April 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm_adsp: Remove uneeded semicolon
by Charles Keepax 18 Apr '14

18 Apr '14

Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d9686dc..ff15eec 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1625,7 +1625,7 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, break; default: break; - }; + } return 0; } -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: document that the regulators are mandatory
by Alexandre Belloni 18 Apr '14

18 Apr '14

The driver will not probe if regulators are not provided. Move the regulator properties to the required section. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt index 74c66dee3e14..eff12be5e789 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -13,6 +13,9 @@ Required properties: "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) - reg - <int> - I2C slave address +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt Optional properties: @@ -24,9 +27,6 @@ Optional properties: 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD If this node is not mentioned or if the value is unknown, then micbias is set to 2.0V. -- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, - DVDD-supply : power supplies for the device as covered in - Documentation/devicetree/bindings/regulator/regulator.txt CODEC output pins: * HPL -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: fsl_spdif: Fix wrong OFFSET of STC_SYSCLK_DIV
by Nicolin Chen 18 Apr '14

18 Apr '14

It should use STC_SYSCLK_DIV_OFFSET. Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index b126679..605a10b 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -144,8 +144,8 @@ enum spdif_gainsel { /* SPDIF Clock register */ #define STC_SYSCLK_DIV_OFFSET 11 -#define STC_SYSCLK_DIV_MASK (0x1ff << STC_TXCLK_SRC_OFFSET) -#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK) +#define STC_SYSCLK_DIV_MASK (0x1ff << STC_SYSCLK_DIV_OFFSET) +#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_SYSCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK) #define STC_TXCLK_SRC_OFFSET 8 #define STC_TXCLK_SRC_MASK (0x7 << STC_TXCLK_SRC_OFFSET) #define STC_TXCLK_SRC_SET(x) ((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK) -- 1.8.4

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Re: [alsa-devel] [PATCH] ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols
by Austin, Brian 18 Apr '14

18 Apr '14

On Apr 18, 2014, at 4:23 AM, Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> wrote: > Dear Brian Austin, > > On Thu, 17 Apr 2014 10:08:23 -0500, Brian Austin wrote: > >> can you try this quick check please? >> >> static const struct snd_kcontrol_new cs42l51_snd_controls[] = { >> SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", >> CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, >> - 0, 0x19, 0x7F, adc_pcm_tlv), >> + 0, 0x19, 0x80, adc_pcm_tlv), >> SOC_DOUBLE_R("PCM Playback Switch", >> CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), >> SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", >> CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, >> - 0, 0x34, 0xE4, aout_tlv), >> + 0, 0x34, 0xE5, aout_tlv), >> SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", >> CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, >> - 0, 0x19, 0x7F, adc_pcm_tlv), >> + 0, 0x19, 0x80, adc_pcm_tlv), >> SOC_DOUBLE_R("ADC Mixer Switch", >> CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), >> SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, >> 0), > > Unfortunately, it still doesn't work: at boot time the PCM and ADC > mixer values are 100 which looks OK, but the Analog value is 170, which > doesn't look ok. And inspecting the values set at the register level > for the PCM control seems to reveal that things are still not working > correctly (read on below). > > But this time around, I've collected a lot of debugging informations, > which hopefully will help understand the problem. My understanding is > that: > > SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", > CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, > 0, 0x34, 0xE5, aout_tlv), > > cannot work, because it assumes that the range -102.0 dB to 12.0 dB > described in the datasheet is represented by using register values from > 0x34 to 0xE5. Which misses the fact that when going from -102.0 dB to 0 > dB, the register value goes from 0x34 to 0, and then when going from 0 > to 12.0 dB, the register value should go from 0 to 0xE5. > > Ok, Thanks for the output. Something is for sure broken, but I could have sworn I checked this. I’ll hook up my L51 board today and see what the deal is. Thanks, Brian

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[alsa-devel] [PATCH 1/6] sound: lx_core: Remove unused defines
by Maxime Ripard 18 Apr '14

18 Apr '14

Commit f9367f3fbe3c ("ALSA: lx6464es: Remove unused function in pci/lx6464es/lx_core.c") removed the lx_dsp_es_check_pipeline function that was the only user of these defines. Since they're useless now, simply remove them. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/pci/lx6464es/lx_core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 2d8e95e9fbe5..6a1d90a308ea 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -429,11 +429,6 @@ int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data) return ret; } -#define CSES_TIMEOUT 100 /* microseconds */ -#define CSES_CE 0x0001 -#define CSES_BROADCAST 0x0002 -#define CSES_UPDATE_LDSV 0x0004 - #define PIPE_INFO_TO_CMD(capture, pipe) \ ((u32)((u32)(pipe) | ((capture) ? ID_IS_CAPTURE : 0L)) << ID_OFFSET) -- 1.9.1

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[alsa-devel] [PATCHv3 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 18 Apr '14

18 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- No changes since v1. --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, @@ -3377,6 +3378,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); MODULE_ALIAS("snd-hda-codec-id:10de001b"); MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0028"); MODULE_ALIAS("snd-hda-codec-id:10de0040"); MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); -- 1.8.1.3.605.g02339dd

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[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 18 Apr '14

18 Apr '14

Does the output help?

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[alsa-devel] [PATCH] ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols
by Brian Austin 17 Apr '14

17 Apr '14

The shift values for the ADC,PCM, and Analog kcontrols were wrong causing wrong values for the SOC_DOUBLE_R_SX_TLV macros Fixed the TLV for aout_tlv to show -102dB correctly Reported-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l51.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 3eab460..65fc37c 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -106,9 +106,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); -/* This is a lie. after -102 db, it stays at -102 */ -/* maybe a range would be better */ -static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); static const char *chan_mix[] = { @@ -123,7 +122,7 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", @@ -131,7 +130,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), -- 1.7.9.5

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[alsa-devel] [PATCH] sound: lx_core: Few misc cleanups
by Maxime Ripard 17 Apr '14

17 Apr '14

This file had a few comments in french that are now in english, and took the opportunity to cleanup a bunch of #if 0 .. #endif and commented out code Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/pci/lx6464es/lx_core.c | 70 +++++++++++++++++--------------------------- 1 file changed, 27 insertions(+), 43 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 626ecad4dae7..002049770ff7 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -22,8 +22,7 @@ * */ -/* #define RMH_DEBUG 1 */ - +#include <linux/bitops.h> #include <linux/module.h> #include <linux/pci.h> #include <linux/delay.h> @@ -496,15 +495,17 @@ int lx_dsp_es_check_pipeline(struct lx6464es *chip) int i; for (i = 0; i != CSES_TIMEOUT; ++i) { - /* - * le bit CSES_UPDATE_LDSV est Ã 1 dÃ©s que le macprog - * est pret. il re-passe Ã 0 lorsque le premier read a - * Ã©tÃ© fait. pour l'instant on retire le test car ce bit - * passe a 1 environ 200 Ã 400 ms aprÃ©s que le registre - * confES Ã Ã©tÃ© Ã©crit (kick du xilinx ES). + /* + * The bit CSES_UPDATE_LDSV is set to 1 as soon as the + * macprog is ready. He goes back to 0 whenever the + * first read has been issued. + * + * For now, we don't bother testing this bit because + * it's set to 1 between 200 to 400ms after the confES + * register has been written to (xilinx ES kick). * - * On ne teste que le bit CE. - * */ + * We only test the CE bit. + */ u32 cses = lx_dsp_reg_read(chip, eReg_CSES); @@ -602,8 +603,7 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, /* free */ *r_needed += 1; } - -#if 0 +#if DEBUG snd_printdd(LXP "CMD_08_ASK_BUFFERS: needed %d, freed %d\n", *r_needed, *r_freed); for (i = 0; i < MAX_STREAM_BUFFER; ++i) { @@ -1049,9 +1049,9 @@ int lx_level_peaks(struct lx6464es *chip, int is_capture, int channels, /* interrupt handling */ #define PCX_IRQ_NONE 0 -#define IRQCS_ACTIVE_PCIDB 0x00002000L /* Bit nÃÂ¸ 13 */ -#define IRQCS_ENABLE_PCIIRQ 0x00000100L /* Bit nÃÂ¸ 08 */ -#define IRQCS_ENABLE_PCIDB 0x00000200L /* Bit nÃÂ¸ 09 */ +#define IRQCS_ACTIVE_PCIDB BIT(13) +#define IRQCS_ENABLE_PCIIRQ BIT(8) +#define IRQCS_ENABLE_PCIDB BIT(9) static u32 lx_interrupt_test_ack(struct lx6464es *chip) { @@ -1108,25 +1108,21 @@ static int lx_interrupt_handle_async_events(struct lx6464es *chip, u32 irqsrc, int err; u32 stat[9]; /* answer from CMD_04_GET_EVENT */ - /* On peut optimiser pour ne pas lire les evenements vides - * les mots de rÃÂ©ponse sont dans l'ordre suivant : - * Stat[0] mot de status gÃÂ©nÃÂ©ral - * Stat[1] fin de buffer OUT pF - * Stat[2] fin de buffer OUT pf - * Stat[3] fin de buffer IN pF - * Stat[4] fin de buffer IN pf - * Stat[5] underrun poid fort - * Stat[6] underrun poid faible - * Stat[7] overrun poid fort - * Stat[8] overrun poid faible + /* We can optimize this to not read dumb events. + * Answer words are in the following order: + * Stat[0] general status + * Stat[1] end of buffer OUT pF + * Stat[2] end of buffer OUT pf + * Stat[3] end of buffer IN pF + * Stat[4] end of buffer IN pf + * Stat[5] MSB underrun + * Stat[6] LSB underrun + * Stat[7] MSB overrun + * Stat[8] LSB overrun * */ u64 orun_mask; u64 urun_mask; -#if 0 - int has_underrun = (irqsrc & MASK_SYS_STATUS_URUN) ? 1 : 0; - int has_overrun = (irqsrc & MASK_SYS_STATUS_ORUN) ? 1 : 0; -#endif int eb_pending_out = (irqsrc & MASK_SYS_STATUS_EOBO) ? 1 : 0; int eb_pending_in = (irqsrc & MASK_SYS_STATUS_EOBI) ? 1 : 0; @@ -1272,7 +1268,7 @@ irqreturn_t lx_interrupt(int irq, void *dev_id) if (irqsrc & MASK_SYS_STATUS_CMD_DONE) goto exit; -#if 0 +#if DEBUG if (irqsrc & MASK_SYS_STATUS_EOBI) snd_printdd(LXP "interrupt: EOBI\n"); @@ -1310,18 +1306,6 @@ irqreturn_t lx_interrupt(int irq, void *dev_id) "error during audio transfer\n"); } - if (async_escmd) { -#if 0 - /* backdoor for ethersound commands - * - * for now, we do not need this - * - * */ - - snd_printdd("lx6464es: interrupt requests escmd handling\n"); -#endif - } - exit: spin_unlock(&chip->lock); return IRQ_HANDLED; /* this device caused the interrupt */ -- 1.9.1

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Re: [alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by Clemens Ladisch 17 Apr '14

17 Apr '14

mick wrote: > I tried the patch but it does not help. Still crackles. Do you get changing values, and any out-of-range values, when you execute this while it's playing? (while sleep 0.123; do grep 'Momentary freq' /proc/asound/card*/stream0; done) | uniq Regards, Clemens

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