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

[alsa-devel] [PATCH 0/13] ASoC: Renesas adds DPCM based sampling rate convert
by Kuninori Morimoto 16 Mar '15

16 Mar '15

Hi Mark These are DPCM based sampling rate convert support patch-set for Rensas sound driver. Then, it doesn't use simple-card. it uses new Renesas specific sound card driver, but it is copied from simple-card. Current ASoC DPCM needs route settings for FE/BE, but sometimes it will have trouble, because many drivers are using same name for this routing. This patch set solved this issue too. 1 - 2 : prepare patch for Renesas new sound card driver 3 - 4 : DAPM routing issue solution patches 5 - 6 : new Renesas sound card driver 7 - 11 : fixup/tidyup patches for rsnd 12 - 13 : DPCM support patch for rsnd Kuninori Morimoto (13): 1) ASoC: core: remove of_xlate_dai_name from snd_soc_component_driver 2) ASoC: core: add snd_soc_get_dai_info() to get driver infomation 3) ASoC: dapm: add snd_soc_dapm_route_scan macro for DAPM route 4) ASoC: dapm: add driver prefix matching for snd_soc_dapm_add_route() 5) ASoC: rsnd-dpcm-card: add Renesas sound card support for DPCM 6) ASoC: rsnd-dpcm-card: add .be_hw_params_fixup support for convert rate 7) ASoC: rsnd: no more SSI restart when unusual situation 8) ASoC: rsnd: no more SRC restart when unusual situation 9) ASoC: rsnd: tidyup error message format 10) ASoC: rsnd: show debug info for sampling rate convert 11) ASoC: rsnd: don't call clk_prepare_enable/unprepare() from inside spin lock 12) ASoC: rsnd: add dai_link stream name 13) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsnd-dpcm-card.txt | 72 ++++++++++++++ include/sound/soc.h | 6 +- sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 + sound/soc/sh/rcar/core.c | 48 ++++++++- sound/soc/sh/rcar/dvc.c | 12 +-- sound/soc/sh/rcar/rsnd-dpcm-card.c | 491 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 16 ++- sound/soc/sh/rcar/src.c | 63 ++++++++++-- sound/soc/sh/rcar/ssi.c | 31 ++++-- sound/soc/soc-core.c | 71 ++++++++------ sound/soc/soc-dapm.c | 33 +++++-- 13 files changed, 780 insertions(+), 75 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 6/6] ESI W192M : Add sampling rate control of the ADC/DAC
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index a6e7afa..f3e3879 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include <linux/init.h> #include <sound/core.h> #include <sound/tlv.h> +#include <linux/slab.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -388,6 +446,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -487,21 +583,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } -- 2.1.4

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[alsa-devel] [PATCH 4/6] ESI W192M : Add TLV support for control value in dB scale
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add TLV support to control volume using dB scale for input and ouput on ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 59483b4..c7ffafa 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> #include "ice1712.h" #include "envy24ht.h" @@ -380,17 +381,25 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/*Limits value in dB for fader*/ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ static struct snd_kcontrol_new stac9640_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Switch", .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +428,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +448,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; -- 2.1.4

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[alsa-devel] [PATCH 3/6] ESI W192M : Enable midi i/o of port envy24 chip as available
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Enable midi i/o port of envy24 chip as their are available on ESI W192M soundcard. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6e1026e..59483b4 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,7 +485,8 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_SYSCONF] = 0x67, /*SYSCONF: clock 192KHz, mpu401, + 4ADC, 8DAC */ [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ -- 2.1.4

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[alsa-devel] [PATCH 2/6] ESI W192M : Update eeprom structure to C99 standard
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Update eeprom structure to C99 standard to be compliant with change in alsa. It's just a notation change, no configuration change. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index f65ac19..6e1026e 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,19 +485,19 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ + [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, }; -- 2.1.4

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[alsa-devel] [PATCH 1/6] ESI W192M : Correct copy/paste from prodigy driver
by klem.dev＠gmail.com 16 Mar '15

16 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Correct copy/paste name from prodigy driver, no behaviour change, only name. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..f65ac19 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -475,7 +475,7 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); -- 2.1.4

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[alsa-devel] Issues and/or possible bugs in alsa
by Yomi Ogunwumi 15 Mar '15

15 Mar '15

Hello. I seem to have run into a few issues with alsa. First, when I plug my headphones into my laptop, sound continues to play out of the speakers and doesn't switch over to (or unmute) the headphones. Auto-Mute Mode is enabled in alsamixer. The same thing happens when I unplug the headphones, it doesn't switch over to the speakers. I have to do the same thing I described above, (no unmuting, I just have to turn up the volume for the speakers in alsamixer.) Second. When I plug in these headphones, I can't use the internal microphone on this laptop, it mutes. (AFAICT) These headphones do not have a microphone built in, they're just an ordinary pair of headphones. However, I was given a workaround in #alsa on freenode. In order to use the laptop's internal microphone with my headphones plugged in, I have to do : sudo hda-verb /dev/snd/hwC1D0 0x22 SET_CONNECT_SEL 6 — it works, but it isn't really ideal. alsa-info is here : https://gist.github.com/Yomi0/9bfc3e72588cf8a35811 The first file in that gist is alsa-info with headphones and the second is without headphones. Third. This is just a question. Is this : https://bugs.freedesktop.org/show_bug.cgi?id=86676 actually a pulseaudio bug, or is it an issue with alsa? I'm only asking since Raymond linked something that seemed to belong to the alsa project. Thank you. -- *Yomi*

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[alsa-devel] [PATCH] ASoC: fsl-ssi: Fix bitclock calculation for master mode
by Marc Kleine-Budde 15 Mar '15

15 Mar '15

From: Sascha Hauer <s.hauer(a)pengutronix.de> in fsl_ssi_set_bclk we are looking for the best divider by iterating over all possible dividers. The divider is in the range 1..256, so make this clear by using 'div' as name for the loop counter and iterate from 1 to 256 instead of 0 to 255. This makes obvious that tmprate is calculated wrong, it has to be freq * factor * div, not freq * factor * (i + 2). Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Marc Kleine-Budde <mkl(a)pengutronix.de> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b9fabbf69db6..b8af08e49085 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -578,7 +578,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct regmap *regs = ssi_private->regs; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; - u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, div; unsigned long clkrate, baudrate, tmprate; u64 sub, savesub = 100000; unsigned int freq; @@ -602,8 +602,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; - for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + for (div = 1; div < 256; div++) { + tmprate = freq * factor * div; if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); @@ -618,7 +618,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, continue; clkrate /= factor; - afreq = clkrate / (i + 1); + afreq = clkrate / div; if (freq == afreq) sub = 0; @@ -636,7 +636,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (sub < savesub) { baudrate = tmprate; savesub = sub; - pm = i; + pm = div - 1; } /* We are lucky */ -- 2.1.4

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Re: [alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Raymond Yau 15 Mar '15

15 Mar '15

>> >> >> >> >> > > >> >> >> > > > amixer: Control hw:2 element read error: Invalid argument >> >> >> > > > usb 1-1.3: 10:0: cannot get min/max values for control 3 (id 10) >> >> >> > > > usb 1-1.3: 2:1: cannot get freq at ep 0x1 >> >> >> >> numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument >> >> >> >> > AudioControl Interface Descriptor: >> >> > bLength 13 >> >> > bDescriptorType 36 >> >> > bDescriptorSubtype 4 (MIXER_UNIT) >> >> > Warning: Descriptor too short >> >> > bUnitID 10 >> >> > bNrInPins 2 >> >> > baSourceID( 0) 9 >> >> > baSourceID( 1) 1 >> >> > bNrChannels 8 >> >> > wChannelConfig 0x063f >> >> > Left Front (L) >> >> > Right Front (R) >> >> > Center Front (C) >> >> > Low Freqency Enhancement (LFE) >> >> > Left Surround (LS) >> >> > Right Surround (RS) >> >> > Side Left (SL) >> >> > Side Right (SR) >> >> > iChannelNames 0 >> >> > bmControls 0x00 >> >> > bmControls 0x00 >> >> > iMixer 33 (error) Refer to usb audio specification 1.0 Table 4-5: Mixer Unit Descriptor bLength = 10 + p + N = 14 Number of Input Pins of this Unit: p > >> > bNrInPins 2 The ID of the Unit or Terminal to which the first and second Input Pins of this Mixer Unit is connected. > >> > baSourceID( 0) 9 >> >> > baSourceID( 1) 1 Number of logical output channels in the Mixer’s output audio channel cluster. >> >> > bNrChannels 8 Describes the spatial location of the logical channels. >> >> > wChannelConfig 0x063f >> >> > Left Front (L) >> >> > Right Front (R) >> >> > Center Front (C) >> >> > Low Freqency Enhancement (LFE) >> >> > Left Surround (LS) >> >> > Right Surround (RS) >> >> > Side Left (SL) >> >> > Side Right (SR) Bit map indicating which mixing Controls are programmable. >> >> > bmControls 0x00 >> >> > bmControls 0x00 The Mixer Unit Descriptor reports which Controls are programmable in the bmControls bitmap field. This bitmap must be interpreted as a two-dimensional bit array that has a row for each logical input channel and a column for each logical output channel. If a bit at position [u, v] is set, this means that the Mixer Unit contains a programmable mixing Control that connects input channel u to output channel v. If bit [u, v] is clear, this indicates that the connection between input channel u and output channel v is non-programmable. Its fixed value can be retrieved through the appropriate request. The valid range for u is from one to n. The valid range for v is from one to m. The bmControls field stores the bit array row after row where the MSb of the first byte corresponds to the connection between input channel 1 and output channel 1. If (n x m) is not an integer multiple of 8, the bit array is padded with zeros until an integer number of bytes is occupied. The number of bytes used to store the bit array, N, can be calculated as follows: IF ((n x m) MOD 8) <> 0 THEN N = ((n x m) DIV 8) + 1 ELSE N = ((n x m) DIV 8) Index of a string descriptor, describing the name of the first logical channel. >> > iChannelNames 0 Index of a string descriptor, describing the Mixer Unit. >> >> > iMixer 33 >> >> >> if (check_matrix_bitmap(c, ich, och, num_outs)) { >> ich_has_controls = 1; >> break; >> } >> } This mean value returned by check_matrix_bitmap() is incorrect as >> >> > bmControls 0x00 >> >> > bmControls 0x00 >> if (ich_has_controls) >> build_mixer_unit_ctl(state, desc, pin, ich, >> unitid, &iterm); >> } >> } >> return 0; >> } > >

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[alsa-devel] [PATCH 5/6] ESI W192M : Add text Line in/Mic for selecting input gain state
by klem.dev＠gmail.com 14 Mar '15

14 Mar '15

From: Clément Guedez <klem.dev(a)gmail.com> Add text Line in/Mic for selecting input gain state in mixer for ESI W192M. Signed-off-by: Clément Guedez <klem.dev(a)gmail.com> diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 3af9daf..497ff1a 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,23 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { "Line In", "Mic" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items + - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +369,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +385,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) -- 2.1.4

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