May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: uniphier: add digital output volume for UniPhier sound system
by Katsuhiro Suzuki 09 May '18

09 May '18

This patch adds controllers for digital volume of PCM output. Volume effects simply linear, not dB scale as follows: Gained PCM = Original * 0x4000 / Volume The value range of volume is from 0x0001 to 0xffff. 0x0000 works as mute. Initial value is 0x4000 (+0dB). Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 58 ++++++++++++++ sound/soc/uniphier/aio-cpu.c | 140 ++++++++++++++++++++++++++++++++++ sound/soc/uniphier/aio-reg.h | 33 ++++++-- sound/soc/uniphier/aio.h | 7 ++ 4 files changed, 233 insertions(+), 5 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index e37b80921abb..638cb3fc5f7b 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -659,6 +659,64 @@ void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable) } } +/** + * aio_port_get_volume - get volume of AIO port block + * @sub: the AIO substream pointer + * + * Return: current volume, range is 0x0000 - 0xffff + */ +int aio_port_get_volume(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 v; + + regmap_read(r, OPORTMXTYVOLGAINSTATUS(sub->swm->oport.map, 0), &v); + + return FIELD_GET(OPORTMXTYVOLGAINSTATUS_CUR_MASK, v); +} + +/** + * aio_port_set_volume - set volume of AIO port block + * @sub: the AIO substream pointer + * @vol: target volume, range is 0x0000 - 0xffff. + * + * Change digital volume and perfome fade-out/fade-in effect for specified + * output slot of port. Gained PCM value can calculate as the following: + * Gained = Original * vol / 0x4000 + */ +void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) +{ + struct regmap *r = sub->aio->chip->regmap; + int oport_map = sub->swm->oport.map; + int cur, diff, slope = 0, fs; + + if (sub->swm->dir == PORT_DIR_INPUT) + return; + + cur = aio_port_get_volume(sub); + diff = abs(vol - cur); + fs = params_rate(&sub->params); + if (fs) + slope = diff / AUD_VOL_FADE_TIME * 1000 / fs; + slope = max(1, slope); + + regmap_update_bits(r, OPORTMXTYVOLPARA1(oport_map, 0), + OPORTMXTYVOLPARA1_SLOPEU_MASK, slope << 16); + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_TARGET_MASK, vol); + + if (cur < vol) + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_FADE_MASK, + OPORTMXTYVOLPARA2_FADE_FADEIN); + else + regmap_update_bits(r, OPORTMXTYVOLPARA2(oport_map, 0), + OPORTMXTYVOLPARA2_FADE_MASK, + OPORTMXTYVOLPARA2_FADE_FADEOUT); + + regmap_write(r, AOUTFADECTR0, BIT(oport_map)); +} + /** * aio_if_set_param - set parameters of AIO DMA I/F block * @sub: the AIO substream pointer diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 00b6441bf195..80daec17be25 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -32,6 +32,35 @@ static bool is_valid_pll(struct uniphier_aio_chip *chip, int pll_id) return chip->plls[pll_id].enable; } +/** + * find_volume - find volume supported HW port by HW port number + * @chip: the AIO chip pointer + * @oport_hw: HW port number, one of AUD_HW_XXXX + * + * Find AIO device from device list by HW port number. Volume feature is + * available only in Output and PCM ports, this limitation comes from HW + * specifications. + * + * Return: The pointer of AIO substream if successful, otherwise NULL on error. + */ +static struct uniphier_aio_sub *find_volume(struct uniphier_aio_chip *chip, + int oport_hw) +{ + int i; + + for (i = 0; i < chip->num_aios; i++) { + struct uniphier_aio_sub *sub = &chip->aios[i].sub[0]; + + if (!sub->swm) + continue; + + if (sub->swm->oport.hw == oport_hw) + return sub; + } + + return NULL; +} + static bool match_spec(const struct uniphier_aio_spec *spec, const char *name, int dir) { @@ -287,6 +316,7 @@ static int uniphier_aio_hw_params(struct snd_pcm_substream *substream, sub->setting = 1; aio_port_reset(sub); + aio_port_set_volume(sub, sub->vol); aio_src_reset(sub); return 0; @@ -373,6 +403,8 @@ int uniphier_aio_dai_probe(struct snd_soc_dai *dai) sub->swm = &spec->swm; sub->spec = spec; + + sub->vol = AUD_VOL_INIT; } aio_iecout_set_enable(aio->chip, true); @@ -449,8 +481,116 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) } EXPORT_SYMBOL_GPL(uniphier_aio_dai_resume); +static int uniphier_aio_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = AUD_VOL_MAX; + + return 0; +} + +static int uniphier_aio_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct uniphier_aio_chip *chip = snd_soc_component_get_drvdata(comp); + struct uniphier_aio_sub *sub; + int oport_hw = kcontrol->private_value; + + sub = find_volume(chip, oport_hw); + if (!sub) + return 0; + + ucontrol->value.integer.value[0] = sub->vol; + + return 0; +} + +static int uniphier_aio_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct uniphier_aio_chip *chip = snd_soc_component_get_drvdata(comp); + struct uniphier_aio_sub *sub; + int oport_hw = kcontrol->private_value; + + sub = find_volume(chip, oport_hw); + if (!sub) + return 0; + + if (sub->vol == ucontrol->value.integer.value[0]) + return 0; + sub->vol = ucontrol->value.integer.value[0]; + + aio_port_set_volume(sub, sub->vol); + + return 0; +} + +static const struct snd_kcontrol_new uniphier_aio_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "HPCMOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_HPCMOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT2 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT2, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "PCMOUT3 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_PCMOUT3, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "HIECOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_HIECOUT1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .name = "IECOUT1 Volume", + .info = uniphier_aio_vol_info, + .get = uniphier_aio_vol_get, + .put = uniphier_aio_vol_put, + .private_value = AUD_HW_IECOUT1, + }, +}; + static const struct snd_soc_component_driver uniphier_aio_component = { .name = "uniphier-aio", + .controls = uniphier_aio_controls, + .num_controls = ARRAY_SIZE(uniphier_aio_controls), }; int uniphier_aio_probe(struct platform_device *pdev) diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 511ea3c01847..45fdc6ae358a 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -169,6 +169,7 @@ #define PBINMXPAUSECTR1(n) (0x20208 + 0x40 * (n)) /* AOUT */ +#define AOUTFADECTR0 0x40020 #define AOUTENCTR0 0x40040 #define AOUTENCTR1 0x40044 #define AOUTENCTR2 0x40048 @@ -179,6 +180,9 @@ #define AOUTSRCRSTCTR1 0x400c4 #define AOUTSRCRSTCTR2 0x400c8 +/* AOUT PCMOUT has 5 slots, slot0-3: D0-3, slot4: DMIX */ +#define OPORT_SLOT_MAX 5 + /* AOUT(PCMOUTN) */ #define OPORTMXCTR1(n) (0x42000 + 0x400 * (n)) #define OPORTMXCTR1_I2SLRSEL_MASK (0x11 << 10) @@ -359,11 +363,30 @@ #define OPORTMXMASK_XCKMSK_ON (0x0 << 0) #define OPORTMXMASK_XCKMSK_OFF (0x7 << 0) #define OPORTMXDEBUG(n) (0x420fc + 0x400 * (n)) -#define OPORTMXT0RSTCTR(n) (0x4211c + 0x400 * (n)) -#define OPORTMXT1RSTCTR(n) (0x4213c + 0x400 * (n)) -#define OPORTMXT2RSTCTR(n) (0x4215c + 0x400 * (n)) -#define OPORTMXT3RSTCTR(n) (0x4217c + 0x400 * (n)) -#define OPORTMXT4RSTCTR(n) (0x4219c + 0x400 * (n)) +#define OPORTMXTYVOLPARA1(n, m) (0x42100 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLPARA1_SLOPEU_MASK GENMASK(31, 16) +#define OPORTMXTYVOLPARA2(n, m) (0x42104 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLPARA2_FADE_MASK GENMASK(17, 16) +#define OPORTMXTYVOLPARA2_FADE_NOOP (0x0 << 16) +#define OPORTMXTYVOLPARA2_FADE_FADEOUT (0x1 << 16) +#define OPORTMXTYVOLPARA2_FADE_FADEIN (0x2 << 16) +#define OPORTMXTYVOLPARA2_TARGET_MASK GENMASK(15, 0) +#define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) +#define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT2 (0xa << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT3 (0xb << 8) +#define OPORTMXTYSLOTCTR_SLOTSEL_SLOT4 (0xc << 8) +#define OPORTMXT0SLOTCTR_MUTEOFF_MASK BIT(1) +#define OPORTMXT0SLOTCTR_MUTEOFF_MUTE (0x0 << 1) +#define OPORTMXT0SLOTCTR_MUTEOFF_UNMUTE (0x1 << 1) +#define OPORTMXTYRSTCTR(n, m) (0x4211c + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXT0RSTCTR_RST_MASK BIT(1) +#define OPORTMXT0RSTCTR_RST_OFF (0x0 << 1) +#define OPORTMXT0RSTCTR_RST_ON (0x1 << 1) #define SBF_(frame, shift) (((frame) * 2 - 1) << shift) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 52670126084f..aa89c2f6fa24 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -130,6 +130,10 @@ enum IEC61937_PC { #define AUD_PLLDIV_1_1 2 #define AUD_PLLDIV_2_3 3 +#define AUD_VOL_INIT 0x4000 /* +0dB */ +#define AUD_VOL_MAX 0xffff /* +6dB */ +#define AUD_VOL_FADE_TIME 20 /* 20ms */ + #define AUD_RING_SIZE (128 * 1024) #define AUD_MIN_FRAGMENT 4 @@ -231,6 +235,7 @@ struct uniphier_aio_sub { /* For PCM audio */ struct snd_pcm_substream *substream; struct snd_pcm_hw_params params; + int vol; /* For compress audio */ struct snd_compr_stream *cstream; @@ -323,6 +328,8 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); +int aio_port_get_volume(struct uniphier_aio_sub *sub); +void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol); int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through); int aio_oport_set_stream_type(struct uniphier_aio_sub *sub, enum IEC61937_PC pc); -- 2.17.0

2 1

[alsa-devel] [PATCH] ASoC: nau8824: fix spelling mistake: "semaphone" -> "semaphore"
by Colin King 09 May '18

09 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_warn messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 76502c090654..6bd14453f06e 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -205,11 +205,11 @@ static int nau8824_sema_acquire(struct nau8824 *nau8824, long timeout) if (timeout) { ret = down_timeout(&nau8824->jd_sem, timeout); if (ret < 0) - dev_warn(nau8824->dev, "Acquire semaphone timeout\n"); + dev_warn(nau8824->dev, "Acquire semaphore timeout\n"); } else { ret = down_interruptible(&nau8824->jd_sem); if (ret < 0) - dev_warn(nau8824->dev, "Acquire semaphone fail\n"); + dev_warn(nau8824->dev, "Acquire semaphore fail\n"); } return ret; -- 2.17.0

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[alsa-devel] [PATCH] ASoC: uniphier: evea: use DAPM to change source of line-in
by Katsuhiro Suzuki 09 May '18

09 May '18

This patch replaces mixer switch to DAPM one for changing audio source of line-in. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 55 +++++++++++++++++++++++---------------- 1 file changed, 33 insertions(+), 22 deletions(-) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index 73fd6730095c..f9c10165fbc1 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -54,8 +54,21 @@ struct evea_priv { int switch_hp; }; +static const char * const linsw1_sel1_text[] = { + "LIN1", "LIN2", "LIN3" +}; + +static SOC_ENUM_SINGLE_DECL(linsw1_sel1_enum, + ALINSW1, ALINSW1_SEL1_SHIFT, + linsw1_sel1_text); + +static const struct snd_kcontrol_new linesw1_mux[] = { + SOC_DAPM_ENUM("Line In 1 Source", linsw1_sel1_enum), +}; + static const struct snd_soc_dapm_widget evea_widgets[] = { - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("Line In 1 Mux", SND_SOC_NOPM, 0, 0, linesw1_mux), SND_SOC_DAPM_INPUT("LIN1_LP"), SND_SOC_DAPM_INPUT("LIN1_RP"), SND_SOC_DAPM_INPUT("LIN2_LP"), @@ -63,7 +76,9 @@ static const struct snd_soc_dapm_widget evea_widgets[] = { SND_SOC_DAPM_INPUT("LIN3_LP"), SND_SOC_DAPM_INPUT("LIN3_RP"), - SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC HP", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC LO1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC LO2", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUTPUT("HP1_L"), SND_SOC_DAPM_OUTPUT("HP1_R"), SND_SOC_DAPM_OUTPUT("LO2_L"), @@ -71,17 +86,22 @@ static const struct snd_soc_dapm_widget evea_widgets[] = { }; static const struct snd_soc_dapm_route evea_routes[] = { - { "ADC", NULL, "LIN1_LP" }, - { "ADC", NULL, "LIN1_RP" }, - { "ADC", NULL, "LIN2_LP" }, - { "ADC", NULL, "LIN2_RP" }, - { "ADC", NULL, "LIN3_LP" }, - { "ADC", NULL, "LIN3_RP" }, - - { "HP1_L", NULL, "DAC" }, - { "HP1_R", NULL, "DAC" }, - { "LO2_L", NULL, "DAC" }, - { "LO2_R", NULL, "DAC" }, + { "Line In 1", NULL, "ADC" }, + { "ADC", NULL, "Line In 1 Mux" }, + { "Line In 1 Mux", "LIN1", "LIN1_LP" }, + { "Line In 1 Mux", "LIN1", "LIN1_RP" }, + { "Line In 1 Mux", "LIN2", "LIN2_LP" }, + { "Line In 1 Mux", "LIN2", "LIN2_RP" }, + { "Line In 1 Mux", "LIN3", "LIN3_LP" }, + { "Line In 1 Mux", "LIN3", "LIN3_RP" }, + + { "DAC HP", NULL, "Headphone 1" }, + { "DAC LO1", NULL, "Line Out 1" }, + { "DAC LO2", NULL, "Line Out 2" }, + { "HP1_L", NULL, "DAC HP" }, + { "HP1_R", NULL, "DAC HP" }, + { "LO2_L", NULL, "DAC LO2" }, + { "LO2_R", NULL, "DAC LO2" }, }; static void evea_set_power_state_on(struct evea_priv *evea) @@ -280,16 +300,7 @@ static int evea_set_switch_hp(struct snd_kcontrol *kcontrol, return evea_update_switch_hp(evea); } -static const char * const linsw1_sel1_text[] = { - "LIN1", "LIN2", "LIN3" -}; - -static SOC_ENUM_SINGLE_DECL(linsw1_sel1_enum, - ALINSW1, ALINSW1_SEL1_SHIFT, - linsw1_sel1_text); - static const struct snd_kcontrol_new evea_controls[] = { - SOC_ENUM("Line Capture Source", linsw1_sel1_enum), SOC_SINGLE_BOOL_EXT("Line Capture Switch", 0, evea_get_switch_lin, evea_set_switch_lin), SOC_SINGLE_BOOL_EXT("Line Playback Switch", 0, -- 2.17.0

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[alsa-devel] [PATCH -next] ASoC: fix return value check in mt6351_codec_driver_probe()
by Wei Yongjun 09 May '18

09 May '18

In case of error, the function dev_get_regmap() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/mt6351.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mt6351.c b/sound/soc/codecs/mt6351.c index e739f07..f73dcd7 100644 --- a/sound/soc/codecs/mt6351.c +++ b/sound/soc/codecs/mt6351.c @@ -1472,8 +1472,8 @@ static int mt6351_codec_driver_probe(struct platform_device *pdev) priv->dev = &pdev->dev; priv->regmap = dev_get_regmap(pdev->dev.parent, NULL); - if (IS_ERR(priv->regmap)) - return PTR_ERR(priv->regmap); + if (!priv->regmap) + return -ENODEV; dev_dbg(priv->dev, "%s(), dev name %s\n", __func__, dev_name(&pdev->dev));

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[alsa-devel] [PATCH v4 00/13] ALSA: hda/ca0132: Patch Series for Recon3Di and Sound Blaster Z Support
by Connor McAdams 08 May '18

08 May '18

This patchset adds support for the Sound Blaster Z and the Recon3Di. In order to figure out how to get these cards to work, I made a program called QemuHDADump[1], which uses the trace function of qemu to see interactions with the memory mapped pci BAR space of the card being used in the virtual machine. With this, I obtain the CORB buffer location to get the command verbs, and then dump them each time the buffer rolls over. This program may be useful for fixing other HDA related driver issues where there is no documentation for the device. So far, I have been able to get all features supported on the Sound Blaster Z and the Recon3Di. All output and input effects work, all inputs and outputs work, and just about anything else I can think of. I have also added new controls in order to select the new inputs and outputs, as well as controls to change the effect levels and presets. I have also added the ability to use firmware taken from the Windows drivers of both the Sound Blaster Z and Recon3Di. I am trying to get into contact with Creative to get permission to redistribute these along with the current file included with the Chromebook, but they have not been very responsive. Luckily, the cards work with the Chromebook firmware just fine, although I believe there has to be a reason they have different firmware in Windows. I will not link to the firmwares here, but if you look up my thread on Creative Labs forums, you will find the link to download the firmwares there. I am willing to help get the other non-working cards such as the ZxR and the newer AE-5 working too, but I will need someone willing to run QemuHDADump in a virtual machine in order to get the commands. So, in summary: -This patchset makes the cards work better than they did before (they really didn't work before) -This patchset leaves the original chromebook related stuff alone. Thanks. [1] https://github.com/Conmanx360/QemuHDADump Bugs: ------------------------------------------------------------------------------- Recon3Di: (Reported by Mariusz Ceier) ******************* -Occasionally switching between rear and front mic breaks the input until computer is shutdown or put to sleep. -Surround Sound works, but is inconsistent. Sometimes, just updating the volume fixes it, and sometimes, it requires a restart. Sound Blaster Z: ******************* -none that I'm aware of. Version changes: ------------------------------------------------------------------------------- v1: ******************* -Massive patch formatting failure, please ignore v1. v2: ******************* -Fixed patch formatting failure. v3: ******************* -Fixed mem_base unmap, instead of checking for QUIRK_SBZ on exit, have it check if the area is mapped, and if it is, unmap it. Also make it unmap after all other commands are finished. -Change notification of failure to map mem_base from codec_dbg to codec_warn, and use codec_info to tell the user that their card might have been incorrectly identified as a Sound Blaster Z. -Remove commented out commands in sbz_exit_chip function, only reintroduce them when their functions are defined. v4: ******************* -Split patch into smaller pieces. -Added const to alt_out_presets array. -Fixed command that was commented out and only put it in when it was actually used. Connor McAdams (13): ALSA: hda/ca0132: R3Di and SBZ quirk entires + alt firmware loading ALSA: hda/ca0132: Add pincfg for SBZ + R3Di, add fp hp auto-detect ALSA: hda/ca0132: Add PCI region2 iomap for SBZ ALSA: hda/ca0132: Add extra exit functions for R3Di and SBZ ALSA: hda/ca0132: add extra init functions for r3di + sbz ALSA: hda/ca0132: update core functions for sbz + r3di ALSA: hda/ca0132: add dsp setup related commands for the sbz ALSA: hda/ca0132: Add dsp setup + gpio functions for r3di ALSA: hda/ca0132: add the ability to set src_id on scp commands ALSA: hda/ca0132: add alt_select_in/out for R3Di + SBZ ALSA: hda/ca0132: Add DSP Volume set and New mixers for SBZ + R3Di ALSA: hda/ca0132: add ca0132_alt_set_vipsource ALSA: hda/ca0132: Add new control changes for SBZ + R3Di sound/pci/hda/patch_ca0132.c | 3057 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 2941 insertions(+), 116 deletions(-) -- 2.7.4

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[alsa-devel] Help with pcm3168a device tree, and simple-audio-card/simple-scu-audio-card?
by sdaau 08 May '18

08 May '18

Hi all, I am experimenting with PCM3168A audio codec, connected to a Raspberry Pi 3 B running Raspbian 9. I have set up GPIO4 (BCM4) of the Broadcom BCM2837 SoC of the RPi3B as GPCLK0, and use that as SCKI (System clock input) of the PCM3168A; other than that, I have set up the I2S pins of the BCM2837 in I2S/PCM mode, and use the BCLK/LRCLK clock lines to drive both AD and DA (respective) clock lines; and also the PCM DIN and DOUT lines are connected: the PCM3168A codec has 8 (mono, differential) DACs and 6 (mono, differential) ADCs - but I can only test with 2 DACs for stereo out, and 2 ADCs for stereo in, since there is only one line for DIN and DOUT each on the RPi3B/BCM2837 I2S interface. The PCM3168A is controlled by RPi3B's SPI lines, chip select 1. I have had limited success in running this setup with a device tree overlay based on: * simple-audio-card driver: https://github.com/raspberrypi/linux/blob/rpi-4.14.y/sound/soc/generic/simp… * The already existing PCM3168A Linux kernel driver: https://github.com/raspberrypi/linux/blob/rpi-4.14.y/sound/soc/codecs/pcm31… ... however, I cannot get it to properly work, so I hope I can get some help filling in the blanks. First question: * Q1: In the above setup, the pcm3168a.c is a "codec class driver"; but is simple-card.c a platform class or a machine class driver? Also, would the device tree overlay act as a machine class driver, since it "acts as the glue that describes and binds the other component drivers together" (sound/soc/overview.html)? Here are the relevant device tree fragments of the somewhat working device tree overlay, cased on `simple-card.c`: ``` fragment@3 { target = <&spi0>; __overlay__ { #address-cells = <1>; #size-cells = <0>; status = "okay"; pcm3168a: codec@1 { #sound-dai-cells = <0>; reg = <1>; compatible = "ti,pcm3168a"; spi-max-frequency = <1000000>; clocks = <&clocks 38>; // * BCM2835_CLOCK_GP0 * / clock-names = "scki"; pinctrl-names = "default"; pinctrl-0 = <&gpclk0_gpio4>; VDD1-supply = <&vdd_3v3_reg>; VDD2-supply = <&vdd_3v3_reg>; VCCAD1-supply = <&vdd_5v0_reg>; VCCAD2-supply = <&vdd_5v0_reg>; VCCDA1-supply = <&vdd_5v0_reg>; VCCDA2-supply = <&vdd_5v0_reg>; status = "okay"; }; }; }; fragment@4 { /////// a working one: target = <&sound>; slave_sound_overlay: __overlay__ { compatible = "simple-audio-card"; i2s-controller = <&i2s>; status = "okay"; simple-audio-card,name = "my-sound-card"; simple-audio-card,widgets = "Line", "Line In", "Line", "Line Out"; simple-audio-card,routing = "AIN1L", "Line In", "AIN1R", "Line In", "Line Out", "AOUT1L", "Line Out", "AOUT1R"; simple-audio-card,format = "right_j"; //"i2s"; simple-audio-card,mclk-fs = <128>; simple-audio-card,cpu { sound-dai = <&i2s>; }; sound_master: simple-audio-card,codec { sound-dai = <&pcm3168a>; system-clock-frequency = <6144000>; // both these work: 12288000/48000 = 256; 6144000/48000 = 128 }; }; }; ``` When I load this device tree on Raspbian, the following drivers get loaded: ``` pi@raspberrypi:~ $ lsmod Module Size Used by snd_soc_bcm2835_i2s 16384 2 snd_soc_pcm3168a_spi 16384 1 snd_soc_pcm3168a 24576 1 snd_soc_pcm3168a_spi snd_soc_simple_card 16384 0 snd_soc_simple_card_utils 20480 1 snd_soc_simple_card snd_soc_core 204800 4 snd_soc_pcm3168a,snd_soc_simple_card_utils,snd_soc_bcm2835_i2s,snd_soc_simple_card spidev 16384 0 spi_bcm2835 16384 0 snd_compress 20480 1 snd_soc_core snd_pcm_dmaengine 16384 1 snd_soc_core ... snd_pcm 98304 5 snd_pcm_dmaengine,snd_soc_pcm3168a,snd_soc_bcm2835_i2s,snd_bcm2835,snd_soc_core snd_timer 32768 1 snd_pcm snd 69632 7 snd_compress,snd_timer,snd_bcm2835,snd_soc_core,snd_pcm ... ``` ... with relevant syslog messages: ``` kernel: [ 410.676653] bus: 'platform': add driver spi-bcm2835 ... kernel: [ 410.818605] bus: 'platform': add driver snd-soc-dummy kernel: [ 410.819136] bus: 'platform': add driver soc-audio ... kernel: [ 410.830593] bus: 'platform': add driver asoc-simple-card ... kernel: [ 410.833990] bus: 'spi': add driver pcm3168a kernel: [ 410.840144] bus: 'platform': add driver bcm2835-i2s ... kernel: [ 410.857249] asoc-simple-card soc:sound: pcm3168a-dac <-> 3f203000.i2s mapping ok ... systemd-udevd[921]: Execute 'load' 'platform:snd-soc-dummy' systemd-udevd[921]: No module matches 'platform:snd-soc-dummy' ... systemd-udevd[914]: created db file '/run/udev/data/+sound:card1' for '/devices/platform/soc/soc:sound/sound/card1' ... systemd[1]: sys-devices-platform-soc-soc:sound-sound-card1.device: Changed dead -> plugged ... systemd-udevd[921]: handling device node '/dev/snd/pcmC1D0p', devnum=c116:48, mode=0660, uid=0, gid=29 ... systemd-udevd[917]: handling device node '/dev/snd/controlC1', devnum=c116:32, mode=0660, uid=0, gid=29 ... ``` After this, I have a playback card device - but not a capture one: ``` pi@raspberrypi:~ $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ... card 1: mysoundcard [my-sound-card], device 0: bcm2835-i2s-pcm3168a-dac pcm3168a-dac-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 pi@raspberrypi:~ $ arecord -l **** List of CAPTURE Hardware Devices **** pi@raspberrypi:~ $ ``` Note that at this point, with drivers loaded, I do have "card controls" accessible by amixer - but no "simple controls": ``` pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard info Card hw:CARD=mysoundcard 'mysoundcard'/'my-sound-card' Mixer name : '' Components : '' Controls : 45 amixer: Mixer load hw:CARD=mysoundcard error: Device or resource busy pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard scontrols amixer: Mixer hw:CARD=mysoundcard load error: Device or resource busy pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard controls numid=19,iface=MIXER,name='Master Playback Volume' numid=42,iface=MIXER,name='Master Capture Volume' numid=41,iface=MIXER,name='ADC Overflow Flag Polarity' ... numid=13,iface=MIXER,name='DAC4 Zero Flag' pi@raspberrypi:~ $ amixer -d -D hw:CARD=mysoundcard cset iface=MIXER,name='Master Playback Volume' 81% VERIFY ID: numid=0,iface=MIXER,name='Master Playback Volume' numid=19,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=201,step=0 : values=163 | dBscale-min=-100.50dB,step=0.50dB,mute=1 ``` The accessible controls are, in fact, defined in pcm3168a.c (in ...struct snd_kcontrol_new pcm3168a_snd_controls[]) - so, more questions: * Q2: Is the amixer "error: Device or resource busy" an indication that: *no* driver loaded by the device tree overlay, has a definition for any mixer "simple control"? If so, can I set that up via device tree overlay - or do I need to code it in a driver? If coding a driver, which driver should be patched - pcm3168a.c or simple-card.c? * Q3: In the above `amixer cset` of 'Master Playback Volume', I can see `mute=1` in the output. Does that mean master playback is muted? If so, how do I unmute it with amixer? * Q4: As far as I can see, `alsamixer` deals only with "simple controls", not "card controls" (it exits with `cannot load mixer controls: Device or resource busy` as soon as I choose mysoundcard). Is there any ALSA GUI software that can show and manipulate all card controls of a card? Note that pcm3168a.c has notions of a *stereo* DAC and ADC, so the accessible amixer controls refer to ADC1-ADC3 and DAC1-DAC4. Finally, I test this with: ``` pi@raspberrypi:~ $ speaker-test -D hw:CARD=mysoundcard,DEV=0 -c 2 -t sine -f 440 speaker-test 1.1.3 Playback device is hw:CARD=mysoundcard,DEV=0 Stream parameters are 48000Hz, S16_LE, 2 channels Sine wave rate is 440.0000Hz Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 131072 Period size range from 64 to 65536 Using max buffer size 131072 Periods = 4 was set period_size = 32768 was set buffer_size = 131072 0 - Front Left 1 - Front Right Time per period = 2.781788 ... ``` This is what I observe when speaker-test runs: - I've captured I2S and SPI signals with Saleae Logic: GPCLK0 kicks in first, after about 20 ms, SPI sends on MOSI (and SCL) 14 bytes in about 1 ms - immediately after, BCLK starts; and after some 50 ms, LRCLK and DOUT start in synchrony. So, at least from the RPi3B/BCM2837 side, at first glance all seems fine. (I've also used I2S analyzer in the software, and can confirm there's a sinusoid in the I2S DOUT data, though I cannot decode it 100% yet, since I see some discontinuities in the plots) - I've also measured PCM3168A outputs with "analog" scope - unfortunately I had a massive 50 Hz hum where I've measured, so it's difficult to say much with certainty. But basically, when speaker-test runs, there is a significant HF noise appearing on the DAC outputs (which disappears when speaker-test stops). I cannot tell if this is output from the DAC, but faulty, because of misconfiguration of the PCM3168a chip - or if the DAC is actually mute, and this is just noise leaking from GPCLK0 and similar signals (I'll need to find a quieter lab first to re-do measurements) Here are some more questions: * Q5: How would I go about reading the internal configuration registers of the PCM3168A? I guess I could use printk's in pcm3168a.c - but is there an easy/generic way of reading those from user space? * Q6: Is it possible to set internal configuration registers of the PCM3168A from a device tree overlay - say, if I'd like to set up DAC1 and DAC2 (as per datasheet, which is = stereo DAC1 in pcm3168a.c) to work in differential (or single-ended) mode? Note that the device tree overlay sets "right_j" (right justified) digital data exchange (I2S/PCM) format; because whenever I try to set it to "i2s", running speaker-test fails with "Unable to set hw params for playback: Invalid argument" and these syslog messages: ``` pcm3168a spi0.1: 32-bit frames are supported only for slave mode using right justified pcm3168a spi0.1: ASoC: can't set pcm3168a-dac hw params: -22 ``` * Q6: Where did this "32-bit frames" come from, when using format "i2s"? As far as I see it, speaker-test requested S16_LE stereo, meaning 16-bits need to be clocked in when LRCLK is 'L', and 16-bits when LRCLK is 'R' - can't see a "32-bit frame" here anywhere? On the other hand, there is in the PCM3168A datasheet "Table 11. Audio Data Interface Formats and Sampling Rate, Bit Clock, and System Clock Restrictions", with "(1) BCK = 48 fS, 32 fS is supported only in slave mode; BCK = 32 fS is supported only for 16-bit data length." - Does this mean that, since I've requested 16-bit samples, earlier code decided that BCK should be 32 fS, resulting with this "32-bit frame", which then fails at the point it does ?! Can I explicitly override this frame length in the device tree overlay - if so, where? * Q7: I'm very confused about how the master/slave I2S relationship is established in a device tree with `simple-audio-card`. For instance, I've seen `simple-audio-card,bitclock-master` and `,frame-master`, but also `,bitclock-slave` and `,frame-slave`; and yet, my device tree fragment above, uses none of those - and still, at least some of the system works?! So some relationship must have been established implicitly - my guess is (ignoring the `sound_master` label) that since GPCLK0 is set as SCKI of pcm3168a, bcm2835-i2s becomes master, and pcm3168a (codec) becomes slave. Is that correct - if not, how would one derive the actual implied relationship? I should mention, that one of the biggest mysteries for me here is: * Q8: Why is there no capture interface with this usage of `simple-audio-card` in device tree? In fact, I can see nowhere in the above device tree fragment that there is an explicit setup for playback - and yet, I do get a playback interface (and partially, it works)? I would have thought, that the setup of `simple-audio-card,widgets` and `simple-audio-card,routing` (most of the other .dts examples I've seen, have this as the only reference to input/capture vs output/playback) would have been enough to infer that capture interface is needed too, since I refer to "AIN1L", "AIN1R" along with "AOUT1L", "AOUT1R" (all defined in pcm3168a.c) - but apparently, not? Only other example I've seen is `arch/arm/boot/dts/overlays/pibell-overlay.dts`, which sets up `capture_link: simple-audio-card,dai-link@0` and `playback_link: simple-audio-card,dai-link@1` in `target = <&sound>`; but then again, it uses different chips for input and output - is this approach preferable, also when a single chip de als with both playback and capture? Now, an interesting thing is that, even in my above device tree fragment, if I just change `compatible = "simple-audio-card";` to `compatible = "simple-scu-audio-card";` - drivers will typically load with the device tree, but speaker-test will fail with a segfault at very start. To be sure, I tried writing another device tree fragment for "simple-scu-audio-card": ``` fragment@4 { target = <&sound>; slave_sound_overlay: __overlay__ { compatible = "simple-scu-audio-card"; i2s-controller = <&i2s>; status = "okay"; simple-audio-card,name = "my-sound-card"; simple-audio-card,format = "i2s"; simple-audio-card,convert-rate = <48000>; // simple-scu-card.txt (simple-scu-audio-card); prob no effect with simple-card.c (simple-audio-card) only; //simple-audio-card,prefix = "pcm3168a"; // NOTE: this causes the // ASoC: no source widget found for AOUT1L! // without ,widgets: "ASoC: no sink widget found for Line Out/In" simple-audio-card,widgets = //~ "Line", "Line In", "Line", "Line Out"; simple-audio-card,routing = "Line Out","AOUT1L", "Line Out","AOUT1R"; //, //~ "AIN1L","Line In", //~ "AIN1R","Line In"; //"Playback","DAC1"; //passes without simple-audio-card,widgets - but segfaults sndcpu: simple-audio-card,cpu { sound-dai = <&i2s>; }; sndcodec: simple-audio-card,codec { sound-dai = <&pcm3168a>; system-clock-frequency = <12288000>; }; }; }; ``` The funny thing here is - even if the "Line In" references are commented in ,routing - still, both a playback and a capture interface appear, along with the load of device tree and corresponding drivers! However, speaker-test causes segfault with a stack trace like: ``` kernel: [20819.557733] Unable to handle kernel NULL pointer dereference at virtual address 000000b0 ... kernel: [20819.557997] Hardware name: BCM2835 kernel: [20819.558008] task: 9f31ad00 task.stack: 9f3c8000 kernel: [20819.558148] PC is at is_connected_output_ep+0x1c/0x1fc [snd_soc_core] kernel: [20819.558274] LR is at snd_soc_dapm_dai_get_connected_widgets+0x90/0x1e4 [snd_soc_core] ... [ 1190.725416] [<7f4d1e50>] (is_connected_output_ep [snd_soc_core]) from [<7f4d4cf4>] (snd_soc_dapm_dai_get_connected_widgets+0x90/0x1e4 [snd_soc_core]) ### sound/soc/soc-dapm.c [ 1190.725663] [<7f4d4cf4>] (snd_soc_dapm_dai_get_connected_widgets [snd_soc_core]) from [<7f4db064>] (dpcm_path_get+0x2c/0x8c [snd_soc_core]) ### sound/soc/soc-dapm.c [ 1190.725908] [<7f4db064>] (dpcm_path_get [snd_soc_core]) from [<7f4db55c>] (dpcm_fe_dai_open+0x74/0x688 [snd_soc_core]) ### sound/soc/soc-pcm.c [ 1190.726104] [<7f4db55c>] (dpcm_fe_dai_open [snd_soc_core]) from [<7f101e9c>] (snd_pcm_open_substream+0x6c/0x114 [snd_pcm]) ### sound/soc/soc-pcm.c [ 1190.726238] [<7f101e9c>] (snd_pcm_open_substream [snd_pcm]) from [<7f101ffc>] (snd_pcm_open+0xb8/0x218 [snd_pcm]) [ 1190.726370] [<7f101ffc>] (snd_pcm_open [snd_pcm]) from [<7f102214>] (snd_pcm_playback_open+0x4c/0x6c [snd_pcm]) [ 1190.726498] [<7f102214>] (snd_pcm_playback_open [snd_pcm]) from [<7f0d24f8>] (snd_open+0xa4/0x144 [snd]) [ 1190.726573] [<7f0d24f8>] (snd_open [snd]) from [<80288a54>] (chrdev_open+0xe0/0x1b4) ... ``` * Q9: I've added some extra prints to driver code - but as far as I can see, `simple-audio-card` never ends up calling `is_connected_output_ep`, but `simple-scu-audio-card` does; unless I've made a mistake during debugging, would there a reason for that? * Q10: How can I tell `simple-scu-audio-card` which DACs/ADCs should be used in playback and capture device - given that apparently ,widgets and ,routing alone are not enough to specify state (given that it creates a capture device, even if there are no references to AIN*/Line in there) ?! And finally, in general: * Q11: I generally want to use PCM3168A chip DAC1 and DAC2 as stereo OUT (driver: stereo DAC1), and chip ADC1 and ADC2 as stereo IN (driver: stereo ADC1). Am I better off basing myself on `simple-audio-card`, or on `simple-scu-audio-card`, in the device tree overlay? In fact, is it possible to set up this kind of operation using existing drivers in a device tree overlay - or would I have to add my own driver code (either as patch of existing drivers, or as "my own" driver, possibly started from an existing driver)? * Q12: I'd also want to use (driver: stereo DAC1) and (driver: stereo ADC1) in full-duplex mode. Can this be set up solely via `dai-tdm-slot-num`, `-width`, `-tx-mask` and `-rx-mask` under `simple-audio-card,cpu` in device tree overlay - or would I have to add my own driver code (either patching, or "my own")? Many thanks in advance for any pointers!

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[alsa-devel] [PATCH] ALSA: hda - Enable additional HD-Audio codec information
by Shrikant Dharwadkar 08 May '18

08 May '18

Enable additional HD-Audio codec information to be parsed using alsa-info script to generate codec information with CONFIG_SND_SOC enabled Function group nodes -Add power state controls -Add subordinate node count parameters -Add function group type information -Add function group capabilities -Add unsolicited response controls -Add input converter SDI select information -Add GPIO enable control -Add GPIO direction information -Add GPIO wake enable information -Add GPIO unsolicited control information -Add GPIO sticky mask data -Add GPIO data sets associated with GPIO signals Widget nodes -Add pin widget controls -Add power state controls -Add converter stream and channel controls -Add processing state information -Add processing capabilities -Add volume knob capabilities -Add volume knob control information -Add digital converter control information -Add pin sense control information -Add EAPD/BTL enable controls -Add unsolicited response controls Signed-off-by: Shrikant Dharwadkar <shrikant.dharwadkar(a)intel.com> --- sound/hda/hdac_sysfs.c | 167 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 167 insertions(+) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index fb2aa34..b1acff2 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -37,6 +37,9 @@ static ssize_t type##_show(struct device *dev, \ } \ static DEVICE_ATTR_RO(type) +#define CODEC_PRINT(verb) sprintf(buf, "0x%08x\n", \ + snd_hdac_read_parm(codec, nid, verb)) + CODEC_ATTR(type); CODEC_ATTR(vendor_id); CODEC_ATTR(subsystem_id); @@ -173,6 +176,12 @@ static ssize_t pin_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_PIN_CAP)); } +static ssize_t pin_widget_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_WIDGET_CONTROL); +} + static ssize_t pin_cfg_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -243,6 +252,14 @@ static ssize_t power_caps_show(struct hdac_device *codec, hda_nid_t nid, snd_hdac_read_parm(codec, nid, AC_PAR_POWER_STATE)); } +static ssize_t power_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + if (nid != codec->afg && !(get_wcaps(codec, nid) & AC_WCAP_POWER)) + return 0; + return CODEC_PRINT(AC_VERB_GET_POWER_STATE); +} + static ssize_t gpio_caps_show(struct hdac_device *codec, hda_nid_t nid, struct widget_attribute *attr, char *buf) { @@ -266,27 +283,166 @@ static ssize_t connections_show(struct hdac_device *codec, hda_nid_t nid, return ret; } +static ssize_t converter_stream_channel_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_CONV); +} + +static ssize_t processing_states_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PROC_STATE); +} + +static ssize_t subordinate_node_count_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_NODE_COUNT); +} + +static ssize_t function_group_type_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_FUNCTION_TYPE); +} + +static ssize_t processing_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_PROC_CAP); +} + +static ssize_t volume_knob_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_PAR_VOL_KNB_CAP); +} + +static ssize_t volume_knob_ctl_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_VOLUME_KNOB_CONTROL); +} + +static ssize_t digital_converter_ctl_show(struct hdac_device *codec, + hda_nid_t nid, struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_DIGI_CONVERT_1); +} + +static ssize_t pin_sense_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_PIN_SENSE); +} + +static ssize_t EAPD_BTL_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_EAPD_BTLENABLE); +} + +static ssize_t unsol_caps_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_UNSOLICITED_RESPONSE); +} + +static ssize_t sdi_select_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_SDI_SELECT); +} + +static ssize_t gpio_enable_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_MASK); +} + +static ssize_t gpio_direction_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DIRECTION); +} + +static ssize_t gpio_wake_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_WAKE_MASK); +} + +static ssize_t gpio_unsol_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK); +} + +static ssize_t gpio_sticky_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_STICKY_MASK); +} + +static ssize_t gpio_data_show(struct hdac_device *codec, hda_nid_t nid, + struct widget_attribute *attr, char *buf) +{ + return CODEC_PRINT(AC_VERB_GET_GPIO_DATA); +} + static WIDGET_ATTR_RO(caps); static WIDGET_ATTR_RO(pin_caps); static WIDGET_ATTR_RO(pin_cfg); +static WIDGET_ATTR_RO(pin_widget_ctl); static WIDGET_ATTR_RO(pcm_caps); static WIDGET_ATTR_RO(pcm_formats); static WIDGET_ATTR_RO(amp_in_caps); static WIDGET_ATTR_RO(amp_out_caps); static WIDGET_ATTR_RO(power_caps); +static WIDGET_ATTR_RO(power_states); static WIDGET_ATTR_RO(gpio_caps); static WIDGET_ATTR_RO(connections); +static WIDGET_ATTR_RO(converter_stream_channel); +static WIDGET_ATTR_RO(processing_states); +static WIDGET_ATTR_RO(subordinate_node_count); +static WIDGET_ATTR_RO(function_group_type); +static WIDGET_ATTR_RO(processing_caps); +static WIDGET_ATTR_RO(volume_knob_caps); +static WIDGET_ATTR_RO(volume_knob_ctl); +static WIDGET_ATTR_RO(digital_converter_ctl); +static WIDGET_ATTR_RO(pin_sense); +static WIDGET_ATTR_RO(EAPD_BTL); +static WIDGET_ATTR_RO(unsol_caps); +static WIDGET_ATTR_RO(sdi_select); +static WIDGET_ATTR_RO(gpio_enable); +static WIDGET_ATTR_RO(gpio_direction); +static WIDGET_ATTR_RO(gpio_wake); +static WIDGET_ATTR_RO(gpio_unsol); +static WIDGET_ATTR_RO(gpio_sticky); +static WIDGET_ATTR_RO(gpio_data); static struct attribute *widget_node_attrs[] = { &wid_attr_caps.attr, &wid_attr_pin_caps.attr, + &wid_attr_pin_widget_ctl.attr, &wid_attr_pin_cfg.attr, &wid_attr_pcm_caps.attr, &wid_attr_pcm_formats.attr, &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_connections.attr, + &wid_attr_converter_stream_channel.attr, + &wid_attr_processing_states.attr, + &wid_attr_processing_caps.attr, + &wid_attr_volume_knob_caps.attr, + &wid_attr_volume_knob_ctl.attr, + &wid_attr_digital_converter_ctl.attr, + &wid_attr_pin_sense.attr, + &wid_attr_EAPD_BTL.attr, + &wid_attr_unsol_caps.attr, NULL, }; @@ -296,7 +452,18 @@ static struct attribute *widget_afg_attrs[] = { &wid_attr_amp_in_caps.attr, &wid_attr_amp_out_caps.attr, &wid_attr_power_caps.attr, + &wid_attr_power_states.attr, &wid_attr_gpio_caps.attr, + &wid_attr_subordinate_node_count.attr, + &wid_attr_function_group_type.attr, + &wid_attr_unsol_caps.attr, + &wid_attr_sdi_select.attr, + &wid_attr_gpio_enable.attr, + &wid_attr_gpio_direction.attr, + &wid_attr_gpio_wake.attr, + &wid_attr_gpio_unsol.attr, + &wid_attr_gpio_sticky.attr, + &wid_attr_gpio_data.attr, NULL, }; -- 2.7.4

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[alsa-devel] [PATCH] ALSA: usb: mixer: volume quirk for CM102-A+/102S+
by Federico Cuello 07 May '18

07 May '18

Currently it's not possible to set volume lower than 26% (it just mutes). Also fixes this warning: Warning! Unlikely big volume range (=9472), cval->res is probably wrong. [13] FU [PCM Playback Volume] ch = 2, val = -9473/-1/1 , and volume works fine for full range. --- sound/usb/mixer.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 344d7b069d59..bb5ab7a7dfa5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -967,6 +967,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; + case USB_ID(0x0d8c, 0x0103): + if (!strcmp(kctl->id.name, "PCM Playback Volume")) { + usb_audio_info(chip, + "set volume quirk for CM102-A+/102S+\n"); + cval->min = -256; + } + break; + case USB_ID(0x0471, 0x0101): case USB_ID(0x0471, 0x0104): case USB_ID(0x0471, 0x0105): -- 2.17.0

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[alsa-devel] [PATCH v3 0/9] ALSA: hda/ca0132: Patch Series for Recon3Di and Sound Blaster Z Support
by Connor McAdams 07 May '18

07 May '18

This patchset adds support for the Sound Blaster Z and the Recon3Di. In order to figure out how to get these cards to work, I made a program called QemuHDADump[1], which uses the trace function of qemu to see interactions with the memory mapped pci BAR space of the card being used in the virtual machine. With this, I obtain the CORB buffer location to get the command verbs, and then dump them each time the buffer rolls over. This program may be useful for fixing other HDA related driver issues where there is no documentation for the device. So far, I have been able to get all features supported on the Sound Blaster Z and the Recon3Di. All output and input effects work, all inputs and outputs work, and just about anything else I can think of. I have also added new controls in order to select the new inputs and outputs, as well as controls to change the effect levels and presets. I have also added the ability to use firmware taken from the Windows drivers of both the Sound Blaster Z and Recon3Di. I am trying to get into contact with Creative to get permission to redistribute these along with the current file included with the Chromebook, but they have not been very responsive. Luckily, the cards work with the Chromebook firmware just fine, although I believe there has to be a reason they have different firmware in Windows. I will not link to the firmwares here, but if you look up my thread on Creative Labs forums, you will find the link to download the firmwares there. I am willing to help get the other non-working cards such as the ZxR and the newer AE-5 working too, but I will need someone willing to run QemuHDADump in a virtual machine in order to get the commands. So, in summary: -This patchset makes the cards work better than they did before (they really didn't work before) -This patchset leaves the original chromebook related stuff alone. Thanks. [1] https://github.com/Conmanx360/QemuHDADump Bugs: ------------------------------------------------------------------------------- Recon3Di: (Reported by Mariusz Ceier) ******************* -Occasionally switching between rear and front mic breaks the input until computer is shutdown or put to sleep. -Surround Sound works, but is inconsistent. Sometimes, just updating the volume fixes it, and sometimes, it requires a restart. Sound Blaster Z: ******************* -none that I'm aware of. Version changes: ------------------------------------------------------------------------------- v1: ******************* -Massive patch formatting failure, please ignore v1. v2: ******************* -Fixed patch formatting failure. v3: ******************* -Fixed mem_base unmap, instead of checking for QUIRK_SBZ on exit, have it check if the area is mapped, and if it is, unmap it. Also make it unmap after all other commands are finished. -Change notification of failure to map mem_base from codec_dbg to codec_warn, and use codec_info to tell the user that their card might have been incorrectly identified as a Sound Blaster Z. -Remove commented out commands in sbz_exit_chip function, only reintroduce them when their functions are defined. Connor McAdams (9): ALSA: hda/ca0132: R3Di and SBZ quirk entires + alt firmware loading ALSA: hda/ca0132: Add pincfg for SBZ + R3Di, add fp hp auto-detect ALSA: hda/ca0132: Add PCI region2 iomap for SBZ ALSA: hda/ca0132: Add extra exit functions for R3Di and SBZ ALSA: hda/ca0132: add/change helper functions for R3Di and SBZ ALSA: hda/ca0132: add alt_select_in/out for R3Di + SBZ ALSA: hda/ca0132: Add DSP Volume set and New mixers for SBZ + R3Di ALSA: hda/ca0132: add ca0132_alt_set_vipsource ALSA: hda/ca0132: Add new control changes for SBZ + R3Di sound/pci/hda/patch_ca0132.c | 3055 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 2941 insertions(+), 114 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v3 00/20] Allow compile-testing NO_DMA (drivers)
by Geert Uytterhoeven 07 May '18

07 May '18

Hi all, As of v4.17-rc1, patch series "[PATCH v2 0/5] Allow compile-testing NO_DMA (core)" (https://lkml.org/lkml/2018/3/16/435) has been included upstream, and drivers using the DMA API can be compile-tested on platforms selecting NO_DMA. This follow-up patch series removes dependencies on HAS_DMA for symbols that already have platform dependencies implying HAS_DMA, which increases compile-coverage. Please apply to your tree if appropriate. Changes compared to v2: - Add Acked-by, - Rebased to v4.17-rc1, dropping applied patch for scsi/hisi_sas, - Handle new VIDEO_RENESAS_CEU symbol, - Drop obsolete note about FSL_FMAN. Changes compared to v1: - Add Reviewed-by, Acked-by, - Drop dependency of SND_SOC_LPASS_IPQ806X on HAS_DMA, - Drop dependency of VIDEOBUF{,2}_DMA_{CONTIG,SG} on HAS_DMA, - Drop new dependencies of VIDEO_IPU3_CIO2, DVB_C8SECTPFE, and MTD_NAND_MARVELL on HAS_DMA, - Split in per-subsystem patches, - Split-off the core part in a separate series. This series is against v4.17-rc1. It can also be found at https://git.kernel.org/pub/scm/linux/kernel/git/geert/linux-m68k.git/log/?h… It has been compile-tested with allmodconfig and allyesconfig for m68k/sun3, and has received attention from the kbuild test robot. Thanks for applying! Geert Uytterhoeven (20): ASoC: Remove depends on HAS_DMA in case of platform dependency ata: Remove depends on HAS_DMA in case of platform dependency crypto: Remove depends on HAS_DMA in case of platform dependency fbdev: Remove depends on HAS_DMA in case of platform dependency firewire: Remove depends on HAS_DMA in case of platform dependency fpga: Remove depends on HAS_DMA in case of platform dependency i2c: Remove depends on HAS_DMA in case of platform dependency iio: adc: Remove depends on HAS_DMA in case of platform dependency iommu: Remove depends on HAS_DMA in case of platform dependency lightnvm: Remove depends on HAS_DMA in case of platform dependency mailbox: Remove depends on HAS_DMA in case of platform dependency media: Remove depends on HAS_DMA in case of platform dependency mmc: Remove depends on HAS_DMA in case of platform dependency mtd: Remove depends on HAS_DMA in case of platform dependency net: Remove depends on HAS_DMA in case of platform dependency remoteproc: Remove depends on HAS_DMA in case of platform dependency serial: Remove depends on HAS_DMA in case of platform dependency spi: Remove depends on HAS_DMA in case of platform dependency staging: vc04_services: Remove depends on HAS_DMA in case of platform dependency usb: Remove depends on HAS_DMA in case of platform dependency drivers/ata/Kconfig | 2 -- drivers/crypto/Kconfig | 14 +++------ drivers/firewire/Kconfig | 1 - drivers/fpga/Kconfig | 1 - drivers/i2c/busses/Kconfig | 3 -- drivers/iio/adc/Kconfig | 2 -- drivers/iommu/Kconfig | 5 ++- drivers/lightnvm/Kconfig | 2 +- drivers/mailbox/Kconfig | 2 -- drivers/media/common/videobuf2/Kconfig | 2 -- drivers/media/pci/dt3155/Kconfig | 1 - drivers/media/pci/intel/ipu3/Kconfig | 1 - drivers/media/pci/solo6x10/Kconfig | 1 - drivers/media/pci/sta2x11/Kconfig | 1 - drivers/media/pci/tw5864/Kconfig | 1 - drivers/media/pci/tw686x/Kconfig | 1 - drivers/media/platform/Kconfig | 42 +++++++++---------------- drivers/media/platform/am437x/Kconfig | 2 +- drivers/media/platform/atmel/Kconfig | 4 +-- drivers/media/platform/davinci/Kconfig | 6 ---- drivers/media/platform/marvell-ccic/Kconfig | 3 +- drivers/media/platform/rcar-vin/Kconfig | 2 +- drivers/media/platform/soc_camera/Kconfig | 3 +- drivers/media/platform/sti/c8sectpfe/Kconfig | 2 +- drivers/media/v4l2-core/Kconfig | 2 -- drivers/mmc/host/Kconfig | 10 ++---- drivers/mtd/nand/raw/Kconfig | 8 ++--- drivers/mtd/spi-nor/Kconfig | 2 +- drivers/net/ethernet/amd/Kconfig | 2 +- drivers/net/ethernet/apm/xgene-v2/Kconfig | 1 - drivers/net/ethernet/apm/xgene/Kconfig | 1 - drivers/net/ethernet/arc/Kconfig | 6 ++-- drivers/net/ethernet/broadcom/Kconfig | 2 -- drivers/net/ethernet/calxeda/Kconfig | 2 +- drivers/net/ethernet/hisilicon/Kconfig | 2 +- drivers/net/ethernet/marvell/Kconfig | 8 ++--- drivers/net/ethernet/mellanox/mlxsw/Kconfig | 2 +- drivers/net/ethernet/renesas/Kconfig | 2 -- drivers/net/wireless/broadcom/brcm80211/Kconfig | 1 - drivers/net/wireless/quantenna/qtnfmac/Kconfig | 2 +- drivers/remoteproc/Kconfig | 1 - drivers/spi/Kconfig | 12 ++----- drivers/staging/media/davinci_vpfe/Kconfig | 1 - drivers/staging/media/omap4iss/Kconfig | 1 - drivers/staging/vc04_services/Kconfig | 1 - drivers/tty/serial/Kconfig | 4 --- drivers/usb/gadget/udc/Kconfig | 4 +-- drivers/usb/mtu3/Kconfig | 2 +- drivers/video/fbdev/Kconfig | 3 +- sound/soc/bcm/Kconfig | 3 +- sound/soc/kirkwood/Kconfig | 1 - sound/soc/pxa/Kconfig | 1 - sound/soc/qcom/Kconfig | 7 ++--- 53 files changed, 56 insertions(+), 142 deletions(-) -- 2.7.4 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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