init_ratio; ratio > 0; ratio--) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); + return ratio; + } + } + + for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / (fll->vco_mult * ratio) < Fref) { @@ -2065,17 +2079,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio > 0; ratio--) { - if (target % (ratio * Fref)) { - cfg->refdiv = refdiv; - cfg->fratio = ratio - 1; - arizona_fll_dbg(fll, - "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", - Fref, refdiv, div, ratio); - return ratio; - } - } - div *= 2; Fref /= 2; refdiv++; -- 1.9.1

2 1

[alsa-devel] [PATCH v8] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 21 Apr '16

21 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v7: Change the code for compatibility Changes since v6: Use subnode to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 ++++ sound/soc/mediatek/mt8173-rt5650.c | 50 ++++++++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..5bfa6b6 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,21 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + codec-capture { + sound-dai = <&rt5650 1>; + }; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..a27a667 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled rt5645 AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 1

[alsa-devel] [sound:topic/hdmi 34/34] include/sound/hda_i915.h:29:1: error: expected identifier or '(' before '{' token
by kbuild test robot 21 Apr '16

21 Apr '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hdmi head: e8de8fa72b640de1e30dcf62bab1a7b012d48cc2 commit: e8de8fa72b640de1e30dcf62bab1a7b012d48cc2 [34/34] ALSA: hda - Move i915 HSW/BDW BCLK setup to a common function config: i386-randconfig-i0-201616 (attached as .config) reproduce: git checkout e8de8fa72b640de1e30dcf62bab1a7b012d48cc2 # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): In file included from sound/pci/hda/patch_hdmi.c:41:0: >> include/sound/hda_i915.h:29:1: error: expected identifier or '(' before '{' token { ^ >> include/sound/hda_i915.h:28:20: warning: 'snd_hdac_i915_set_bclk' used but never defined static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus); ^ vim +29 include/sound/hda_i915.h 98d8fc6c Mengdong Lin 2015-05-19 22 return 0; 98d8fc6c Mengdong Lin 2015-05-19 23 } 98d8fc6c Mengdong Lin 2015-05-19 24 static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) 98d8fc6c Mengdong Lin 2015-05-19 25 { 98d8fc6c Mengdong Lin 2015-05-19 26 return 0; 98d8fc6c Mengdong Lin 2015-05-19 27 } e8de8fa7 Takashi Iwai 2016-04-21 @28 static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus); 98d8fc6c Mengdong Lin 2015-05-19 @29 { 98d8fc6c Mengdong Lin 2015-05-19 30 } d745f5e7 Takashi Iwai 2016-03-21 31 static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, d745f5e7 Takashi Iwai 2016-03-21 32 hda_nid_t nid, int rate) :::::: The code at line 29 was first introduced by commit :::::: 98d8fc6c5d3652e91c61d78941e0fa6f94771d67 ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda :::::: TO: Mengdong Lin <mengdong.lin(a)intel.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix possible race on regmap bypass flip
by Takashi Iwai 21 Apr '16

21 Apr '16

HD-audio driver uses regmap cache bypass feature for reading a raw value without the cache. But this is racy since both the cached and the uncached reads may occur concurrently. The former is done via the normal control API access while the latter comes from the proc file read. Even though the regmap itself has the protection against the concurrent accesses, the flag set/reset is done without the protection, so it may lead to inconsistent state of bypass flag that doesn't match with the current read and occasionally result in a kernel WARNING like: WARNING: CPU: 3 PID: 2731 at drivers/base/regmap/regcache.c:499 regcache_cache_only+0x78/0x93 One way to work around such a problem is to wrap with a mutex. But in this case, the solution is simpler: for the uncached read, we just skip the regmap and directly calls its accessor. The verb execution there is protected by itself, so basically it's safe to call individually. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116171 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 2 ++ sound/hda/hdac_device.c | 10 ++++------ sound/hda/hdac_regmap.c | 40 ++++++++++++++++++++++++++++------------ 3 files changed, 34 insertions(+), 18 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 2767c55a641e..ca64f0f50b45 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -17,6 +17,8 @@ int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec, unsigned int verb); int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, unsigned int *val); +int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, + unsigned int reg, unsigned int *val); int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, unsigned int val); int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index d1a4d6973330..03c9872c31cf 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -299,13 +299,11 @@ EXPORT_SYMBOL_GPL(_snd_hdac_read_parm); int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid, int parm) { - int val; + unsigned int cmd, val; - if (codec->regmap) - regcache_cache_bypass(codec->regmap, true); - val = snd_hdac_read_parm(codec, nid, parm); - if (codec->regmap) - regcache_cache_bypass(codec->regmap, false); + cmd = snd_hdac_regmap_encode_verb(nid, AC_VERB_PARAMETERS) | parm; + if (snd_hdac_regmap_read_raw_uncached(codec, cmd, &val) < 0) + return -1; return val; } EXPORT_SYMBOL_GPL(snd_hdac_read_parm_uncached); diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index bdbcd6b75ff6..87041ddd29cb 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -453,14 +453,30 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, EXPORT_SYMBOL_GPL(snd_hdac_regmap_write_raw); static int reg_raw_read(struct hdac_device *codec, unsigned int reg, - unsigned int *val) + unsigned int *val, bool uncached) { - if (!codec->regmap) + if (uncached || !codec->regmap) return hda_reg_read(codec, reg, val); else return regmap_read(codec->regmap, reg, val); } +static int __snd_hdac_regmap_read_raw(struct hdac_device *codec, + unsigned int reg, unsigned int *val, + bool uncached) +{ + int err; + + err = reg_raw_read(codec, reg, val, uncached); + if (err == -EAGAIN) { + err = snd_hdac_power_up_pm(codec); + if (!err) + err = reg_raw_read(codec, reg, val, uncached); + snd_hdac_power_down_pm(codec); + } + return err; +} + /** * snd_hdac_regmap_read_raw - read a pseudo register with power mgmt * @codec: the codec object @@ -472,19 +488,19 @@ static int reg_raw_read(struct hdac_device *codec, unsigned int reg, int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, unsigned int *val) { - int err; - - err = reg_raw_read(codec, reg, val); - if (err == -EAGAIN) { - err = snd_hdac_power_up_pm(codec); - if (!err) - err = reg_raw_read(codec, reg, val); - snd_hdac_power_down_pm(codec); - } - return err; + return __snd_hdac_regmap_read_raw(codec, reg, val, false); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_read_raw); +/* Works like snd_hdac_regmap_read_raw(), but this doesn't read from the + * cache but always via hda verbs. + */ +int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, + unsigned int reg, unsigned int *val) +{ + return __snd_hdac_regmap_read_raw(codec, reg, val, true); +} + /** * snd_hdac_regmap_update_raw - update a pseudo register with power mgmt * @codec: the codec object -- 2.8.1

1 0

[alsa-devel] [PATCH] ALSA: pcxhr: Fix missing mutex unlock
by Takashi Iwai 21 Apr '16

21 Apr '16

The commit [9bef72bdb26e: ALSA: pcxhr: Use nonatomic PCM ops] converted to non-atomic PCM ops, but shamelessly with an unbalanced mutex locking, which leads to the hangup easily. Fix it. Fixes: 9bef72bdb26e ('ALSA: pcxhr: Use nonatomic PCM ops') Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116441 Cc: <stable(a)vger.kernel.org> # 3.18+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/pcxhr/pcxhr_core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index c5194f5b150a..d7e71f309299 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1341,5 +1341,6 @@ irqreturn_t pcxhr_threaded_irq(int irq, void *dev_id) } pcxhr_msg_thread(mgr); + mutex_unlock(&mgr->lock); return IRQ_HANDLED; } -- 2.8.1

1 0

[alsa-devel] [PATCH v2 0/6] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 21 Apr '16

21 Apr '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 2 of the patch: * add missing TC channel in dmas properties (for compatibility with the new EDMA3 binding) * remove "-audio" postfix from the compatible string * remove "channel-combine" property * of_match_table renamed consistently with the rest of the file (davinci_i2s_match) * devicetree DMA configuration in probe simplified

4 15

[alsa-devel] [PATCH 0/2] Yet more hardening of ALSA hrtimer races
by Takashi Iwai 21 Apr '16

21 Apr '16

Hi, this is a resubmission of patches for hardening the ALSA hrtimer backend against open/close races. Takashi === Takashi Iwai (2): ALSA: timer: Allow backend disabling start/stop from handler ALSA: hrtimer: Use manual start/stop in callback include/sound/timer.h | 12 +++++++++++- sound/core/hrtimer.c | 51 ++++++++++++++++++++++++++++++++++++++------------- sound/core/timer.c | 24 ++++++++++++++++++------ 3 files changed, 67 insertions(+), 20 deletions(-) -- 2.8.1

1 2

[alsa-devel] [PATCH v3 0/2] sti: add audio interface to the hdmi driver
by Arnaud Pouliquen 21 Apr '16

21 Apr '16

This patchset implements audio interface in HDMI drm driver. Implementation is based on ASoC generic hdmi codec driver( https://patchwork.kernel.org/patch/8713141/). It also proposes helper functions to compute N and CTS parameters according to HDMI 1.4b specification. V3: - video: hdmi: add helper function for N and CTS Also used on Mediatek platform (https://patchwork.kernel.org/patch/8887341) delta vs V2: - typo fixes - if/else code optimisation - drm: sti: Add ASoC generic hdmi codec support. - typo fixes - add audio registers in debugfs information V2: RFC https://patchwork.kernel.org/patch/8091531/("video: hdmi: add helper function for N and CTS") https://patchwork.kernel.org/patch/8091561/("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") - patch: video: hdmi: add helper function for N and CTS Fixes based on Russel King remarks - Duplicate function to have a separte treatment for coherent and non-coherent clocks - Add ratio field for alternate CTS value - Clock frequency in Hz for TMDS and audio clocks - Add information concerning clocks and CTS calculation. V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") https://patchwork.kernel.org/patch/8062611/ ("video: hdmi: add helper function for N and CTS") Arnaud Pouliquen (2): drm: sti: Add ASoC generic hdmi codec support. video: hdmi: add helper function for N and CTS drivers/gpu/drm/sti/Kconfig | 1 + drivers/gpu/drm/sti/sti_hdmi.c | 248 ++++++++++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 13 +++ drivers/video/hdmi.c | 202 +++++++++++++++++++++++++++++++++ include/linux/hdmi.h | 22 ++++ 5 files changed, 469 insertions(+), 17 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v5 0/4] Propagate errors out from compressed streams
by Charles Keepax 21 Apr '16

21 Apr '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v4: - Fix issue noticed by Vinod whereby the error state was being overwritten when snd_compr_stop was called It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 79 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 ++++-- sound/soc/soc-compress.c | 5 +-- 4 files changed, 91 insertions(+), 9 deletions(-) -- 2.1.4

4 12

[alsa-devel] [PATCH 0/2 v6] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 21 Apr '16

21 Apr '16

Because applied patch (commit 0beeb8d0b434b8b007ac8058fb2cac5997fff6df "ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") is dropped, I upload new version which include dependent patch. Changes since v5: - add "ASoC: core: export snd_soc_get_dai_name" patch. Because snd_soc_of_get_dai_name() fix the property name "sound-dai". Our patch needs to change name to use, so export snd_soc_get_dai_name() which is included in snd_soc_of_get_dai_name(). Changes since v4: Change name from "sound-dai" to "capture-dai" to determine the capture path Changes since v2,v3: Fix unclear code and use sound-dai to determine the capture path Changes since v1: Use codec dai name to determine the capture path. Jean-Francois Moine (1): ASoC: core: export snd_soc_get_dai_name PC Liao (1): ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ include/sound/soc.h | 2 + sound/soc/mediatek/mt8173-rt5650.c | 44 ++++++++++++++++++-- sound/soc/soc-core.c | 5 ++- 4 files changed, 52 insertions(+), 5 deletions(-) -- 1.7.9.5

3 9

[alsa-devel] [PATCH v8 0/2] ASoc: sun4i-codec: Add FM, Line and Mic inputs
by Danny Milosavljevic 21 Apr '16

21 Apr '16

Hi, this is the eighth version of the patch set that adds inputs to sun4i-codec. Changes compared to v7 are: - fix the routes for line and mic capturing. The patches are on top of the branch "sunxi-next" in <git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git>. Regards, Danny --- Danny Milosavljevic (2): sun4i-codec: Add FM, Line and Mic inputs sun4i-codec: make it possible to use different codec_widgets for A10 and A20. b/sound/soc/sunxi/sun4i-codec.c | 230 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 215 insertions(+), 15 deletions(-)

4 19

[alsa-devel] sound: use-after-free in snd_timer_interrupt
by Dmitry Vyukov 21 Apr '16

21 Apr '16

Hello, I am hitting the following use-after-free while running syzkaller fuzzer on commit 8e0f93cda48ed054e1216bab5c60017e1a5fc1e8 ================================================================== BUG: KASAN: use-after-free in __list_del_entry+0x1d3/0x1e0 at addr ffff88002ebf6e20 Read of size 8 by task syz-executor/7684 ============================================================================= BUG kmalloc-256 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- INFO: Allocated in snd_timer_instance_new+0x52/0x3a0 age=5 cpu=0 pid=7693 [< none >] ___slab_alloc+0x578/0x5d0 mm/slub.c:2464 [< none >] __slab_alloc+0x66/0xc0 mm/slub.c:2493 [< inline >] slab_alloc_node mm/slub.c:2556 [< inline >] slab_alloc mm/slub.c:2598 [< none >] kmem_cache_alloc_trace+0x242/0x3b0 mm/slub.c:2615 [< inline >] kmalloc include/linux/slab.h:463 [< inline >] kzalloc include/linux/slab.h:607 [< none >] snd_timer_instance_new+0x52/0x3a0 sound/core/timer.c:106 [< none >] snd_timer_open+0x522/0xd20 sound/core/timer.c:289 [< inline >] snd_timer_user_tselect sound/core/timer.c:1612 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1888 [< none >] snd_timer_user_ioctl+0x8f4/0x2490 sound/core/timer.c:1918 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Freed in snd_timer_close+0x3ee/0x750 age=9 cpu=0 pid=7693 [< none >] __slab_free+0x1fc/0x320 mm/slub.c:2674 [< inline >] slab_free mm/slub.c:2829 [< none >] kfree+0x2f5/0x370 mm/slub.c:3660 [< none >] snd_timer_close+0x3ee/0x750 sound/core/timer.c:375 [< inline >] snd_timer_user_tselect sound/core/timer.c:1602 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1888 [< none >] snd_timer_user_ioctl+0x7cd/0x2490 sound/core/timer.c:1918 [< inline >] vfs_ioctl fs/ioctl.c:43 [< none >] do_vfs_ioctl+0x18c/0xfb0 fs/ioctl.c:674 [< inline >] SYSC_ioctl fs/ioctl.c:689 [< none >] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:680 [< none >] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 INFO: Slab 0xffffea0000bafd00 objects=22 used=11 fp=0xffff88002ebf6d80 flags=0x1fffc0000004080 INFO: Object 0xffff88002ebf6d80 @offset=11648 fp=0xffff88002ebf5110 CPU: 3 PID: 7684 Comm: syz-executor Tainted: G B 4.5.0-rc7+ #337 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87b7d480 ffff88006d707bb0 ffffffff82c2125f ffffffff00bafd00 fffffbfff0f6fa90 ffff88003e807000 ffff88002ebf6d80 ffff88002ebf4000 ffffea0000bafd00 ffff88002ebf6e18 ffff88006d707be0 ffffffff8176dcc4 Call Trace: [<ffffffff81777e4e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:295 [<ffffffff82c88f63>] __list_del_entry+0x1d3/0x1e0 lib/list_debug.c:48 [< inline >] list_del_init include/linux/list.h:145 [<ffffffff852942f9>] snd_timer_interrupt+0x5b9/0xc80 sound/core/timer.c:791 [<ffffffff8529acc9>] snd_hrtimer_callback+0x169/0x230 sound/core/hrtimer.c:54 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1248 [<ffffffff814c3911>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1312 [<ffffffff814c62e2>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1346 [<ffffffff81255ec2>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff812593e9>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff866d256c>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:520 [< inline >] spin_unlock_irqrestore include/linux/spinlock.h:362 [<ffffffff8177369e>] __slab_free+0x1ae/0x320 mm/slub.c:2681 [< inline >] slab_free mm/slub.c:2829 [<ffffffff81773ed8>] kmem_cache_free+0x318/0x440 mm/slub.c:2838 [<ffffffff81704cb8>] ptlock_free+0x38/0x50 mm/memory.c:3912 [< inline >] pgtable_pmd_page_dtor include/linux/mm.h:1702 [<ffffffff812951ca>] ___pmd_free_tlb+0xaa/0x110 arch/x86/mm/pgtable.c:74 [< inline >] __pmd_free_tlb ./arch/x86/include/asm/pgalloc.h:106 [< inline >] free_pmd_range mm/memory.c:432 [< inline >] free_pud_range mm/memory.c:450 [<ffffffff816f4dc3>] free_pgd_range+0x973/0xbe0 mm/memory.c:526 [<ffffffff816f52f5>] free_pgtables+0x2c5/0x3b0 mm/memory.c:558 [<ffffffff817137c3>] exit_mmap+0x233/0x410 mm/mmap.c:2868 [<ffffffff813543f5>] mmput+0x95/0x230 kernel/fork.c:706 [< inline >] exit_mm kernel/exit.c:436 [<ffffffff81366eb2>] do_exit+0x7b2/0x2d00 kernel/exit.c:735 [<ffffffff81369578>] do_group_exit+0x108/0x330 kernel/exit.c:878 [<ffffffff8138cf14>] get_signal+0x634/0x15e0 kernel/signal.c:2307 [<ffffffff811a7db3>] do_signal+0x83/0x1c90 arch/x86/kernel/signal.c:712 [<ffffffff81006685>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:247 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [<ffffffff8100866a>] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 [<ffffffff866d18e2>] int_ret_from_sys_call+0x25/0x9f arch/x86/entry/entry_64.S:281 ================================================================== kasan: GPF could be caused by NULL-ptr deref or user memory accessgeneral protection fault: 0000 [#1] SMP DEBUG_PAGEALLOC KASAN Modules linked in: CPU: 3 PID: 7684 Comm: syz-executor Tainted: G B 4.5.0-rc7+ #337 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff880061062f80 ti: ffff88005ee10000 task.ti: ffff88005ee10000 RIP: 0010:[<ffffffff82c88e16>] [<ffffffff82c88e16>] __list_del_entry+0x86/0x1e0 RSP: 0018:ffff88006d707cd0 EFLAGS: 00010046 RAX: dffffc0000000000 RBX: 0000000000000000 RCX: ffff88002ebf6e18 RDX: 0000000000000000 RSI: 0000000000000001 RDI: 0000000000000000 RBP: ffff88006d707cf0 R08: ffffffff89cbcf00 R09: 0000000000000000 R10: ffffed000dae0f8c R11: 0000000000000000 R12: ffff88005ee0d120 R13: ffff8800670a2298 R14: ffff88005ee0d110 R15: ffff88002ebf6e18 FS: 0000000000000000(0000) GS:ffff88006d700000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b CR2: 00000000006e0000 CR3: 0000000007ae9000 CR4: 00000000000006e0 Stack: ffff88006d707cf0 ffff88002ebf6e18 dffffc0000000000 ffff88005ee0d120 ffff88006d707db0 ffffffff852942f9 ffff8800670a2368 0000000000000082 dffffc0000000000 ffff8800670a234c ffffffff87b7d480 ffffed000ce1446d Call Trace: <IRQ> [< inline >] list_del_init include/linux/list.h:145 [<ffffffff852942f9>] snd_timer_interrupt+0x5b9/0xc80 sound/core/timer.c:791 [<ffffffff8529acc9>] snd_hrtimer_callback+0x169/0x230 sound/core/hrtimer.c:54 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1248 [<ffffffff814c3911>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1312 [<ffffffff814c62e2>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1346 [<ffffffff81255ec2>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff812593e9>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff866d256c>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:520 <EOI> [< inline >] spin_unlock_irqrestore include/linux/spinlock.h:362 [<ffffffff8177369e>] __slab_free+0x1ae/0x320 mm/slub.c:2681 [< inline >] slab_free mm/slub.c:2829 [<ffffffff81773ed8>] kmem_cache_free+0x318/0x440 mm/slub.c:2838 [<ffffffff81704cb8>] ptlock_free+0x38/0x50 mm/memory.c:3912 [< inline >] pgtable_pmd_page_dtor include/linux/mm.h:1702 [<ffffffff812951ca>] ___pmd_free_tlb+0xaa/0x110 arch/x86/mm/pgtable.c:74 [< inline >] __pmd_free_tlb ./arch/x86/include/asm/pgalloc.h:106 [< inline >] free_pmd_range mm/memory.c:432 [< inline >] free_pud_range mm/memory.c:450 [<ffffffff816f4dc3>] free_pgd_range+0x973/0xbe0 mm/memory.c:526 [<ffffffff816f52f5>] free_pgtables+0x2c5/0x3b0 mm/memory.c:558 [<ffffffff817137c3>] exit_mmap+0x233/0x410 mm/mmap.c:2868 [<ffffffff813543f5>] mmput+0x95/0x230 kernel/fork.c:706 [< inline >] exit_mm kernel/exit.c:436 [<ffffffff81366eb2>] do_exit+0x7b2/0x2d00 kernel/exit.c:735 [<ffffffff81369578>] do_group_exit+0x108/0x330 kernel/exit.c:878 [<ffffffff8138cf14>] get_signal+0x634/0x15e0 kernel/signal.c:2307 [<ffffffff811a7db3>] do_signal+0x83/0x1c90 arch/x86/kernel/signal.c:712 [<ffffffff81006685>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:247 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:282 [<ffffffff8100866a>] syscall_return_slowpath+0x2ba/0x340 arch/x86/entry/common.c:344 [<ffffffff866d18e2>] int_ret_from_sys_call+0x25/0x9f arch/x86/entry/entry_64.S:281 Code: c0 0f 84 91 00 00 00 48 b8 00 02 00 00 00 00 ad de 48 39 c3 0f 84 9f 00 00 00 48 89 da 48 b8 00 00 00 00 00 fc ff df 48 c1 ea 03 <80> 3c 02 00 0f 85 f6 00 00 00 4c 8b 23 4c 39 e1 0f 85 95 00 00 RIP [<ffffffff82c88e16>] __list_del_entry+0x86/0x1e0 lib/list_debug.c:57 RSP <ffff88006d707cd0> ---[ end trace fd16e1eaa1720656 ]--- Kernel panic - not syncing: Fatal exception in interrupt Shutting down cpus with NMI Kernel Offset: disabled ---[ end Kernel panic - not syncing: Fatal exception in interrupt It is not easily reproducible. I've hit several times while running fuzzer for a week. Here is one of the logs for the record: https://gist.githubusercontent.com/dvyukov/c84798ee55721563ecb537c4d51dc9f5…

2 8

[alsa-devel] linux-4.6-rc4/sound/pci/ens1370.c:1551: possible bad expression ?
by David Binderman 21 Apr '16

21 Apr '16

Hello there, [linux-4.6-rc4/sound/pci/ens1370.c:1551]: (style) Expression '(X & 0xf0000)>= 0x4' is always false. Source code is if ((ensoniq->ctrl & ES_1371_GPIO_OUTM)>= 4) val = 1; Regards David Binderman

3 3

[alsa-devel] [PATCH 0/4] ASoC: topology: configure more on DAI & DAI links
by mengdong.lin＠linux.intel.com 21 Apr '16

21 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. Two patches are small changes in ASoC core, to allow topology to further configure DAI and DAI links. In the other two patches, topology will configure more on FE DAI links. Guneshwor Singh (1): ASoC: topology: Set CPU DAI name and enable DPCM by default for FE link Mengdong Lin (3): ASoC: Export snd_soc_find_dai() ASoC: Change DAI link's be_id to a generic id ASoC: topology: Set the link ID when creating a FE DAI link include/sound/soc.h | 5 ++++- include/uapi/sound/asoc.h | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 12 ++++++------ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 12 ++++++------ sound/soc/intel/boards/skl_rt286.c | 10 +++++----- sound/soc/soc-core.c | 3 ++- sound/soc/soc-topology.c | 11 +++++++++++ 14 files changed, 42 insertions(+), 27 deletions(-) -- 2.5.0

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Re: [alsa-devel] Sound: BUG: KASAN: use-after-free in kill_fasync
by Baozeng Ding 21 Apr '16

21 Apr '16

On 2016/4/6 19:37, Baozeng Ding wrote: > > > On 2016/4/5 22:18, Takashi Iwai wrote: >> On Tue, 05 Apr 2016 15:51:30 +0200, >> Baozeng Ding wrote: >>> Hi all, >>> I've got the following report (use-after-free in kill_fasync) while >>> running syzkaller. >>> Unfortunately no reproducer.The kernel version is 4.5 (on Mar 16 commit >>> 09fd671ccb2475436bd5f597f751ca4a7d177aea). >>> >>> ================================================================== >>> BUG: KASAN: use-after-free in kill_fasync+0x3fb/0x420 at addr >>> ffff880067691d88 >>> Read of size 8 by task swapper/2/0 >>> ============================================================================= >>> >>> BUG kmalloc-2048 (Not tainted): kasan: bad access detected >>> ----------------------------------------------------------------------------- >>> >>> >>> Disabling lock debugging due to kernel taint >>> INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446678412249576073 >>> cpu=2245704320 pid=-1 >>> [< inline >] kmalloc /kernel/include/linux/slab.h:472 >>> [< inline >] kzalloc /kernel/include/linux/slab.h:616 >>> [< none >] snd_pcm_attach_substream+0x3b4/0xb10 >>> /kernel/sound/core/pcm.c:966 >>> [< none >] ___slab_alloc+0x4c7/0x500 /kernel/mm/slub.c:2446 >>> [< none >] __slab_alloc+0x4c/0x90 /kernel/mm/slub.c:2475 >>> [< inline >] slab_alloc_node /kernel/mm/slub.c:2538 >>> [< inline >] slab_alloc /kernel/mm/slub.c:2580 >>> [< none >] kmem_cache_alloc_trace+0x262/0x300 >>> /kernel/mm/slub.c:2597 >>> [< inline >] kmalloc /kernel/include/linux/slab.h:472 >>> [< inline >] kzalloc /kernel/include/linux/slab.h:616 >>> [< none >] snd_pcm_attach_substream+0x3b4/0xb10 >>> /kernel/sound/core/pcm.c:966 >>> [< none >] snd_pcm_open_substream+0x84/0x450 >>> /kernel/sound/core/pcm_native.c:2262 >>> [< inline >] snd_pcm_oss_open_file >>> /kernel/sound/core/oss/pcm_oss.c:2346 >>> [< none >] snd_pcm_oss_open.part.17+0x5a4/0x1100 >>> /kernel/sound/core/oss/pcm_oss.c:2428 >>> [< none >] snd_pcm_oss_open+0x35/0x50 >>> /kernel/sound/core/oss/pcm_oss.c:2392 >>> [< none >] soundcore_open+0x30f/0x640 >>> /kernel/sound/sound_core.c:639 >>> [< none >] chrdev_open+0x22a/0x4c0 /kernel/fs/char_dev.c:388 >>> [< none >] do_dentry_open+0x6a2/0xcb0 /kernel/fs/open.c:736 >>> [< none >] vfs_open+0x17b/0x1f0 /kernel/fs/open.c:853 >>> [< inline >] do_last /kernel/fs/namei.c:3258 >>> [< none >] path_openat+0x4837/0x5830 /kernel/fs/namei.c:3394 >>> [< none >] do_filp_open+0x18e/0x250 /kernel/fs/namei.c:3429 >>> [< none >] do_sys_open+0x201/0x420 /kernel/fs/open.c:1022 >>> [< inline >] SYSC_open /kernel/fs/open.c:1040 >>> [< none >] SyS_open+0x2d/0x40 /kernel/fs/open.c:1035 >>> INFO: Freed in 0x10000b076 age=18446678416544543380 cpu=0 pid=0 >>> [< none >] snd_pcm_detach_substream+0x134/0x280 >>> /kernel/sound/core/pcm.c:1017 >>> [< none >] __slab_free+0x1e8/0x300 /kernel/mm/slub.c:2657 >>> [< inline >] slab_free /kernel/mm/slub.c:2810 >>> [< none >] kfree+0x24e/0x2d0 /kernel/mm/slub.c:3661 >>> [< none >] snd_pcm_detach_substream+0x134/0x280 >>> /kernel/sound/core/pcm.c:1017 >>> [< none >] snd_pcm_release_substream.part.38+0x219/0x2f0 >>> /kernel/sound/core/pcm_native.c:2250 >>> [< none >] snd_pcm_release_substream+0x59/0x70 >>> /kernel/sound/core/pcm_native.c:2251 >>> [< none >] snd_pcm_oss_release_file+0x45/0xb0 >>> /kernel/sound/core/oss/pcm_oss.c:2305 >>> [< none >] snd_pcm_oss_release+0xfa/0x250 >>> /kernel/sound/core/oss/pcm_oss.c:2485 >>> [< none >] __fput+0x236/0x780 /kernel/fs/file_table.c:208 >>> [< none >] ____fput+0x15/0x20 /kernel/fs/file_table.c:244 >>> [< none >] task_work_run+0x16b/0x200 >>> /kernel/kernel/task_work.c:115 >>> [< inline >] exit_task_work >>> /kernel/include/linux/task_work.h:21 >>> [< none >] do_exit+0x87f/0x2c90 /kernel/kernel/exit.c:748 >>> [< none >] do_group_exit+0x108/0x330 >>> /kernel/kernel/exit.c:878 >>> [< inline >] SYSC_exit_group /kernel/kernel/exit.c:889 >>> [< none >] SyS_exit_group+0x1d/0x20 /kernel/kernel/exit.c:887 >>> [< none >] entry_SYSCALL_64_fastpath+0x23/0xc1 >>> /kernel/arch/x86/entry/entry_64.S:207 >>> INFO: Slab 0xffffea00019da400 objects=13 used=8 fp=0xffff880067691be0 >>> flags=0x5fffc0000004080 >>> INFO: Object 0xffff880067691bd8 @offset=7128 fp=0xbbbbbbbbbbbbbbbb >>> Call Trace: >>> <IRQ> [< inline >] __dump_stack /kernel/lib/dump_stack.c:15 >>> <IRQ> [<ffffffff82945051>] dump_stack+0xb3/0x112 >>> /kernel/lib/dump_stack.c:51 >>> [<ffffffff817009ad>] print_trailer+0x10d/0x190 /kernel/mm/slub.c:668 >>> [<ffffffff817074af>] object_err+0x2f/0x40 /kernel/mm/slub.c:675 >>> [< inline >] print_address_description >>> /kernel/mm/kasan/report.c:138 >>> [<ffffffff81709ca5>] kasan_report_error+0x215/0x530 >>> /kernel/mm/kasan/report.c:236 >>> [< inline >] ? spin_lock >>> /kernel/include/linux/spinlock.h:302 >>> [<ffffffff84a1c550>] ? snd_pcm_stream_lock+0x80/0xd0 >>> /kernel/sound/core/pcm_native.c:104 >>> [< inline >] kasan_report /kernel/mm/kasan/report.c:259 >>> [<ffffffff8170a0be>] __asan_report_load8_noabort+0x3e/0x40 >>> /kernel/mm/kasan/report.c:280 >>> [<ffffffff81792a0b>] ? kill_fasync+0x3fb/0x420 >>> /kernel/fs/fcntl.c:729 >>> [<ffffffff81792a0b>] kill_fasync+0x3fb/0x420 /kernel/fs/fcntl.c:729 >>> [<ffffffff84a36e08>] snd_pcm_period_elapsed+0x1c8/0x230 >>> /kernel/sound/core/pcm_lib.c:1890 >>> [<ffffffff84b6e2e0>] ? get_bound_vga.isra.21.part.22+0x140/0x140 >>> /kernel/sound/pci/hda/hda_intel.c:1327 >>> [<ffffffff84b0637d>] stream_update+0xad/0xe0 >>> /kernel/sound/pci/hda/hda_controller.c:923 >>> [<ffffffff84c1583e>] snd_hdac_bus_handle_stream_irq+0x24e/0x350 >>> /kernel/sound/hda/hdac_controller.c:449 >>> [<ffffffff84b062d0>] ? azx_init_chip+0x100/0x100 >>> /kernel/sound/pci/hda/hda_controller.c:890 >>> [<ffffffff84b04d24>] ? azx_interrupt+0x14/0x4d0 >>> /kernel/sound/pci/hda/hda_controller.c:929 >>> [<ffffffff84b04eee>] azx_interrupt+0x1de/0x4d0 >>> /kernel/sound/pci/hda/hda_controller.c:954 >>> [<ffffffff84b04d10>] ? azx_stop_chip+0x20/0x20 >>> /kernel/sound/pci/hda/hda_controller.c:908 >>> [<ffffffff8143d9e3>] handle_irq_event_percpu+0xf3/0x790 >>> /kernel/kernel/irq/handle.c:145 >>> [<ffffffff8143e127>] handle_irq_event+0xa7/0x140 >>> /kernel/kernel/irq/handle.c:192 >>> [<ffffffff81447f81>] handle_edge_irq+0x1e1/0x8d0 >>> /kernel/kernel/irq/chip.c:623 >>> [< inline >] generic_handle_irq_desc >>> /kernel/include/linux/irqdesc.h:146 >>> [<ffffffff811ad739>] handle_irq+0x109/0x2a0 >>> /kernel/arch/x86/kernel/irq_64.c:78 >>> [< inline >] ? rcu_lock_release >>> /kernel/include/linux/rcupdate.h:491 >>> [< inline >] ? rcu_read_unlock >>> /kernel/include/linux/rcupdate.h:926 >>> [< inline >] ? __atomic_notifier_call_chain >>> /kernel/kernel/notifier.c:184 >>> [<ffffffff81369c4f>] ? atomic_notifier_call_chain+0xbf/0x140 >>> /kernel/kernel/notifier.c:193 >>> [<ffffffff81369b90>] ? __atomic_notifier_call_chain+0x150/0x150 >>> /kernel/include/linux/rcupdate.h:922 >>> [<ffffffff811ac36d>] do_IRQ+0x7d/0x1a0 >>> /kernel/arch/x86/kernel/irq.c:240 >>> [<ffffffff85dac40c>] common_interrupt+0x8c/0x8c >>> /kernel/arch/x86/entry/entry_64.S:454 >>> <EOI> [<ffffffff812245c6>] ? native_safe_halt+0x6/0x10 >>> /kernel/./arch/x86/include/asm/irqflags.h:49 >>> [<ffffffff81405ced>] ? trace_hardirqs_on+0xd/0x10 >>> /kernel/kernel/locking/lockdep.c:2635 >>> [< inline >] arch_safe_halt >>> /kernel/./arch/x86/include/asm/paravirt.h:117 >>> [<ffffffff811bf152>] default_idle+0x22/0x2d0 >>> /kernel/arch/x86/kernel/process.c:307 >>> [<ffffffff811c053a>] arch_cpu_idle+0xa/0x10 >>> /kernel/arch/x86/kernel/process.c:298 >>> [<ffffffff813eb618>] default_idle_call+0x48/0x70 >>> /kernel/kernel/sched/idle.c:93 >>> [< inline >] cpuidle_idle_call >>> /kernel/kernel/sched/idle.c:151 >>> [< inline >] cpu_idle_loop /kernel/kernel/sched/idle.c:242 >>> [<ffffffff813ebaa7>] cpu_startup_entry+0x467/0x600 >>> /kernel/kernel/sched/idle.c:291 >>> [<ffffffff81201432>] start_secondary+0x2b2/0x380 >>> /kernel/arch/x86/kernel/smpboot.c:259 >>> [<ffffffff81201180>] ? set_cpu_sibling_map+0x18a0/0x18a0 >>> /kernel/include/linux/topology.h:80 >>> Memory state around the buggy address: >>> ffff880067691c80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >>> ffff880067691d00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >>> >ffff880067691d80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >>> ^ >>> ffff880067691e00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >>> ffff880067691e80: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >>> ================================================================== >>> >>> It seems that in function snd_pcm_detach_substream, when freeing >>> runtime >>> structure, it did not handle its fasync >>> data pointer. Any ideas? Thanks. >> Well, fasync doesn't need a cleanup usually. Do you have a proper >> reproducer code? > Sorry, I cannot reproduce it. I will apply your patch and test it. I > will tell you the result whether it is okay a few days later. > Thanks. > >> In anyway, try the patch below. I'm not sure whether this would help, >> but if the path is via kill_fasync(), it'd be good to put in the >> protected context. >> During the last few weeks, I have not seen the bug any more after applying your patch. Thanks. >> >> Takashi >> >> --- >> diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c >> index 3a9b66c6e09c..0aca39762ed0 100644 >> --- a/sound/core/pcm_lib.c >> +++ b/sound/core/pcm_lib.c >> @@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct >> snd_pcm_substream *substream) >> snd_timer_interrupt(substream->timer, 1); >> #endif >> _end: >> - snd_pcm_stream_unlock_irqrestore(substream, flags); >> kill_fasync(&runtime->fasync, SIGIO, POLL_IN); >> + snd_pcm_stream_unlock_irqrestore(substream, flags); >> } >> EXPORT_SYMBOL(snd_pcm_period_elapsed); >

2 1

[alsa-devel] [RFC PATCH v2 0/5] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 21 Apr '16

21 Apr '16

Refactor and go v2 because of the 60KB limitation This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. The interface to connect with BT chip is not finished yet, but as an upstream rookie I really need some comments. Any comments will be greatly appreciated. Garlic Tseng (5): ASoC: mediatek: Refine mt8173 driver ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add clock and irq control for 2701 platform driver ASoC: mediatek: add mt2701 platform driver implementation. ASoC: mediatek: Add mt2701-cs42448 driver and config option. .../devicetree/bindings/sound/mt2701-cs42448.txt | 39 + .../devicetree/bindings/sound/mtk2701-afe-pcm.txt | 150 ++ sound/soc/mediatek/Kconfig | 21 + sound/soc/mediatek/Makefile | 15 +- sound/soc/mediatek/mt2701/Makefile | 20 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 376 +++++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c | 259 +++ sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h | 28 + sound/soc/mediatek/mt2701/mtk2701-afe-common.h | 217 +++ sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c | 1752 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mtk2701-irq.c | 109 ++ sound/soc/mediatek/mt2701/mtk2701-irq.h | 30 + sound/soc/mediatek/mt2701/mtk2701-reg.h | 165 ++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/{ => mt8173}/mtk-afe-common.h | 0 sound/soc/mediatek/{ => mt8173}/mtk-afe-pcm.c | 0 20 files changed, 3186 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 Documentation/devicetree/bindings/sound/mtk2701-afe-pcm.txt create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.c create mode 100644 sound/soc/mediatek/mt2701/mtk2701-irq.h create mode 100644 sound/soc/mediatek/mt2701/mtk2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) rename sound/soc/mediatek/{ => mt8173}/mtk-afe-common.h (100%) rename sound/soc/mediatek/{ => mt8173}/mtk-afe-pcm.c (100%) -- 1.9.1

4 11

[alsa-devel] [PATCH v7] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 20 Apr '16

20 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v6: Use subnode to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 ++++ sound/soc/mediatek/mt8173-rt5650.c | 49 ++++++++++++++++++-- 2 files changed, 56 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..5bfa6b6 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,21 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + codec-capture { + sound-dai = <&rt5650 1>; + }; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..b8ed965 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,28 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_err(card->dev, "Can't get the right codec dai\n"); + return -EINVAL; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +126,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +170,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +198,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +222,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ASoC: ak4642: Remove CLK_IS_ROOT
by Stephen Boyd 20 Apr '16

20 Apr '16

This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> --- sound/soc/codecs/ak4642.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index cda27c22812a..1ee8506c06c7 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -608,9 +608,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev) of_property_read_string(np, "clock-output-names", &clk_name); - clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, - (parent_clk_name) ? 0 : CLK_IS_ROOT, - rate); + clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate); if (!IS_ERR(clk)) of_clk_add_provider(np, of_clk_src_simple_get, clk); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Remove CLK_IS_ROOT
by Stephen Boyd 20 Apr '16

20 Apr '16

This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Stephen Boyd <sboyd(a)codeaurora.org> --- sound/soc/sh/rcar/adg.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 606399de684d..49354d17ea55 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, req_rate); + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, else { for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->onecell.clks = adg->clkout; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

3 2

[alsa-devel] [PATCH 0/4] ASoC: add graph base connection on simple-card
by Kuninori Morimoto 20 Apr '16

20 Apr '16

Hi Mark Current simple-card is using "sound-dai" base connection on DT, but V4L2 is using graph base connection. For example HDMI case, we would like to use both connection. To above confusable connection method, and to reuse current resource, this patch adds new function, and detect both "sound-dai" and "remote-endpoint" on simple-card. like this sound { compatible = "simple-audio-card"; ... sndcpu: simple-audio-card,cpu { sound-dai = <&xxx>; }; sndcodec: simple-audio-card,codec { => remote-endpoint = <&out_hdmi>; }; }; xxx { ... ports { ... port@0 { reg = <0>; out_rgb: endpoint { }; }; port@1 { reg = <1>; => out_hdmi: endpoint { }; }; ... }; }; Kuninori Morimoto (4): of_graph: add of_graph_get_endpoint_count() ASoC: add of_parse_snd_connection_with_args() for sound-dai/graph connection ASoC: snd_soc_of_get_dai_name() uses of_parse_snd_soc_connection_with_args() ASoC: simple-card: probe both sound-dai and remote-endpoint Documentation/devicetree/bindings/sound/simple-card.txt | 19 +++++++++++++++++++ include/linux/of_graph.h | 11 +++++++++++ include/sound/soc.h | 2 ++ sound/soc/generic/simple-card.c | 9 ++++++--- sound/soc/soc-core.c | 45 +++++++++++++++++++++++++++++++++++++++++++-- 5 files changed, 81 insertions(+), 5 deletions(-)

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Re: [alsa-devel] Issues with ALSA:ice1724 sound cards on powerpc machines
by Julian Margetson 20 Apr '16

20 Apr '16

On 2/17/2016 2:29 PM, Julian Margetson wrote: > Hi > > Having an issue issue with ice1724 sound cards on powerpc motherboards. > Motherboards in question Acube Sam460ex . > The card appears as a valid device but there is no sound output on the > majority of boots of the computer. > Logs attached. > On the first set boot audio playback produced sound ok. > On the second set boot no sound was produce with audio playback > > > Regards > Julian Margetson > > > Anyone have any ideas ? Resubmitting again with links to logs instead of included as attachments. The issue also seen on A-Eon Cyrus P5020 motherboard . https://www.dropbox.com/s/72labcje9z1r1b7/1alsa-info.txt.7CLHMG0Gap?dl=0 https://www.dropbox.com/s/v1pn5kjqrvwu16f/boot_1.log?dl=0 https://www.dropbox.com/s/uuaa14rkpmmz1o0/2alsa-info.txt.VKiEVuytiJ?dl=0 https://www.dropbox.com/s/wwu2f5t93l8g26d/boot_2.log?dl=0 Thanks

3 7

[alsa-devel] GPIO control from ALSA
by Petr Kulhavy 20 Apr '16

20 Apr '16

Hello, I'm writing a driver for ASoC "sound card" hardware which uses CPU GPIO to mute/unmute an analogue amplifier and another GPIO to control an analogue switch for line/mic switching. Is there an abstraction in ALSA to model this kind of hardware? Thanks Petr

2 5

[alsa-devel] [PATCH 1/1] ALSA: hda - add PCI ID for Intel Broxton-T
by han.lu＠intel.com 20 Apr '16

20 Apr '16

From: "Lu, Han" <han.lu(a)intel.com> Add HD Audio Device PCI ID for the Intel Broxton-T platform. It is an HDA Intel PCH controller. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b680be0..637b8a0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2232,6 +2232,9 @@ static const struct pci_device_id azx_ids[] = { /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, + /* Broxton-T */ + { PCI_DEVICE(0x8086, 0x1a98), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, -- 2.5.0

2 1

[alsa-devel] [asoc:topic/mtk 1/1] sound/soc/mediatek/mt8173-rt5650.c:236:3: error: implicit declaration of function 'snd_soc_get_dai_name'
by kbuild test robot 20 Apr '16

20 Apr '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/mtk head: 0beeb8d0b434b8b007ac8058fb2cac5997fff6df commit: 0beeb8d0b434b8b007ac8058fb2cac5997fff6df [1/1] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver config: arm-allmodconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 0beeb8d0b434b8b007ac8058fb2cac5997fff6df # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc/mediatek/mt8173-rt5650.c: In function 'mt8173_rt5650_dev_probe': >> sound/soc/mediatek/mt8173-rt5650.c:236:3: error: implicit declaration of function 'snd_soc_get_dai_name' [-Werror=implicit-function-declaration] ret = snd_soc_get_dai_name(&args, &codec_dai_capture); ^ cc1: some warnings being treated as errors vim +/snd_soc_get_dai_name +236 sound/soc/mediatek/mt8173-rt5650.c 230 dev_err(&pdev->dev, 231 "%s capture-dai name fail %d\n", 232 __func__, ret); 233 return ret; 234 } 235 > 236 ret = snd_soc_get_dai_name(&args, &codec_dai_capture); 237 mt8173_rt5650_codecs[1].dai_name = codec_dai_capture; 238 } 239 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: hda - Keep powering up ADCs on Cirrus codecs
by Takashi Iwai 19 Apr '16

19 Apr '16

Although one weird behavior about the input path (inconsistent D0/D3 switch) on Cirrus CS420x codecs was fixed in the previous commit, there is still an issue on some Mac machines: the capture stream stalls when switching the ADCs on the fly. More badly, this keeps stuck until the next reboot. The dynamic ADC switching is already a bit fragile and assuming optimistically that the chip accepts the frequent power changes. On Cirrus codecs, this doesn't seem applicable. As a quick workaround, we pin down the ADCs to keep up in D0 when spec->dyn_adc_switch is set. In this way, the ADCs are kept up only for the system that were confirmed to be broken. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116171 Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a47e8ae0eb30..80bbadc83721 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -361,6 +361,7 @@ static int cs_parse_auto_config(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; int err; + int i; err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); if (err < 0) @@ -370,6 +371,19 @@ static int cs_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + /* keep the ADCs powered up when it's dynamically switchable */ + if (spec->gen.dyn_adc_switch) { + unsigned int done = 0; + for (i = 0; i < spec->gen.input_mux.num_items; i++) { + int idx = spec->gen.dyn_adc_idx[i]; + if (done & (1 << idx)) + continue; + snd_hda_gen_fix_pin_power(codec, + spec->gen.adc_nids[idx]); + done |= 1 << idx; + } + } + return 0; } -- 2.8.1

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Add 96k sample rate
by Jeremy McDermond 19 Apr '16

19 Apr '16

The TLV320AIC32x4 series supports 96ksps rates in hardware. This patch adds the necessary PLL divider values and clock settings to the table to make 96ksps work. Signed-off-by: Jeremy McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6b6d55..2f8480c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -159,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 48k rate */ {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, - {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}, + + /* 96k rate */ + {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1}, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -564,7 +567,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.5.0

2 1

[alsa-devel] [PATCH v2] ASoC: sti: update dt-bindings for tdm mode
by Moise Gergaud 19 Apr '16

19 Apr '16

- Add "TDM" in the st,mode property list - st,mode property is also mandatory for reader - add tdm playback dai-link example Signed-off-by: Moise Gergaud <moise.gergaud(a)st.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- I forgot to include this patch into "[PATCH v2] ASoC: sti: unip tdm mode" serie that has been applied. history: - V2: rename patch with a subject line reflecting the style for the subsystem .../devicetree/bindings/sound/st,sti-asoc-card.txt | 44 +++++++++++++++++++--- 1 file changed, 39 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index f546dd9..4d9a83d 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -37,17 +37,18 @@ Required properties: - dai-name: DAI name that describes the IP. + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec and support IEC HDMI formats (player only). + "SPDIF" connected to SPDIF codec and support SPDIF formats (player only). + "PCM" PCM standard mode for I2S or TDM bus. + "TDM" TDM mode for TDM bus. + Required properties ("st,sti-uni-player" compatibility only): - clocks: CPU_DAI IP clock source, listed in the same order than the CPU_DAI properties. - uniperiph-id: internal SOC IP instance ID. - - IP mode: IP working mode depending on associated codec. - "HDMI" connected to HDMI codec IP and IEC HDMI formats. - "SPDIF"connected to SPDIF codec and support SPDIF formats. - "PCM" PCM standard mode for I2S or TDM bus. - Optional properties: - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for external codecs connection. @@ -56,6 +57,22 @@ Optional properties: Example: + sti_uni_player1: sti-uni-player@1 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + st,dai-name = "Uni Player #1 (I2S)"; + dma-names = "tx"; + st,uniperiph-id = <1>; + st,version = <5>; + st,mode = "TDM"; + }; + sti_uni_player2: sti-uni-player@2 { compatible = "st,sti-uni-player"; status = "okay"; @@ -99,6 +116,7 @@ Example: dma-names = "rx"; dai-name = "Uni Reader #1 (HDMI RX)"; version = <3>; + st,mode = "PCM"; }; 2) sti-sas-codec: internal audio codec IPs driver @@ -152,4 +170,20 @@ Example of audio card declaration: sound-dai = <&sti_sasg_codec 0>; }; }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; }; -- 1.9.1

2 1

[alsa-devel] [PATCH v4 0/4] Propagate errors out from compressed streams
by Charles Keepax 19 Apr '16

19 Apr '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v3: - Add missing comment for new field in snd_compr_stream - Return -EPIPE on XRUN to match what the PCM framework does It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 76 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 ++++-- sound/soc/soc-compress.c | 5 +-- 4 files changed, 88 insertions(+), 9 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: core: export snd_soc_get_dai_name
by PC Liao 19 Apr '16

19 Apr '16

From: Jean-Francois Moine <moinejf(a)free.fr> snd_soc_get_dai_name() may be used to define a sound card with a different syntax from the one of the simple-card. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- [1] is applied and needs this patch. Otherwise, [1] will build fail. [1] https://patchwork.kernel.org/patch/8878491/ ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 5 +++-- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a21..03b2d19 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1669,6 +1669,8 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, struct device_node **framemaster); +int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name); int snd_soc_of_get_dai_link_codecs(struct device *dev, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..69cbbb0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3786,8 +3786,8 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); -static int snd_soc_get_dai_name(struct of_phandle_args *args, - const char **dai_name) +int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name) { struct snd_soc_component *pos; struct device_node *component_of_node; @@ -3838,6 +3838,7 @@ static int snd_soc_get_dai_name(struct of_phandle_args *args, mutex_unlock(&client_mutex); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_get_dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name) -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/2] ASoC: da7219: Small clocking related updates for PLL and sysclk
by Adam Thomson 19 Apr '16

19 Apr '16

The patch set contains 2 small fixes/updates relating to clocking in the codec. The first improves PLL locking for a corner scenario with a low frequency MCLK provided, and the second is to remove 32KHz support from the codec set_dai_sysclk() function, to align with updates already made to the set_dai_pll() function. Patches are based on v4.6-rc4 kernel version. Adam Thomson (2): ASoC: da7219: Update PLL ranges and dividers to improve locking ASoC: da7219: Disallow unsupported 32KHz clock setting in set_dai_sysclk() sound/soc/codecs/da7219.c | 30 +++++++++++++++--------------- sound/soc/codecs/da7219.h | 20 ++++++++++---------- 2 files changed, 25 insertions(+), 25 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH v9 0/7] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 19 Apr '16

19 Apr '16

Couple of minor changes since v8 to ALSA/ASoC side patches. The series is based on top of drm-next. However, it does not look like there is enough attention to the DRM side so that the tda998x patches would get in any time soon. So I am happy to rebase the first ALSA-side patches on top of Mark Brown's for-next tree, for instance, if they would have a change to go in on their own. The first two IEC958 helper patches should apply as such. There is currently two other patch series[2][3] that depend on the first three (ALSA-) patches of this series. The third ("ASoC: hdmi-codec: Add audio abort() callback for video side to use") is currently not used and can be dropped if so decided. The rest depends on those the first patches and adds hdmi-audio support for Beaglebone-black. Best regards, Jyri [1] "[PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105524.html [2] "[PATCH v6 00/10] ASoC: Add mediatek HDMI codec support" by Philipp Zabel http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [3] "[RFC v2 0/6] sti: add audio interface to the hdmi driver" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/103374.ht… and "[PATCH v4 0/6] add IEC958 channel status control helpers" http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html by Arnaud Pouliquen Changes since v8 * "ALSA: pcm: add IEC958 channel status helper for hw_params" - Split out the 32 bit support - add: "ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper" * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Set hw_params->msbits to 24 in hw_params() callback if sample width is greater than 24 bits Changes since v7 * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Remove the unused "int ret" variable from tda998x_configure_audio() Changes since v6 * "ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders" - Add "Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com>" and "Tested-by: Philipp Zabel <p.zabel(a)pengutronix.de>", no other changes * "ALSA: pcm: add IEC958 channel status helper for hw_param" - Added kernel-doc for snd_pcm_create_iec958_consumer_hw_params() * "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Use PTR_ERR_OR_ZERO() in tda998x_audio_codec_init() - Fix possible uninitialized use of 'size' in tda998x_get_audio_ports() - Store audio configuration in tda998x_audio_hw_params() instead of tda998x_configure_audio(). Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (7): ALSA: pcm: add IEC958 channel status helper for hw_params ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 275 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 65 +++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 414 +++++++++++++++++++++ 12 files changed, 917 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

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[alsa-devel] [PATCH v5] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 19 Apr '16

19 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. This patch depends on [1], because snd_soc_of_get_dai_name() fix the property name "sound-dai". This patch needs to change name to use, so export snd_soc_get_dai_name() which is included in snd_soc_of_get_dai_name(). [1] https://patchwork.kernel.org/patch/5671961/ ("ASoC: core: export snd_soc_get_dai_name") Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v4: Change name from "sound-dai" to "capture-dai" to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 44 ++++++++++++++++++-- 2 files changed, 47 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..802dd1b 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,17 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional properties: +- capture-dai: audio codec dai on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + capture-dai = <&rt5650 1>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..45dea22 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,26 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_dai_capture = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_dai_capture, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_dai_capture, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_err(card->dev, "Can't get the right codec dai\n"); + return -EINVAL; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +124,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +168,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +196,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + const char *codec_dai_capture; + struct of_phandle_args args; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +220,23 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (device_property_present(&pdev->dev, "capture-dai")) { + ret = of_parse_phandle_with_args(pdev->dev.of_node, + "capture-dai", + "#sound-dai-cells", 0, &args); + if (ret < 0) { + dev_err(&pdev->dev, + "%s capture-dai name fail %d\n", + __func__, ret); + return ret; + } + + ret = snd_soc_get_dai_name(&args, &codec_dai_capture); + mt8173_rt5650_codecs[1].dai_name = codec_dai_capture; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 4

[alsa-devel] [PATCH v3] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 19 Apr '16

19 Apr '16

Provide *our* view of what the rules are for the different DAI formats, so that we do not have to trust external interpretations for this crucial bit of interoperability. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- Documentation/sound/alsa/soc/clocking.txt | 146 +++++++++++++++++++++++++++++- 1 file changed, 144 insertions(+), 2 deletions(-) Changed PCM mode A to start left word one BCLK after positive LRC edge. Added link to I2S spec as indicated by Mark Brown. Cheers, Peter diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index b1300162e01c..7eb02f0758e7 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -47,5 +47,147 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. - - +The below diagrams all have BCLK as the first signal, LRC as the second signal +and DATA as the third. Below that is an indication about which DATA bits belong +in what channel. + +A "..." marking as DATA indicates that there may be more bits that are not +shown. Also, all DATA bits marked X may or may not be present. DAI +transmitters must add them should LRC not match the word size exactly and DAI +receivers must be prepared to ignore them. DAI transmitters must insert zeros +for I2S and Left Justified modes, and preferably not drive DATA for the DSP +modes for these extra X bits. + + +I2S + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word starts one BCLK cycle after a negative flank of LRC, and +the right channel word starts one BCLK cycle after a positive flank of LRC. The +words start with the MSB. Receivers must truncate words if more bits per word +are transmitted than they can use, and transmitters must pad words with zeros +if LRC is not matching the word size. Also, see +https://www.sparkfun.com/datasheets/BreakoutBoards/I2SBUS.pdf + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified (aka MSB) + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word starts directly after a positive flank of LRC, and the +right channel word starts directly after a negative flank of LRC. The words +start with the MSB. Receivers must truncate words if more bits per word are +transmitted than they can use, and transmitters must pad words with zeros if +LRC is not matching the word size. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Right Justified (aka LSB) + +LRC should have its flanks synchronized with a negative flank of BCLK. +The left channel word end right before a negative flank of LRC, and the right +channel word ends right before a positive flank of LRC. The words end with LSB. +Transmitters must pad words if LRC is not matching the word size and the +transmitter and the receiver have to agree on the word leghth. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | X |...| X |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +DSP mode A + +LRC should have positive flanks synchronized with a negative flank of BCLK. +The left channel word starts one BCLK after a positive flank of LRC, and the +right channel word starts directly after the left channel word. The words +start with MSB. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | | + +LRC examples for DSP mode A + .---. .---. +---' '-------------------------------------------------------' '----- + .-. .-. +---' '---------------------------------------------------------' '------- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +DSP mode B + +LRC should have positive flanks synchronized with a negative flank of BCLK. +The left channel word starts directly after a positive flank of LRC, and the +right channel word starts directly after the left channel word. The words +start with MSB. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | | + +LRC examples for DSP mode B + .---. .---. +---' '-------------------------------------------------------' '----- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +The above diagrams show normal BCLK and LRC clocking where DAI transmitters +change DATA on the falling edge of BCLK and DAI receivers read DATA on the +rising edge. For inverted BCLK it is naturally the other way around. Inverted +LRC is just that. Note that DSP mode A is compatible with DSP mode B with +inverted LRC, and vice versa. + + +Left Justified, inverted BCLK +-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .- + '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified, inverted LRC + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | -- 2.1.4

2 1

[alsa-devel] [PATCH] Documentation: sti-asoc-card: update tdm mode
by Moise Gergaud 19 Apr '16

19 Apr '16

- Add "TDM" in the st,mode property list - st,mode property is also mandatory for reader - add tdm playback dai-link example Signed-off-by: Moise Gergaud <moise.gergaud(a)st.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- I forgot to include this patch into "[PATCH v2] ASoC: sti: unip tdm mode" serie that has been applied. This patch is the same than the one previously sent in http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106392.html .../devicetree/bindings/sound/st,sti-asoc-card.txt | 44 +++++++++++++++++++--- 1 file changed, 39 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index f546dd9..4d9a83d 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -37,17 +37,18 @@ Required properties: - dai-name: DAI name that describes the IP. + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec and support IEC HDMI formats (player only). + "SPDIF" connected to SPDIF codec and support SPDIF formats (player only). + "PCM" PCM standard mode for I2S or TDM bus. + "TDM" TDM mode for TDM bus. + Required properties ("st,sti-uni-player" compatibility only): - clocks: CPU_DAI IP clock source, listed in the same order than the CPU_DAI properties. - uniperiph-id: internal SOC IP instance ID. - - IP mode: IP working mode depending on associated codec. - "HDMI" connected to HDMI codec IP and IEC HDMI formats. - "SPDIF"connected to SPDIF codec and support SPDIF formats. - "PCM" PCM standard mode for I2S or TDM bus. - Optional properties: - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for external codecs connection. @@ -56,6 +57,22 @@ Optional properties: Example: + sti_uni_player1: sti-uni-player@1 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = <GIC_SPI 85 IRQ_TYPE_NONE>; + dmas = <&fdma0 3 0 1>; + st,dai-name = "Uni Player #1 (I2S)"; + dma-names = "tx"; + st,uniperiph-id = <1>; + st,version = <5>; + st,mode = "TDM"; + }; + sti_uni_player2: sti-uni-player@2 { compatible = "st,sti-uni-player"; status = "okay"; @@ -99,6 +116,7 @@ Example: dma-names = "rx"; dai-name = "Uni Reader #1 (HDMI RX)"; version = <3>; + st,mode = "PCM"; }; 2) sti-sas-codec: internal audio codec IPs driver @@ -152,4 +170,20 @@ Example of audio card declaration: sound-dai = <&sti_sasg_codec 0>; }; }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; }; -- 1.9.1

2 1

[alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Tan, Seng Kai 19 Apr '16

19 Apr '16

Hi Takashi, I am facing similar problem like this issue, hope to get your advice. http://mailman.alsa-project.org/pipermail/alsa-devel/2012-August/054241.html I try to turn on S24_LE support for HDMI Audio using below patch. When I aplay S24_LE file it works fine. The problem happen when I aplay S24_3LE file, I get and extremely low volume. When I aplay S24_LE file, I get an left justified value in I2S, this is an expected result: Front end S24_LE : 0x12345600(in 4bytes, LSB 0 padding) Back end S24_3LE : 0x12345600(in 4bytes, LSB 0 padding) When I aplay S24_3LE file, I get an extremely low output volume. Front end S24_3LE : 0x123456(in 3bytes, No padding) Back end S24_LE : 0x00123456(in 4bytes, MSB 0 padding) Can you advice how to tell the alsa plugin to pad in LSB instead of MSB? Below are the patch I used to turn on S24_LE for HDA . Regards, Seng Kai >From 33f182fe1f6d1e49d0618559d1212b619ec902ce Mon Sep 17 00:00:00 2001 From: "SengKai,Tan" <seng.kai.tan(a)intel.com> Date: Fri, 8 Apr 2016 17:34:51 +0800 Subject: [PATCH] ALSA: hda: S24_LE format support for HDA Audio This patch is to add S24_LE support for HDMI audio and HDA codec. Before this patch, HDMI and HDA codec audio only supported S16_LE and S32_LE. Let's say if user would like to play S24_LE file, it has to use plugin to convert to either S16_LE or S32_LE which will increase in processing and less efficient in utilize the resources. On the other side, let's say connected HDMI monitor support S16_LE and S24_LE. Before this patch is applied audio has to be converted to S16_LE. That will reduce the audio quality. Signed-off-by: Tan, Seng Kai <seng.kai.tan(a)intel.com> --- sound/pci/hda/hda_codec.c | 6 ++++-- sound/pci/hda/hda_eld.c | 6 ++++-- sound/pci/hda/patch_hdmi.c | 3 ++- 3 files changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 57197be..9e737bb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3501,14 +3501,16 @@ int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, if (val & AC_SUPPCM_BITS_32) formats |= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE; if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24)) - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (val & AC_SUPPCM_BITS_24) bps = 24; else if (val & AC_SUPPCM_BITS_20) bps = 20; } else if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24| AC_SUPPCM_BITS_32)) { - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (val & AC_SUPPCM_BITS_32) bps = 32; else if (val & AC_SUPPCM_BITS_24) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 0e6d753..65bfe10 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -613,12 +613,14 @@ void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { if (a->sample_bits & AC_SUPPCM_BITS_20) { - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (maxbps < 20) maxbps = 20; } if (a->sample_bits & AC_SUPPCM_BITS_24) { - formats |= SNDRV_PCM_FMTBIT_S32_LE; + formats |= SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE; if (maxbps < 24) maxbps = 24; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d02eccd..8cbde13 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2489,7 +2489,8 @@ static const struct hda_verb nvhdmi_basic_init_7x_8ch[] = { SNDRV_PCM_RATE_192000) #define SUPPORTED_MAXBPS 24 #define SUPPORTED_FORMATS \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) #endif static int nvhdmi_7x_init_2ch(struct hda_codec *codec) -- 1.9.1

2 5

[alsa-devel] Can we re-use dw-hdmi-ahb-audio ?
by Kuninori Morimoto 19 Apr '16

19 Apr '16

Hi I know we already have dw-hdmi-ahb-audio.c which is based on normal ALSA style, but I would like to use it as ASoC. I'm thinking that we can separate this existing dw-hdmi-ahb-audio driver into "common parts" and ahb "specific parts", and share common parts with ASoC, like below, but what do you think ? dw-hdmi-audio.c : dw-hdmi audio common function set dw-hdmi-ahb-audio.c : dw-hdmi audio for ahb, using dw-hdmi-audio.c dw-hdmi-soc-audio.c : dw-hdmi audio for ASoC, using dw-hdmi-audio.c

2 2

Re: [alsa-devel] MC13783 ALSA driver with asynchronous DAI support
by Edison 19 Apr '16

19 Apr '16

HI Sascha, I google a post from you. http://comments.gmane.org/gmane.linux.alsa.devel/68979 You said you used asynchronous mode on MC13783 (with two SSI ports, one for capture, one for play back). I think it is exactly the case I met, could you please show me the information to make it works? Thanks very much. At 2016-04-18 11:25:28, "Edison" <13880996366(a)163.com> wrote: >HI Community, > > I'm working on kernel upgrading for 2.6.22 BSP to new community version 4.X (download from kernel.org without any vendor patches) for our customized IMX27 board, but met some problem when porting the MC13783 audio device driver. On our board, the hardware design for audio part is the same as IMX27 ADS board, in which MC13783 audio is connect to IMX27 with two SSI interface (SSI1/SS2). In the FSL 2.6.22 BSP, driver design is the same as the hardware schematic - alsa driver use SSI1 / SSI2 as codec and DAC respectively. In the community BSP (4.X), I tried to assign two difference SSI port for MC13783 codec as below > /* MC13783 */ > static struct mc13xxx_codec_platform_data mx27_ads_codec = { > .dac_ssi_port = MC13783_SSI2_PORT, > .adc_ssi_port = MC13783_SSI1_PORT, > }; > > In mc13783 codec driver it will be bind to asynchronous DAI for separated routing for AD/DA (mc13783.c), one called mc13783-hifi-playback and the other called mc13783-hifi-capture, but the DAI bind will be certainly unsuccessful because in the mc13783 sound soc driver (imx-mc13783.c), the dai_link name seems fixed to be "mc13783_hifi", so looks like DAI bind can only be successfully if we use synchronous mode (AD/DA use the same SSI port rather than different ports). That seems to be a big difference between community BSP and FSL BSP. My question: >1. Has someone used the asynchronous DAI in their design with community BSP (I cannot find any reference for mc13783-hifi-playback/mc13783-hifi-capture in the kernel source code)? >2. Can I use synchronous DAI (force ADC/DAC use same SSI) instead of asynchronous DAI make audio works without hardware change? (Unfortunately I have done some experiments but seems not work, I cannot get the proper voice on the headphone when I say sometime on Mic with my self loop program). > BTW, I met another problem even using synchronous DAI, the driver always return "coherent DMA mask is unset", now I have to patch it with default DMA mask. >

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: tlv320aic32x4: Add SPI Support
by Jeremy McDermond 19 Apr '16

19 Apr '16

The TLV320AIC32x4 support the control channel over either SPI or I2C. Added the proper interfacing with regmap to make the driver able to handle either possibility. Jeremy McDermond (3): ASoC: tlv320aic32x4: Change name of probe function ASoC: tlv320aic32x4: Break out I2C support into separate module ASoC: tlv320aic32x4: Add SPI support. sound/soc/codecs/Kconfig | 13 +++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/tlv320aic32x4-i2c.c | 74 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4-spi.c | 76 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 80 ++++++++++++------------------------ sound/soc/codecs/tlv320aic32x4.h | 7 ++++ 6 files changed, 199 insertions(+), 55 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic32x4-i2c.c create mode 100644 sound/soc/codecs/tlv320aic32x4-spi.c -- 2.5.0

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Re: [alsa-devel] [PATCH] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 18 Apr '16

18 Apr '16

On 2016-04-18 17:11, Mark Brown wrote: > On Mon, Apr 18, 2016 at 01:18:47PM +0000, Peter Rosin wrote: >> Mark Brown wrote: > >>> >>> There aren't any (beyond the usual references to the Wolfson datasheets >>> which I'd suggest should be in here) but that doesn't mean we should >>> ignore this spec when we have it. > >> This is exactly the problem. For an outsider, it's impossible to know that >> wolfson has the correct definition of the modes. Why should wolfson datasheets >> trumph nxp or ti datasheets (or whatever), if there is an inconsistency? > > I'm not quite sure what your concern is here? I'm saying that where > there are specs we should link to them. I'm not saying we can't add > to that. I'm just saying that I don't know which specs to trust. E.g. a random wolfson datasheet (wm8778) says that DSP mode B will have BCLK and LRC like this when it is clk master .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. -' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- .---. .---. -------' '-------------------------------------------------------' '----- and that it will accept the negative LRC flank later when it is clk slave, with a figure showing this as the extreme: .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. -' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- ---. .-------------------------------------------------------. .----- '---' '---' Then we have the tfa9879 datasheet from nxp which has this for its long sync format: -. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .- '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' -----. .---------------------------------------------------------. .----- '-' '-' This is all very difficult to match up for someone who doesn't work with this on a day-to-day basis and know it by heart. Why should I go to some random wolfson datasheet for info? I don't know if the nxp long sync format is compatible with DSP mode B with inverted BCLK, but it certainly looks like it might be. The LRC is low for only half a cycle and that is not acceptable according to the wolfson datasheet, but is that a limitation in wm8778 or is the tfa9879 problematic? How would you suppose I figure out if the tfa9879 driver should declare compatibility with DSP mode B with inverted BCLK if there is no documentation of what *ASoC* thinks that DSP mode B really is? This info is desperately missing, that is all that I'm saying. With this background, I'm a bit reluctant to add links to some datasheet, because they tend to describe how the chip in question behaves, and not the protocol as such. But sure, the I2S spec is something else as it's independent from any particular chip. BTW, I found out that I had misunderstood DSP mode A in v1/v2 of the patch, so a v3 is coming up where I'll also add a link to the I2S spec. If there is any particular link that you think I should add for any other format spec, please holler. Cheers, Peter

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[alsa-devel] [PATCH v4] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 18 Apr '16

18 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v2,v3: Fix unclear code and use sound-dai to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 6 +++ sound/soc/mediatek/mt8173-rt5650.c | 40 ++++++++++++++++++-- 2 files changed, 43 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..4ed768e 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,17 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional properties: +- sound-dai: audio codec dai on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + sound-dai = <&rt5650 1>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..f2af2dd 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,26 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_dai_capture = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_dai_capture, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_dai_capture, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_err(card->dev, "Can't get the right codec dai\n"); + return -EINVAL; + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +124,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +168,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +196,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + const char *codec_dai_capture; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +219,20 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (device_property_present(&pdev->dev, "sound-dai")) { + ret = snd_soc_of_get_dai_name(pdev->dev.of_node, + &codec_dai_capture); + if (ret < 0) { + dev_err(&pdev->dev, + "%s snd_soc_of_get_dai_name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_dai_capture; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

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[alsa-devel] ASoC: TLV320AIC3x: Adding additional functionality for 3106 with [Patch] for discuss
by Timur Karaldin 18 Apr '16

18 Apr '16

Hi, colleagues! I'm newbie with ALSA devel. I need to get extra functionality from codec through ALSA mixer. Task: 1. making possibility to set-up gain for each input (XXX_2_LADCCTRL and XXX_2_RADCCTRL registers) 2. making possibility to set-up debounce for button and headset detection ( HEADSET_DETECT_CTRL_A register) 3. making possibility to enable high-power AC-coupled output (HEADSET_DETECT_CTRL_A register) 4. making possibility to enable stereo pseudo differential output (HEADSET_DETECT_CTRL_B register) 5. enable headset detection (HEADSET_DETECT_CTRL_A) 6. getting headset detect status (HEADSET_DETECT_CTRL_A) Well, my patch works in generally, but I don't like it very much, because: 1. All controls for input amplifiers' gain are shown in Alsamixer-Playback section. But not only mine, capture controls from base codec's version are shown in playback section too, for example (Right or Left) PGA Mixer Line1(L or R)/2(L or R)/MIC3(L or R), which is definitely capture settings. Should I worried about it? because, as I understand, it could not be setted correctly during changing playback/capture mode for power-saving reason. 2. "Right PGA Mixer Line1R" control (and similar) using the same bits in LINE1R_2_RADCCTRL register as my "Line1R RADC Gain" control (and similar) and put there b"1111" when user switch disconnect it from ADC. So my control is not shown when user disconnect it from ADC. 3. When I start/stop arecord some of my options were reset (debounces, headset enable, high-power AC-coupled, pseudo differential output). When I set up it in shell and then start arecord, it's reset again. Is it the reason because of displaying controls in playback mixer section? I will be very pleased for any advise, thank you! here is my patch for tlv320aic3x.c --- sound/soc/codecs/tlv320aic3x.c | 55 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 52 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d7349bc..27d031e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -122,11 +122,24 @@ static const struct reg_default aic3x_reg[] = { { 108, 0x00 }, { 109, 0x00 }, }; +static bool aic3106_volatile(struct device *dev, unsigned int reg) +{ + switch(reg) + { + case AIC3X_HEADSET_DETECT_CTRL_A: + case AIC3X_HEADSET_DETECT_CTRL_B: + return true; + default: + return false; + } +} + static const struct regmap_config aic3x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DAC_ICC_ADJ, + .volatile_reg = aic3106_volatile, .reg_defaults = aic3x_reg, .num_reg_defaults = ARRAY_SIZE(aic3x_reg), .cache_type = REGCACHE_RBTREE, @@ -196,7 +209,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -270,6 +283,26 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), }; +static const char *aic3x_adc_gain[] = { "0dB", "-1.5dB", "-3dB", "-4.5dB", "-6dB", "-7.5dB", "-9dB", "-10.5dB", "-12dB" }; +static const struct soc_enum aic3x_adc_gain_enum[] = { + SOC_ENUM_SINGLE(MIC3LR_2_LADC_CTRL, 4, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(MIC3LR_2_RADC_CTRL, 0, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), + SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), +}; +static const char *aic3x_headset_debounce[] = { "16ms", "32ms", "64ms", "128ms", "256ms", "512ms" }; +static const struct soc_enum aic3x_headset_debounce_enum[] = { + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 2, 6, aic3x_headset_debounce), +}; +static const char *aic3x_button_debounce[] = { "0ms", "8ms", "16ms", "32ms" }; +static const struct soc_enum aic3x_button_debounce_enum[] = { + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 0, 4, aic3x_button_debounce), +}; + + + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -399,6 +432,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + /* Additional controls */ + SOC_ENUM("Mic3L LADC Gain", aic3x_adc_gain_enum[0]), + SOC_ENUM("Mic3R RADC Gain", aic3x_adc_gain_enum[1]), + SOC_ENUM("Line1L LADC Gain", aic3x_adc_gain_enum[2]), + SOC_ENUM("Line1R RADC Gain", aic3x_adc_gain_enum[3]), + SOC_ENUM("Line2L LADC Gain", aic3x_adc_gain_enum[4]), + SOC_ENUM("Line2R RADC Gain", aic3x_adc_gain_enum[5]), + SOC_ENUM("Headset Debounce", aic3x_headset_debounce_enum), + SOC_ENUM("Button Debounce", aic3x_button_debounce_enum), + SOC_SINGLE("Headset Detected", AIC3X_HEADSET_DETECT_CTRL_A, AIC3X_HEADSET_DETECT_SHIFT, AIC3X_HEADSET_DETECT_MASK, 0), + SOC_SINGLE("Headset Detect Enable", AIC3X_HEADSET_DETECT_CTRL_A, 7, 1, 0), + SOC_SINGLE("High power output Ac-coupled", AIC3X_HEADSET_DETECT_CTRL_B, 7, 1, 0), + SOC_SINGLE("Stereo pseudodifferential output", AIC3X_HEADSET_DETECT_CTRL_B, 3, 1, 0), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { @@ -614,7 +660,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), - /* Mic Bias */ + /* Mic Bias */ SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -692,7 +738,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, + {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, @@ -814,6 +860,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"}, {"Right HP Com", NULL, "Right HPCOM Mux"}, {"HPRCOM", NULL, "Right HP Com"}, + }; static const struct snd_soc_dapm_route intercon_mono[] = { @@ -918,7 +965,9 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data = (LDAC2LCH | RDAC2RCH); data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; if (params_rate(params) >= 64000) + { data |= DUAL_RATE_MODE; + } snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); /* codec sample rate select */ -- and tlv320aic3x.h --- sound/soc/codecs/tlv320aic3x.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e521ac3..8ff42d2 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -271,6 +271,12 @@ enum { AIC3X_BUTTON_DEBOUNCE_32MS = 3 }; +typedef struct { + struct platform_device* pdev; + struct proc_dir_entry *proc_value; + struct snd_soc_codec *codec; +} aic3106_detect_t; + #define AIC3X_HEADSET_DETECT_ENABLED 0x80 #define AIC3X_HEADSET_DETECT_SHIFT 5 #define AIC3X_HEADSET_DETECT_MASK 3 -- -- С уважением, Руководитель проектов Морские комплексы и системы плюс Каралдин Т.М. ---------------------------------------------------------------------------------- Best regards, Projects director Marine complexes and systems plus Timur Karaldin

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Add SPI support
by Jeremy McDermond 18 Apr '16

18 Apr '16

The TLV320AIC32x4 support the control channel over either SPI or I2C. Added the proper interfacing with regmap to make the driver able to handle either possibility. Signed-off-by: Jeremy McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/Kconfig | 5 +- sound/soc/codecs/tlv320aic32x4.c | 181 +++++++++++++++++++++++++++++++++++---- 2 files changed, 165 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 907c804..1ddf7b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,7 +129,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4 if SND_SOC_I2C_AND_SPI select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -768,7 +768,8 @@ config SND_SOC_TLV320AIC31XX select REGMAP_I2C config SND_SOC_TLV320AIC32X4 - tristate + tristate "Texas Instruments TLV320AIC32x4 CODECs" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_TLV320AIC3X tristate "Texas Instruments TLV320AIC3x CODECs" diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f2d3191..1c888b7 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -31,6 +31,7 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/cdev.h> #include <linux/slab.h> #include <linux/clk.h> @@ -287,15 +288,6 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { }, }; -static const struct regmap_config aic32x4_regmap = { - .reg_bits = 8, - .val_bits = 8, - - .max_register = AIC32X4_RMICPGAVOL, - .ranges = aic32x4_regmap_pages, - .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), -}; - static inline int aic32x4_get_divs(int mclk, int rate) { int i; @@ -668,7 +660,7 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { }; static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4, - struct device_node *np) + struct device_node *np, struct device *dev) { aic32x4->swapdacs = false; aic32x4->micpga_routing = 0; @@ -777,6 +769,23 @@ error_ldo: return ret; } +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static const struct regmap_config aic32x4_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AIC32X4_RMICPGAVOL, + .ranges = aic32x4_regmap_pages, + .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), +}; + static int aic32x4_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -790,7 +799,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, if (aic32x4 == NULL) return -ENOMEM; - aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); + aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_i2c_regmap); if (IS_ERR(aic32x4->regmap)) return PTR_ERR(aic32x4->regmap); @@ -802,7 +811,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->micpga_routing = pdata->micpga_routing; aic32x4->rstn_gpio = pdata->rstn_gpio; } else if (np) { - ret = aic32x4_parse_dt(aic32x4, np); + ret = aic32x4_parse_dt(aic32x4, np, &i2c->dev); if (ret) { dev_err(&i2c->dev, "Failed to parse DT node\n"); return ret; @@ -862,12 +871,6 @@ static const struct i2c_device_id aic32x4_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); -static const struct of_device_id aic32x4_of_id[] = { - { .compatible = "ti,tlv320aic32x4", }, - { /* senitel */ } -}; -MODULE_DEVICE_TABLE(of, aic32x4_of_id); - static struct i2c_driver aic32x4_i2c_driver = { .driver = { .name = "tlv320aic32x4", @@ -877,8 +880,148 @@ static struct i2c_driver aic32x4_i2c_driver = { .remove = aic32x4_i2c_remove, .id_table = aic32x4_i2c_id, }; +#endif -module_i2c_driver(aic32x4_i2c_driver); +#if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config aic32x4_spi_regmap = { + .reg_bits = 7, + .pad_bits = 1, + .val_bits = 8, + .read_flag_mask = 0x01, + + .max_register = AIC32X4_RMICPGAVOL, + .ranges = aic32x4_regmap_pages, + .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), +}; + +static int aic32x4_spi_probe(struct spi_device *spi) +{ + struct aic32x4_pdata *pdata = spi->dev.platform_data; + struct aic32x4_priv *aic32x4; + struct device_node *np = spi->dev.of_node; + int ret; + + aic32x4 = devm_kzalloc(&spi->dev, sizeof(struct aic32x4_priv), + GFP_KERNEL); + if (aic32x4 == NULL) + return -ENOMEM; + + aic32x4->regmap = devm_regmap_init_spi(spi, &aic32x4_spi_regmap); + if (IS_ERR(aic32x4->regmap)) + return PTR_ERR(aic32x4->regmap); + + spi_set_drvdata(spi, aic32x4); + + if (pdata) { + aic32x4->power_cfg = pdata->power_cfg; + aic32x4->swapdacs = pdata->swapdacs; + aic32x4->micpga_routing = pdata->micpga_routing; + aic32x4->rstn_gpio = pdata->rstn_gpio; + } else if (np) { + ret = aic32x4_parse_dt(aic32x4, np, &spi->dev); + if (ret) { + dev_err(&spi->dev, "Failed to parse DT node\n"); + return ret; + } + } else { + aic32x4->power_cfg = 0; + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + aic32x4->rstn_gpio = -1; + } + + aic32x4->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(aic32x4->mclk)) { + dev_err(&spi->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); + return PTR_ERR(aic32x4->mclk); + } + + if (gpio_is_valid(aic32x4->rstn_gpio)) { + ret = devm_gpio_request_one(&spi->dev, aic32x4->rstn_gpio, + GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); + if (ret != 0) + return ret; + } + + ret = aic32x4_setup_regulators(&spi->dev, aic32x4); + if (ret) { + dev_err(&spi->dev, "Failed to setup regulators\n"); + return ret; + } + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_aic32x4, &aic32x4_dai, 1); + if (ret) { + dev_err(&spi->dev, "Failed to register codec\n"); + aic32x4_disable_regulators(aic32x4); + return ret; + } + + spi_set_drvdata(spi, aic32x4); + + return 0; +} + +static int aic32x4_spi_remove(struct spi_device *dev) +{ + struct aic32x4_priv *aic32x4 = spi_get_drvdata(dev); + + aic32x4_disable_regulators(aic32x4); + + snd_soc_unregister_codec(&dev->dev); + return 0; +} + +static const struct spi_device_id aic32x4_spi_id[] = { + { "tlv320aic32x4", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, aic32x4_spi_id); + +static struct spi_driver aic32x4_spi_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_spi_probe, + .remove = aic32x4_spi_remove, + .id_table = aic32x4_spi_id, +}; +#endif + +static int __init aic32x4_modinit(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&aic32x4_i2c_driver); + if (ret) + pr_err("Failed to register tlv320aic32x4 I2C driver: %d\n", + ret); +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&aic32x4_spi_driver); + if (ret) + pr_err("Failed to register tlv320aic32x4 SPI driver: %d\n", + ret); +#endif + + return ret; +} +module_init(aic32x4_modinit); + +static void __exit aic32x4_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic32x4_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&aic32x4_spi_driver); +#endif +} +module_exit(aic32x4_exit); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); -- 2.5.0

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Re: [alsa-devel] [PATCH] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 18 Apr '16

18 Apr '16

Mark Brown wrote: > On Mon, Apr 18, 2016 at 11:39:50AM +0200, Peter Rosin wrote: > > > +I2S > > + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. > > +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- > > There is an actual spec for I2S which is going to be clearer than ASCII > art: > > https://www.sparkfun.com/datasheets/BreakoutBoards/I2SBUS.pdf > > is the first hit on google (there's a bunch of electrical stuff in there > that gets roundly ignord these days but the clocking stuff is in there). Yes, but where are the equivalent docs for the other formats? > In general I'd rather have some words rather than just pictures, ASCII > art as the only thing is pretty hard to read. I'll post a v2 with some words about each format. Or will you reject anything with ASCII art? Cheers, Peter

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