August 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH 08/10] ASoC: topology: change pass number of DAI smaller than graph
by mengdong.lin＠intel.com 15 Aug '15

15 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The PCM DAIs need to be loaded and added to ASoC core ealier than the graph (route). Otherwise, adding routes will fail for missing DAIs. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1b6728b..0f8fa0d 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -45,12 +45,12 @@ #define SOC_TPLG_PASS_VENDOR 1 #define SOC_TPLG_PASS_MIXER 2 #define SOC_TPLG_PASS_WIDGET 3 -#define SOC_TPLG_PASS_GRAPH 4 -#define SOC_TPLG_PASS_PINS 5 -#define SOC_TPLG_PASS_PCM_DAI 6 +#define SOC_TPLG_PASS_PCM_DAI 4 +#define SOC_TPLG_PASS_GRAPH 5 +#define SOC_TPLG_PASS_PINS 6 #define SOC_TPLG_PASS_START SOC_TPLG_PASS_MANIFEST -#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PCM_DAI +#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PINS struct soc_tplg { const struct firmware *fw; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] USB audio fixes
by Pierre-Louis Bossart 14 Aug '15

14 Aug '15

While trying to understand how asynchronous endpoints are supported, I identified two major issues. On my M-Audio transit, the following script tries to play using the 3 different endpoints supported and fails (error and no audio) after the first playback without the patches. With the patches on top of 4.1.5 or Takashi's for-next (4.2.0-rc5), I stopped the audio after an hour. while true do aplay -d5 -Dhw:2,0 ~/16_48.wav # play 48kHz, 16 bit on async ep aplay -d5 -Dhw:2,0 ~/24_48.wav # play 48kHz, 24 bit on 'none'/async ep aplay -d5 -Dhw:2,0 ~/24_96.wav # play 96 kHz, 24 bit on adaptive ep aplay -d5 -Dhw:2,0 ~/24_48.wav aplay -d5 -Dhw:2,0 ~/16_48.wav aplay -d5 -Dhw:2,0 ~/24_96.wav done The first issue is pretty bad with corrupted pointers. When transitioning from a alternate setting with a sync_endpoint to another which doesn't have one by construction (synchronous, adaptive, async on capture), the pointers are not reset and the code tries to set the parameters of a non-existent endpoint. The second issue is that the M-audio Transit descriptors are illegal but the logic can be changed without harm. I sometimes hear noise on playback start, probably the code needs to be changed to add enough silent samples to let the PLL lock. I haven't had time to work on this and this is more of an enhancement. Comments welcome. Pierre-Louis Bossart (2): ALSA: usb: fix corrupted pointers due to interface setting change ALSA: usb: handle descriptor with SYNC_NONE illegal value sound/usb/pcm.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 14 Aug '15

14 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../devicetree/bindings/sound/ics43432.txt | 17 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 111 ++++++++++++++++++++ 5 files changed, 135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..902daae --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,111 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + unsigned int rate = params_rate(params); + + if (rate < ICS43432_RATE_MIN || rate > ICS43432_RATE_MAX) + return -EINVAL; + + return 0; +} + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .hw_params = ics43432_hw_params, + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPL2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 2

[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- Changes in v2: - Removed superfluous int32 checks - Check control value length in function - Compare against 0 - Use a tmp variable to access audio_feature_info[control-1] Thanks for review by Takashi Iwai :) --- sound/usb/mixer.c | 64 ++++++++++++++++++++++++++++++++++++++----------------- sound/usb/mixer.h | 2 ++ 2 files changed, 46 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..7046f54 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -282,6 +282,21 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) return val; } +static int uac2_ctl_value_size(int val_type) +{ + switch (val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + return 4; + case USB_MIXER_S16: + case USB_MIXER_U16: + return 2; + default: + return 1; + } + return 0; /* unreachable */ +} + /* * retrieve a mixer value @@ -328,14 +343,14 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + size = uac2_ctl_value_size(cval->val_type); } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +461,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + val_len = uac2_ctl_value_size(cval->val_type); /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +483,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +815,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1216,6 +1233,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, int readonly_mask) { struct uac_feature_unit_descriptor *desc = raw_desc; + struct usb_feature_control_info *ctl_info; unsigned int len = 0; int mapped_name = 0; int nameid = uac_feature_unit_iFeature(desc); @@ -1241,7 +1259,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + ctl_info = &audio_feature_info[control-1]; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = ctl_info->type; + else /* UAC_VERSION_2 */ + cval->val_type = ctl_info->type_uac2 >= 0 ? + ctl_info->type_uac2 : ctl_info->type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 83 +++++++++++++++++++++++++++++++++++++++++-------------- sound/usb/mixer.h | 2 ++ 2 files changed, 65 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..675725b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -233,6 +233,14 @@ static int convert_signed_value(struct usb_mixer_elem_info *cval, int val) if (val >= 0x8000) val -= 0x10000; break; + case USB_MIXER_U32: + val &= 0xffffffff; + break; + case USB_MIXER_S32: + val &= 0xffffffff; + if (val >= 0x80000000) + val -= 0x100000000; + break; } return val; } @@ -253,6 +261,9 @@ static int convert_bytes_value(struct usb_mixer_elem_info *cval, int val) case USB_MIXER_S16: case USB_MIXER_U16: return val & 0xffff; + case USB_MIXER_S32: + case USB_MIXER_U32: + return val & 0xffffffff; } return 0; /* not reached */ } @@ -328,14 +339,26 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + size = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + size = 2; + break; + default: + size = 1; + break; + } } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +477,19 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + val_len = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + val_len = 2; + break; + default: + val_len = 1; + break; + } /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +503,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +835,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1241,7 +1278,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = audio_feature_info[control-1].type; + else /* UAC_VERSION_2 */ + cval->val_type = audio_feature_info[control-1].type_uac2 > -1 ? + audio_feature_info[control-1].type_uac2 : + audio_feature_info[control-1].type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

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[alsa-devel] [PATCH v2 1/3] ASoC: dapm: dapm_dai_get_connected_widgets: Fix missing mutex unlock
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

Make sure to unlock the DAPM mutex when dapm_widget_list_create() fails. This means the function will now generate a trace_snd_soc_dapm_connected event, even if the creation of the list fails. But that was the behavior before the patch that introduced the unlock issue, so that should be fine. Fixes: 1ce43acff0c0 ("ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- New in v2 --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7482c5d..c391bfe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1191,7 +1191,7 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, ret = dapm_widget_list_create(list, &widgets); if (ret) - return ret; + paths = ret; trace_snd_soc_dapm_connected(paths, stream); mutex_unlock(&card->dapm_mutex); -- 2.1.4

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[alsa-devel] [PATCH 1/5] ASoC: dapm: Drop always true checks
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

list_first_entry() always returns non NULL and since the code previously checked that list is not empty it will also be a valid pointer. Furthermore a path has always a sink or a source widget. So both checks are redundant and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7f834a0..10f7c7c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3485,11 +3485,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, list_source); - if (WARN_ON(!source_p || !sink_p) || - WARN_ON(!sink_p->source || !source_p->sink) || - WARN_ON(!source_p->source || !sink_p->sink)) - return -EINVAL; - source = source_p->source->priv; sink = sink_p->sink->priv; -- 2.1.4

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[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Specify codec_dai_name for DAI links
by Nicolin Chen 14 Aug '15

14 Aug '15

The dev->name of CODEC might not be identical to its codec_dai_name, so using dev->name to probe the CODEC dai is not a correct approach. This patch specifies each supporting codec_dai_name instead of using dev->name any more. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..040362f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -407,6 +407,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; struct clk *codec_clk; + const char *codec_dai_name; u32 width; int ret; @@ -459,6 +460,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + codec_dai_name = "cs42888"; priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; @@ -467,9 +469,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + codec_dai_name = "wm8962"; priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; @@ -521,7 +525,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].codec_dai_name = codec_dai_name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +534,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; -- 1.9.1

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[alsa-devel] [PATCH] Add rudimentary device tree support for SSM2518
by Ricard Wanderlof 13 Aug '15

13 Aug '15

For some reason the SSM2518 codec driver doesn't have DT support, even though there are bindings described for it in Documentation/devicetree/sound/ssm2518.txt . This patch adds rudimentary DT support for the compatible property which is a step forward. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/ssm2518.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..ecf9996 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +810,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +828,9 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(ssm2518_dt_ids), +#endif }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH] Add rudimentary DT support to SSM2518 codec driver
by Ricard Wanderlof 13 Aug '15

13 Aug '15

For some reason, the SSM2518 codec driver has device tree bindings specified (Documentation/devicetree/bindings/sound/ssm2518.txt) but the actual driver doesn't seem to support DT. This patch adds rudimentary support for the compatible property so that the driver can be used in a DT based setup. --- sound/soc/codecs/ssm2518.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..ecf9996 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +810,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +828,9 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(ssm2518_dt_ids), +#endif }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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