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[alsa-devel] [PATCH] ASoC: SOF: Makefile - fix the top-level kernel module names (add snd- prefix)
by Jaroslav Kysela 25 Jul '19

25 Jul '19

Use the proper module name. The objs assignments are already there. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org>0 --- sound/soc/sof/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 8f14c9d2950b..d2f7a1bc09a1 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -11,8 +11,8 @@ obj-$(CONFIG_SND_SOC_SOF) += snd-sof.o obj-$(CONFIG_SND_SOC_SOF_NOCODEC) += snd-sof-nocodec.o -obj-$(CONFIG_SND_SOC_SOF_ACPI) += sof-acpi-dev.o -obj-$(CONFIG_SND_SOC_SOF_PCI) += sof-pci-dev.o +obj-$(CONFIG_SND_SOC_SOF_ACPI) += snd-sof-acpi-dev.o +obj-$(CONFIG_SND_SOC_SOF_PCI) += snd-sof-pci-dev.o obj-$(CONFIG_SND_SOC_SOF_INTEL_TOPLEVEL) += intel/ obj-$(CONFIG_SND_SOC_SOF_XTENSA) += xtensa/ -- 2.20.1

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[alsa-devel] [PATCH 0/6] ASoC: improve codec to codec link support
by Jerome Brunet 25 Jul '19

25 Jul '19

As explained in this previous series [0], on Amlogic, we are using codec to codec links to deal with the glue which is between the i2s backends and the synopsys hdmi controller. This worked well until I tried to .get_eld() support in the dw-hdmi-i2s driver. Doing so adds channel mapping controls to the codec. This shown several problem 1) .pcm_new() is not called on codec to codec links. struct snd_soc_pcm_runtime do not even have a valid .pcm 2) struct snd_pcm_substream and struct snd_pcm_runtime are ephemeral This is a problem if a control needs to access them The goal of this patchset is to resolve the above issues and improve the codec to codec link support enough to correctly handle the hdmi-codec The support of these codec to codec links could probably be improved in the future to behave like any other link and use soc_pcm_open(), soc_pcm_hw_params(), etc... The challenge lies in the dapm mutex. The soc_pcm call dapm function locking this mutex but the dapm mutex is already held in snd_soc_dai_link_event() [0]: https://lkml.kernel.org/r/20190515131858.32130-1-jbrunet@baylibre.com Jerome Brunet (6): ASoC: codec2codec: run callbacks in order ASoC: codec2codec: name link using stream direction ASoC: codec2codec: deal with params when necessary ASoC: create pcm for codec2codec links as well ASoC: codec2codec: remove ephemeral variables ASoC: codec2codec: fill some of the runtime stream parameters sound/soc/soc-core.c | 42 +++------- sound/soc/soc-dapm.c | 186 +++++++++++++++++++++++++++---------------- sound/soc/soc-pcm.c | 35 +++++++- 3 files changed, 162 insertions(+), 101 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH] ASoC: pcm3168a: Allow all channels in case of parallel DIN/DOUT setup
by Peter Ujfalusi 25 Jul '19

25 Jul '19

If multi DIN/DOUT mode is selected (tdm_slots == 2) then configure the channel constraint to allow all channels. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 5d59ce254821..e84a1509fe65 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -599,6 +599,10 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, sample_min, 32); + /* Allow all channels in multi DIN/DOUT mode */ + if (pcm3168a->tdm_slots == 2) + channel_max = channel_maxs[tx]; + snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, channel_max); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: ti: davinci-mcasp: Add support for RIGHT_J format
by Peter Ujfalusi 25 Jul '19

25 Jul '19

The formater unit's rotation needs to be programmed differently for right aligned bus format to have the data moved to the correct place. Take the opportunity and simplify the formater unit setup code. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 43 ++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 24 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index ac59b509ead5..09ba3d3c70a0 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -465,6 +465,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* FS need to be inverted */ inv_fs = true; break; + case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_LEFT_J: /* configure a full-word SYNC pulse (LRCLK) */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); @@ -758,34 +759,28 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, int sample_width) { u32 fmt; - u32 tx_rotate = (sample_width / 4) & 0x7; + u32 tx_rotate, rx_rotate, slot_width; u32 mask = (1ULL << sample_width) - 1; - u32 slot_width = sample_width; - - /* - * For captured data we should not rotate, inversion and masking is - * enoguh to get the data to the right position: - * Format data from bus after reverse (XRBUF) - * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| - * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| - * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| - * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| - */ - u32 rx_rotate = 0; + if (mcasp->slot_width) + slot_width = mcasp->slot_width; + else + slot_width = sample_width; /* - * Setting the tdm slot width either with set_clkdiv() or - * set_tdm_slot() allows us to for example send 32 bits per - * channel to the codec, while only 16 of them carry audio - * payload. + * TX rotation: + * right aligned formats: rotate w/ slot_width + * left aligned formats: rotate w/ sample_width + * + * RX rotation: + * right aligned formats: no rotation needed + * left aligned formats: rotate w/ (slot_width - sample_width) */ - if (mcasp->slot_width) { - /* - * When we have more bclk then it is needed for the - * data, we need to use the rotation to move the - * received samples to have correct alignment. - */ - slot_width = mcasp->slot_width; + if ((mcasp->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_RIGHT_J) { + tx_rotate = (slot_width / 4) & 0x7; + rx_rotate = 0; + } else { + tx_rotate = (sample_width / 4) & 0x7; rx_rotate = (slot_width - sample_width) / 4; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode
by Peter Ujfalusi 25 Jul '19

25 Jul '19

When running McASP as master capture alone will not record any audio unless a parallel playback stream is running. As soon as the playback stops the captured data is going to be silent again. In McASP master mode we need to set the PDIR for the clock pins and fix the mcasp_set_axr_pdir() to skip the bits in the PDIR registers above AMUTE. This went unnoticed as most of the boards uses McASP as slave and neither of these issues are visible (audible) in those setups. Fixes: 2302be4126f52 ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 09ba3d3c70a0..6a87455d7c42 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -195,7 +195,7 @@ static inline void mcasp_set_axr_pdir(struct davinci_mcasp *mcasp, bool enable) { u32 bit; - for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AFSR) { + for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AMUTE) { if (enable) mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit)); else @@ -223,6 +223,7 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) if (mcasp_is_synchronous(mcasp)) { mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_clk_pdir(mcasp, true); } /* Activate serializer(s) */ -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: ti: davinci-mcasp: Improve serializer handling in multi AXR setups
by Peter Ujfalusi 25 Jul '19

25 Jul '19

When multiple serializers are used we need to track the number of serializers used by the other stream direction to avoid killing data lines when the first stream used more serializers than the second would need. We are still protected against the case when the second stream uses more serializers which had affected the running stream as well. To take advantage of the improved serializer logic we need to modify the channel constraints rule as well to allow the use of multiple serializers for the second stream as additional ones will not affect the FS/BCLK on the bus. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 92 +++++++++++++++++++++--------------- 1 file changed, 53 insertions(+), 39 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 6a87455d7c42..7c2081b63a60 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -109,6 +109,7 @@ struct davinci_mcasp { /* Used for comstraint setting on the second stream */ u32 channels; + u8 active_serializers[2]; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; @@ -814,6 +815,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; + u8 max_rx_serializers, max_tx_serializers; int active_serializers, numevt; u32 reg; /* Default configuration */ @@ -823,22 +825,28 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, if (stream == SNDRV_PCM_STREAM_PLAYBACK) { mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + max_tx_serializers = max_active_serializers; + max_rx_serializers = + mcasp->active_serializers[SNDRV_PCM_STREAM_CAPTURE]; } else { mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); mcasp_clr_bits(mcasp, DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS); + max_tx_serializers = + mcasp->active_serializers[SNDRV_PCM_STREAM_PLAYBACK]; + max_rx_serializers = max_active_serializers; } for (i = 0; i < mcasp->num_serializer; i++) { mcasp_set_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), mcasp->serial_dir[i]); if (mcasp->serial_dir[i] == TX_MODE && - tx_ser < max_active_serializers) { + tx_ser < max_tx_serializers) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), mcasp->dismod, DISMOD_MASK); set_bit(PIN_BIT_AXR(i), &mcasp->pdir); tx_ser++; } else if (mcasp->serial_dir[i] == RX_MODE && - rx_ser < max_active_serializers) { + rx_ser < max_rx_serializers) { clear_bit(PIN_BIT_AXR(i), &mcasp->pdir); rx_ser++; } else { @@ -885,7 +893,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, } else { dma_data->maxburst = 0; } - return 0; + + goto out; } if (period_words % active_serializers) { @@ -915,6 +924,9 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, numevt = 0; dma_data->maxburst = numevt; +out: + mcasp->active_serializers[stream] = active_serializers; + return 0; } @@ -1154,6 +1166,37 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + word_length = 8; + break; + + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + word_length = 16; + break; + + case SNDRV_PCM_FORMAT_U24_3LE: + case SNDRV_PCM_FORMAT_S24_3LE: + word_length = 24; + break; + + case SNDRV_PCM_FORMAT_U24_LE: + case SNDRV_PCM_FORMAT_S24_LE: + word_length = 24; + break; + + case SNDRV_PCM_FORMAT_U32_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_length = 32; + break; + + default: + printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); + return -EINVAL; + } + ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); if (ret) return ret; @@ -1188,37 +1231,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: - word_length = 8; - break; - - case SNDRV_PCM_FORMAT_U16_LE: - case SNDRV_PCM_FORMAT_S16_LE: - word_length = 16; - break; - - case SNDRV_PCM_FORMAT_U24_3LE: - case SNDRV_PCM_FORMAT_S24_3LE: - word_length = 24; - break; - - case SNDRV_PCM_FORMAT_U24_LE: - case SNDRV_PCM_FORMAT_S24_LE: - word_length = 24; - break; - - case SNDRV_PCM_FORMAT_U32_LE: - case SNDRV_PCM_FORMAT_S32_LE: - word_length = 32; - break; - - default: - printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); - return -EINVAL; - } - davinci_config_channel_size(mcasp, word_length); if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) @@ -1405,12 +1417,13 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, max_channels *= tdm_slots; /* * If the already active stream has less channels than the calculated - * limnit based on the seirializers * tdm_slots, we need to use that as - * a constraint for the second stream. - * Otherwise (first stream or less allowed channels) we use the - * calculated constraint. + * limit based on the seirializers * tdm_slots, and only one serializer + * is in use we need to use that as a constraint for the second stream. + * Otherwise (first stream or less allowed channels or more than one + * serializer in use) we use the calculated constraint. */ - if (mcasp->channels && mcasp->channels < max_channels) + if (mcasp->channels && mcasp->channels < max_channels && + ruledata->serializers == 1) max_channels = mcasp->channels; /* * But we can always allow channels upto the amount of @@ -1471,6 +1484,7 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); mcasp->substreams[substream->stream] = NULL; + mcasp->active_serializers[substream->stream] = 0; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v2 1/2] ASoC: cs47l15: Add codec driver for Cirrus Logic CS47L15
by Charles Keepax 25 Jul '19

25 Jul '19

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Adds the codec driver for the CS47L15 SmartCodec. This is a multi-functional codec based on the Cirrus Logic Madera platform. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Jaswinder Jassal <jjassal(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Minor tweak to commit message Thanks, Charles sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs47l15.c | 1498 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/madera.c | 4 + 4 files changed, 1510 insertions(+) create mode 100644 sound/soc/codecs/cs47l15.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a5346299d..b027af8e9f39f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 + select SND_SOC_CS47L15 if MFD_CS47L15 select SND_SOC_CS47L35 if MFD_CS47L35 select SND_SOC_CS47L85 if MFD_CS47L85 select SND_SOC_CS47L90 if MFD_CS47L90 @@ -584,6 +585,9 @@ config SND_SOC_CS4349 config SND_SOC_CS47L24 tristate +config SND_SOC_CS47L15 + tristate + config SND_SOC_CS47L35 tristate @@ -722,9 +726,11 @@ config SND_SOC_LOCHNAGAR_SC config SND_SOC_MADERA tristate + default y if SND_SOC_CS47L15=y default y if SND_SOC_CS47L35=y default y if SND_SOC_CS47L85=y default y if SND_SOC_CS47L90=y + default m if SND_SOC_CS47L15=m default m if SND_SOC_CS47L35=m default m if SND_SOC_CS47L85=m default m if SND_SOC_CS47L90=m diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8cf43256..2de423621ba4a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-cs43130-objs := cs43130.o snd-soc-cs4341-objs := cs4341.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o +snd-soc-cs47l15-objs := cs47l15.o snd-soc-cs47l35-objs := cs47l35.o snd-soc-cs47l85-objs := cs47l85.o snd-soc-cs47l90-objs := cs47l90.o @@ -346,6 +347,7 @@ obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o obj-$(CONFIG_SND_SOC_CS4341) += snd-soc-cs4341.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o +obj-$(CONFIG_SND_SOC_CS47L15) += snd-soc-cs47l15.o obj-$(CONFIG_SND_SOC_CS47L35) += snd-soc-cs47l35.o obj-$(CONFIG_SND_SOC_CS47L85) += snd-soc-cs47l85.o obj-$(CONFIG_SND_SOC_CS47L90) += snd-soc-cs47l90.o diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c new file mode 100644 index 0000000000000..919f1c49d7cf1 --- /dev/null +++ b/sound/soc/codecs/cs47l15.c @@ -0,0 +1,1498 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// ALSA SoC Audio driver for CS47L15 codec +// +// Copyright (C) 2016-2019 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <linux/irqchip/irq-madera.h> +#include <linux/mfd/madera/core.h> +#include <linux/mfd/madera/registers.h> + +#include "madera.h" +#include "wm_adsp.h" + +#define CS47L15_NUM_ADSP 1 +#define CS47L15_MONO_OUTPUTS 1 + +/* Mid-mode registers */ +#define CS47L15_ADC_INT_BIAS_MASK 0x3800 +#define CS47L15_ADC_INT_BIAS_SHIFT 11 +#define CS47L15_PGA_BIAS_SEL_MASK 0x03 +#define CS47L15_PGA_BIAS_SEL_SHIFT 0 + +#define DRV_NAME "cs47l15-codec" + +struct cs47l15 { + struct madera_priv core; + struct madera_fll fll[2]; + + bool in1_lp_mode; +}; + +static const struct wm_adsp_region cs47l15_dsp1_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x080000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x0e0000 }, + { .type = WMFW_ADSP2_XM, .base = 0x0a0000 }, + { .type = WMFW_ADSP2_YM, .base = 0x0c0000 }, +}; + +static const char * const cs47l15_outdemux_texts[] = { + "HPOUT", + "EPOUT", +}; + +static SOC_ENUM_SINGLE_DECL(cs47l15_outdemux_enum, SND_SOC_NOPM, 0, + cs47l15_outdemux_texts); + +static const struct snd_kcontrol_new cs47l15_outdemux = + SOC_DAPM_ENUM_EXT("HPOUT1 Demux", cs47l15_outdemux_enum, + madera_out1_demux_get, madera_out1_demux_put); + +static int cs47l15_adsp_power_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera_priv *priv = &cs47l15->core; + struct madera *madera = priv->madera; + unsigned int freq; + int ret; + + ret = regmap_read(madera->regmap, MADERA_DSP_CLOCK_2, &freq); + if (ret != 0) { + dev_err(madera->dev, + "Failed to read MADERA_DSP_CLOCK_2: %d\n", ret); + return ret; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = madera_set_adsp_clk(&cs47l15->core, w->shift, freq); + if (ret) + return ret; + break; + default: + break; + } + + return wm_adsp_early_event(w, kcontrol, event); +} + +#define CS47L15_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTL Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static int cs47l15_in1_adc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !!cs47l15->in1_lp_mode; + + return 0; +} + +static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + + switch (ucontrol->value.integer.value[0]) { + case 0: + /* Set IN1 to normal mode */ + snd_soc_component_update_bits(component, MADERA_DMIC1L_CONTROL, + MADERA_IN1_OSR_MASK, + 5 << MADERA_IN1_OSR_SHIFT); + snd_soc_component_update_bits(component, CS47L15_ADC_INT_BIAS, + CS47L15_ADC_INT_BIAS_MASK, + 4 << CS47L15_ADC_INT_BIAS_SHIFT); + snd_soc_component_update_bits(component, CS47L15_PGA_BIAS_SEL, + CS47L15_PGA_BIAS_SEL_MASK, 0); + cs47l15->in1_lp_mode = false; + break; + default: + /* Set IN1 to LP mode */ + snd_soc_component_update_bits(component, MADERA_DMIC1L_CONTROL, + MADERA_IN1_OSR_MASK, + 4 << MADERA_IN1_OSR_SHIFT); + snd_soc_component_update_bits(component, CS47L15_ADC_INT_BIAS, + CS47L15_ADC_INT_BIAS_MASK, + 1 << CS47L15_ADC_INT_BIAS_SHIFT); + snd_soc_component_update_bits(component, CS47L15_PGA_BIAS_SEL, + CS47L15_PGA_BIAS_SEL_MASK, + 3 << CS47L15_PGA_BIAS_SEL_SHIFT); + cs47l15->in1_lp_mode = true; + break; + } + + return 0; +} + +static const struct snd_kcontrol_new cs47l15_snd_controls[] = { +SOC_ENUM("IN1 OSR", madera_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", madera_in_dmic_osr[1]), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", MADERA_IN1L_CONTROL, + MADERA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", MADERA_IN1R_CONTROL, + MADERA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv), + +SOC_ENUM("IN HPF Cutoff Frequency", madera_in_hpf_cut_enum), + +SOC_SINGLE("IN1L HPF Switch", MADERA_IN1L_CONTROL, MADERA_IN1L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN1R HPF Switch", MADERA_IN1R_CONTROL, MADERA_IN1R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2L HPF Switch", MADERA_IN2L_CONTROL, MADERA_IN2L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2R HPF Switch", MADERA_IN2R_CONTROL, MADERA_IN2R_HPF_SHIFT, 1, 0), + +SOC_SINGLE_TLV("IN1L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1L, + MADERA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1R, + MADERA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2L, + MADERA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2R, + MADERA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), + +SOC_ENUM("Input Ramp Up", madera_in_vi_ramp), +SOC_ENUM("Input Ramp Down", madera_in_vd_ramp), + +MADERA_MIXER_CONTROLS("EQ1", MADERA_EQ1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("EQ2", MADERA_EQ2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("EQ3", MADERA_EQ3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("EQ4", MADERA_EQ4MIX_INPUT_1_SOURCE), + +MADERA_EQ_CONTROL("EQ1 Coefficients", MADERA_EQ1_2), +SOC_SINGLE_TLV("EQ1 B1 Volume", MADERA_EQ1_1, MADERA_EQ1_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", MADERA_EQ1_1, MADERA_EQ1_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", MADERA_EQ1_1, MADERA_EQ1_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", MADERA_EQ1_2, MADERA_EQ1_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", MADERA_EQ1_2, MADERA_EQ1_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_EQ_CONTROL("EQ2 Coefficients", MADERA_EQ2_2), +SOC_SINGLE_TLV("EQ2 B1 Volume", MADERA_EQ2_1, MADERA_EQ2_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", MADERA_EQ2_1, MADERA_EQ2_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", MADERA_EQ2_1, MADERA_EQ2_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", MADERA_EQ2_2, MADERA_EQ2_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", MADERA_EQ2_2, MADERA_EQ2_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_EQ_CONTROL("EQ3 Coefficients", MADERA_EQ3_2), +SOC_SINGLE_TLV("EQ3 B1 Volume", MADERA_EQ3_1, MADERA_EQ3_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", MADERA_EQ3_1, MADERA_EQ3_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", MADERA_EQ3_1, MADERA_EQ3_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", MADERA_EQ3_2, MADERA_EQ3_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", MADERA_EQ3_2, MADERA_EQ3_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_EQ_CONTROL("EQ4 Coefficients", MADERA_EQ4_2), +SOC_SINGLE_TLV("EQ4 B1 Volume", MADERA_EQ4_1, MADERA_EQ4_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", MADERA_EQ4_1, MADERA_EQ4_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", MADERA_EQ4_1, MADERA_EQ4_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", MADERA_EQ4_2, MADERA_EQ4_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", MADERA_EQ4_2, MADERA_EQ4_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_MIXER_CONTROLS("DRC1L", MADERA_DRC1LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DRC1R", MADERA_DRC1RMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DRC2L", MADERA_DRC2LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DRC2R", MADERA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", MADERA_DRC1_CTRL1, 5, + MADERA_DRC1R_ENA | MADERA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", MADERA_DRC2_CTRL1, 5, + MADERA_DRC2R_ENA | MADERA_DRC2L_ENA), + +MADERA_MIXER_CONTROLS("LHPF1", MADERA_HPLP1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("LHPF2", MADERA_HPLP2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("LHPF3", MADERA_HPLP3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("LHPF4", MADERA_HPLP4MIX_INPUT_1_SOURCE), + +MADERA_LHPF_CONTROL("LHPF1 Coefficients", MADERA_HPLPF1_2), +MADERA_LHPF_CONTROL("LHPF2 Coefficients", MADERA_HPLPF2_2), +MADERA_LHPF_CONTROL("LHPF3 Coefficients", MADERA_HPLPF3_2), +MADERA_LHPF_CONTROL("LHPF4 Coefficients", MADERA_HPLPF4_2), + +SOC_ENUM("LHPF1 Mode", madera_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", madera_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", madera_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", madera_lhpf4_mode), + +MADERA_RATE_ENUM("ISRC1 FSL", madera_isrc_fsl[0]), +MADERA_RATE_ENUM("ISRC2 FSL", madera_isrc_fsl[1]), +MADERA_RATE_ENUM("ISRC1 FSH", madera_isrc_fsh[0]), +MADERA_RATE_ENUM("ISRC2 FSH", madera_isrc_fsh[1]), + +WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), + +MADERA_MIXER_CONTROLS("DSP1L", MADERA_DSP1LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DSP1R", MADERA_DSP1RMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", MADERA_COMFORT_NOISE_GENERATOR, + MADERA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, madera_noise_tlv), + +MADERA_MIXER_CONTROLS("HPOUT1L", MADERA_OUT1LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("HPOUT1R", MADERA_OUT1RMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("SPKOUTL", MADERA_OUT4LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("SPKDAT1L", MADERA_OUT5LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("SPKDAT1R", MADERA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 SC Protect Switch", MADERA_HP1_SHORT_CIRCUIT_CTRL, + MADERA_HP1_SC_ENA_SHIFT, 1, 0), + +SOC_SINGLE("SPKDAT1 High Performance Switch", MADERA_OUTPUT_PATH_CONFIG_5L, + MADERA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_1L, + MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", MADERA_DAC_DIGITAL_VOLUME_4L, + MADERA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_5L, + MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_1L, + MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_VOL_SHIFT, + 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", MADERA_DAC_DIGITAL_VOLUME_4L, + MADERA_OUT4L_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_5L, + MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_VOL_SHIFT, + 0xbf, 0, madera_digital_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", MADERA_PDM_SPK1_CTRL_1, MADERA_SPK1L_MUTE_SHIFT, + MADERA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_ENUM("Output Ramp Up", madera_out_vi_ramp), +SOC_ENUM("Output Ramp Down", madera_out_vd_ramp), + +SOC_SINGLE("Noise Gate Switch", MADERA_NOISE_GATE_CONTROL, + MADERA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", MADERA_NOISE_GATE_CONTROL, + MADERA_NGATE_THR_SHIFT, 7, 1, madera_ng_tlv), +SOC_ENUM("Noise Gate Hold", madera_ng_hold), + +SOC_SINGLE_BOOL_EXT("IN1 LP Mode Switch", 0, + cs47l15_in1_adc_get, cs47l15_in1_adc_put), + +CS47L15_NG_SRC("HPOUT1L", MADERA_NOISE_GATE_SELECT_1L), +CS47L15_NG_SRC("HPOUT1R", MADERA_NOISE_GATE_SELECT_1R), +CS47L15_NG_SRC("SPKOUTL", MADERA_NOISE_GATE_SELECT_4L), +CS47L15_NG_SRC("SPKDAT1L", MADERA_NOISE_GATE_SELECT_5L), +CS47L15_NG_SRC("SPKDAT1R", MADERA_NOISE_GATE_SELECT_5R), + +MADERA_MIXER_CONTROLS("AIF1TX1", MADERA_AIF1TX1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX2", MADERA_AIF1TX2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX3", MADERA_AIF1TX3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX4", MADERA_AIF1TX4MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX5", MADERA_AIF1TX5MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX6", MADERA_AIF1TX6MIX_INPUT_1_SOURCE), + +MADERA_MIXER_CONTROLS("AIF2TX1", MADERA_AIF2TX1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF2TX2", MADERA_AIF2TX2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF2TX3", MADERA_AIF2TX3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF2TX4", MADERA_AIF2TX4MIX_INPUT_1_SOURCE), + +MADERA_MIXER_CONTROLS("AIF3TX1", MADERA_AIF3TX1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF3TX2", MADERA_AIF3TX2MIX_INPUT_1_SOURCE), + +MADERA_GAINMUX_CONTROLS("SPDIF1TX1", MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE), +MADERA_GAINMUX_CONTROLS("SPDIF1TX2", MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), +}; + +MADERA_MIXER_ENUMS(EQ1, MADERA_EQ1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(EQ2, MADERA_EQ2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(EQ3, MADERA_EQ3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(EQ4, MADERA_EQ4MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(DRC1L, MADERA_DRC1LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DRC1R, MADERA_DRC1RMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DRC2L, MADERA_DRC2LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DRC2R, MADERA_DRC2RMIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(LHPF1, MADERA_HPLP1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(LHPF2, MADERA_HPLP2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(LHPF3, MADERA_HPLP3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(LHPF4, MADERA_HPLP4MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(DSP1L, MADERA_DSP1LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DSP1R, MADERA_DSP1RMIX_INPUT_1_SOURCE); +MADERA_DSP_AUX_ENUMS(DSP1, MADERA_DSP1AUX1MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(PWM1, MADERA_PWM1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(PWM2, MADERA_PWM2MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(OUT1L, MADERA_OUT1LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(OUT1R, MADERA_OUT1RMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(SPKOUTL, MADERA_OUT4LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(SPKDAT1L, MADERA_OUT5LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(SPKDAT1R, MADERA_OUT5RMIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(AIF1TX1, MADERA_AIF1TX1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX2, MADERA_AIF1TX2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX3, MADERA_AIF1TX3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX4, MADERA_AIF1TX4MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX5, MADERA_AIF1TX5MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX6, MADERA_AIF1TX6MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(AIF2TX1, MADERA_AIF2TX1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF2TX2, MADERA_AIF2TX2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF2TX3, MADERA_AIF2TX3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF2TX4, MADERA_AIF2TX4MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(AIF3TX1, MADERA_AIF3TX1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF3TX2, MADERA_AIF3TX2MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(SPD1TX1, MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(SPD1TX2, MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC1INT1, MADERA_ISRC1INT1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1INT2, MADERA_ISRC1INT2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1INT3, MADERA_ISRC1INT3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1INT4, MADERA_ISRC1INT4MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC1DEC1, MADERA_ISRC1DEC1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1DEC2, MADERA_ISRC1DEC2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1DEC3, MADERA_ISRC1DEC3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1DEC4, MADERA_ISRC1DEC4MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC2INT1, MADERA_ISRC2INT1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2INT2, MADERA_ISRC2INT2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2INT3, MADERA_ISRC2INT3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2INT4, MADERA_ISRC2INT4MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC2DEC1, MADERA_ISRC2DEC1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2DEC2, MADERA_ISRC2DEC2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2DEC3, MADERA_ISRC2DEC3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2DEC4, MADERA_ISRC2DEC4MIX_INPUT_1_SOURCE); + +static const char * const cs47l15_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "SPKOUTL", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int cs47l15_aec_loopback_values[] = { + 0, 1, 6, 8, 9, +}; + +static const struct soc_enum cs47l15_aec1_loopback = + SOC_VALUE_ENUM_SINGLE(MADERA_DAC_AEC_CONTROL_1, + MADERA_AEC1_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(cs47l15_aec_loopback_texts), + cs47l15_aec_loopback_texts, + cs47l15_aec_loopback_values); + +static const struct soc_enum cs47l15_aec2_loopback = + SOC_VALUE_ENUM_SINGLE(MADERA_DAC_AEC_CONTROL_2, + MADERA_AEC2_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(cs47l15_aec_loopback_texts), + cs47l15_aec_loopback_texts, + cs47l15_aec_loopback_values); + +static const struct snd_kcontrol_new cs47l15_aec_loopback_mux[] = { + SOC_DAPM_ENUM("AEC1 Loopback", cs47l15_aec1_loopback), + SOC_DAPM_ENUM("AEC2 Loopback", cs47l15_aec2_loopback), +}; + +static const struct snd_soc_dapm_widget cs47l15_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, + 0, madera_sysclk_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, + MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, + MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD1", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", MADERA_MIC_BIAS_CTRL_1, + MADERA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("MICBIAS1A", MADERA_MIC_BIAS_CTRL_5, + MADERA_MICB1A_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1B", MADERA_MIC_BIAS_CTRL_5, + MADERA_MICB1B_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1C", MADERA_MIC_BIAS_CTRL_5, + MADERA_MICB1C_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("FXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_FX, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("ISRC1CLK", SND_SOC_NOPM, + MADERA_DOM_GRP_ISRC1, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("ISRC2CLK", SND_SOC_NOPM, + MADERA_DOM_GRP_ISRC2, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("OUTCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_OUT, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("SPDCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_SPD, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, + MADERA_DOM_GRP_DSP1, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF1TXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_AIF1, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2TXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_AIF2, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF3TXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_AIF3, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("PWMCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_PWM, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1ALN"), +SND_SOC_DAPM_INPUT("IN1ALP"), +SND_SOC_DAPM_INPUT("IN1BLN"), +SND_SOC_DAPM_INPUT("IN1BLP"), +SND_SOC_DAPM_INPUT("IN1ARN"), +SND_SOC_DAPM_INPUT("IN1ARP"), +SND_SOC_DAPM_INPUT("IN1BRN"), +SND_SOC_DAPM_INPUT("IN1BRP"), +SND_SOC_DAPM_INPUT("IN2N"), +SND_SOC_DAPM_INPUT("IN2P"), +SND_SOC_DAPM_INPUT("SPKRXDAT"), + +SND_SOC_DAPM_MUX("IN1L Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[0]), +SND_SOC_DAPM_MUX("IN1R Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[1]), + +SND_SOC_DAPM_MUX("IN1L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]), +SND_SOC_DAPM_MUX("IN1R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]), + +SND_SOC_DAPM_MUX("IN2L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]), +SND_SOC_DAPM_MUX("IN2R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]), + +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), +SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), + +SND_SOC_DAPM_OUTPUT("DSP Trigger Out"), + +SND_SOC_DAPM_DEMUX("HPOUT1 Demux", SND_SOC_NOPM, 0, 0, &cs47l15_outdemux), + +SND_SOC_DAPM_PGA("PWM1 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX4_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + MADERA_OUT1L_ENA_SHIFT, 0, NULL, 0, madera_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + MADERA_OUT1R_ENA_SHIFT, 0, NULL, 0, madera_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", SND_SOC_NOPM, + MADERA_OUT4L_ENA_SHIFT, 0, NULL, 0, madera_spk_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", MADERA_OUTPUT_ENABLES_1, + MADERA_OUT5L_ENA_SHIFT, 0, NULL, 0, madera_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", MADERA_OUTPUT_ENABLES_1, + MADERA_OUT5R_ENA_SHIFT, 0, NULL, 0, madera_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("SPD1TX1", MADERA_SPD1_TX_CONTROL, + MADERA_SPD1_VAL1_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPD1TX2", MADERA_SPD1_TX_CONTROL, + MADERA_SPD1_VAL2_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPD1", MADERA_SPD1_TX_CONTROL, + MADERA_SPD1_ENA_SHIFT, 0, NULL, 0), + +/* + * mux_in widgets : arranged in the order of sources + * specified in MADERA_MIXER_INPUT_ROUTES + */ + +SND_SOC_DAPM_PGA("Noise Generator", MADERA_COMFORT_NOISE_GENERATOR, + MADERA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", MADERA_TONE_GENERATOR_1, + MADERA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", MADERA_TONE_GENERATOR_1, + MADERA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_MUX("AEC1 Loopback", MADERA_DAC_AEC_CONTROL_1, + MADERA_AEC1_LOOPBACK_ENA_SHIFT, 0, + &cs47l15_aec_loopback_mux[0]), +SND_SOC_DAPM_MUX("AEC2 Loopback", MADERA_DAC_AEC_CONTROL_2, + MADERA_AEC2_LOOPBACK_ENA_SHIFT, 0, + &cs47l15_aec_loopback_mux[1]), + +SND_SOC_DAPM_PGA_E("IN1L", MADERA_INPUT_ENABLES, MADERA_IN1L_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R", MADERA_INPUT_ENABLES, MADERA_IN1R_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L", MADERA_INPUT_ENABLES, MADERA_IN2L_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R", MADERA_INPUT_ENABLES, MADERA_IN2R_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX4_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA("EQ1", MADERA_EQ1_1, MADERA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", MADERA_EQ2_1, MADERA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", MADERA_EQ3_1, MADERA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", MADERA_EQ4_1, MADERA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", MADERA_DRC1_CTRL1, MADERA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", MADERA_DRC1_CTRL1, MADERA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", MADERA_DRC2_CTRL1, MADERA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", MADERA_DRC2_CTRL1, MADERA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", MADERA_HPLPF1_1, MADERA_LHPF1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", MADERA_HPLPF2_1, MADERA_LHPF2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", MADERA_HPLPF3_1, MADERA_LHPF3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", MADERA_HPLPF4_1, MADERA_LHPF4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC3", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC4", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT3", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT4", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC3", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC4", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT3", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT4", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT4_ENA_SHIFT, 0, NULL, 0), + +WM_ADSP2("DSP1", 0, cs47l15_adsp_power_ev), + +/* end of ordered widget list */ + +MADERA_MIXER_WIDGETS(EQ1, "EQ1"), +MADERA_MIXER_WIDGETS(EQ2, "EQ2"), +MADERA_MIXER_WIDGETS(EQ3, "EQ3"), +MADERA_MIXER_WIDGETS(EQ4, "EQ4"), + +MADERA_MIXER_WIDGETS(DRC1L, "DRC1L"), +MADERA_MIXER_WIDGETS(DRC1R, "DRC1R"), +MADERA_MIXER_WIDGETS(DRC2L, "DRC2L"), +MADERA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +SND_SOC_DAPM_SWITCH("DRC1 Activity Output", SND_SOC_NOPM, 0, 0, + &madera_drc_activity_output_mux[0]), +SND_SOC_DAPM_SWITCH("DRC2 Activity Output", SND_SOC_NOPM, 0, 0, + &madera_drc_activity_output_mux[1]), + +MADERA_MIXER_WIDGETS(LHPF1, "LHPF1"), +MADERA_MIXER_WIDGETS(LHPF2, "LHPF2"), +MADERA_MIXER_WIDGETS(LHPF3, "LHPF3"), +MADERA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +MADERA_MIXER_WIDGETS(PWM1, "PWM1"), +MADERA_MIXER_WIDGETS(PWM2, "PWM2"), + +MADERA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +MADERA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +MADERA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +MADERA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +MADERA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +MADERA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +MADERA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +MADERA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +MADERA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +MADERA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +MADERA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), + +MADERA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +MADERA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), +MADERA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"), +MADERA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"), + +MADERA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +MADERA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +MADERA_MUX_WIDGETS(SPD1TX1, "SPDIF1TX1"), +MADERA_MUX_WIDGETS(SPD1TX2, "SPDIF1TX2"), + +MADERA_DSP_WIDGETS(DSP1, "DSP1"), + +SND_SOC_DAPM_SWITCH("DSP1 Trigger Output", SND_SOC_NOPM, 0, 0, + &madera_dsp_trigger_output_mux[0]), + +MADERA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +MADERA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), +MADERA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"), +MADERA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"), + +MADERA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +MADERA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), +MADERA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"), +MADERA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"), + +MADERA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +MADERA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), +MADERA_MUX_WIDGETS(ISRC2DEC3, "ISRC2DEC3"), +MADERA_MUX_WIDGETS(ISRC2DEC4, "ISRC2DEC4"), + +MADERA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +MADERA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), +MADERA_MUX_WIDGETS(ISRC2INT3, "ISRC2INT3"), +MADERA_MUX_WIDGETS(ISRC2INT4, "ISRC2INT4"), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +SND_SOC_DAPM_OUTPUT("SPDIF1"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define MADERA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC1", "AEC1 Loopback" }, \ + { name, "AEC2", "AEC2 Loopback" }, \ + { name, "IN1L", "IN1L" }, \ + { name, "IN1R", "IN1R" }, \ + { name, "IN2L", "IN2L" }, \ + { name, "IN2R", "IN2R" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF2RX3", "AIF2RX3" }, \ + { name, "AIF2RX4", "AIF2RX4" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1DEC3", "ISRC1DEC3" }, \ + { name, "ISRC1DEC4", "ISRC1DEC4" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC1INT3", "ISRC1INT3" }, \ + { name, "ISRC1INT4", "ISRC1INT4" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2DEC3", "ISRC2DEC3" }, \ + { name, "ISRC2DEC4", "ISRC2DEC4" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" }, \ + { name, "ISRC2INT3", "ISRC2INT3" }, \ + { name, "ISRC2INT4", "ISRC2INT4" }, \ + { name, "DSP1.1", "DSP1" }, \ + { name, "DSP1.2", "DSP1" }, \ + { name, "DSP1.3", "DSP1" }, \ + { name, "DSP1.4", "DSP1" }, \ + { name, "DSP1.5", "DSP1" }, \ + { name, "DSP1.6", "DSP1" } + +static const struct snd_soc_dapm_route cs47l15_dapm_routes[] = { + /* Internal clock domains */ + { "EQ1", NULL, "FXCLK" }, + { "EQ2", NULL, "FXCLK" }, + { "EQ3", NULL, "FXCLK" }, + { "EQ4", NULL, "FXCLK" }, + { "DRC1L", NULL, "FXCLK" }, + { "DRC1R", NULL, "FXCLK" }, + { "DRC2L", NULL, "FXCLK" }, + { "DRC2R", NULL, "FXCLK" }, + { "LHPF1", NULL, "FXCLK" }, + { "LHPF2", NULL, "FXCLK" }, + { "LHPF3", NULL, "FXCLK" }, + { "LHPF4", NULL, "FXCLK" }, + { "PWM1 Mixer", NULL, "PWMCLK" }, + { "PWM2 Mixer", NULL, "PWMCLK" }, + { "OUT1L", NULL, "OUTCLK" }, + { "OUT1R", NULL, "OUTCLK" }, + { "OUT4L", NULL, "OUTCLK" }, + { "OUT5L", NULL, "OUTCLK" }, + { "OUT5R", NULL, "OUTCLK" }, + { "AIF1TX1", NULL, "AIF1TXCLK" }, + { "AIF1TX2", NULL, "AIF1TXCLK" }, + { "AIF1TX3", NULL, "AIF1TXCLK" }, + { "AIF1TX4", NULL, "AIF1TXCLK" }, + { "AIF1TX5", NULL, "AIF1TXCLK" }, + { "AIF1TX6", NULL, "AIF1TXCLK" }, + { "AIF2TX1", NULL, "AIF2TXCLK" }, + { "AIF2TX2", NULL, "AIF2TXCLK" }, + { "AIF2TX3", NULL, "AIF2TXCLK" }, + { "AIF2TX4", NULL, "AIF2TXCLK" }, + { "AIF3TX1", NULL, "AIF3TXCLK" }, + { "AIF3TX2", NULL, "AIF3TXCLK" }, + { "SPD1TX1", NULL, "SPDCLK" }, + { "SPD1TX2", NULL, "SPDCLK" }, + { "DSP1", NULL, "DSP1CLK" }, + { "ISRC1DEC1", NULL, "ISRC1CLK" }, + { "ISRC1DEC2", NULL, "ISRC1CLK" }, + { "ISRC1DEC3", NULL, "ISRC1CLK" }, + { "ISRC1DEC4", NULL, "ISRC1CLK" }, + { "ISRC1INT1", NULL, "ISRC1CLK" }, + { "ISRC1INT2", NULL, "ISRC1CLK" }, + { "ISRC1INT3", NULL, "ISRC1CLK" }, + { "ISRC1INT4", NULL, "ISRC1CLK" }, + { "ISRC2DEC1", NULL, "ISRC2CLK" }, + { "ISRC2DEC2", NULL, "ISRC2CLK" }, + { "ISRC2DEC3", NULL, "ISRC2CLK" }, + { "ISRC2DEC4", NULL, "ISRC2CLK" }, + { "ISRC2INT1", NULL, "ISRC2CLK" }, + { "ISRC2INT2", NULL, "ISRC2CLK" }, + { "ISRC2INT3", NULL, "ISRC2CLK" }, + { "ISRC2INT4", NULL, "ISRC2CLK" }, + + { "OUT1L", NULL, "CPVDD1" }, + { "OUT1R", NULL, "CPVDD1" }, + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + + { "SPD1", NULL, "SYSCLK" }, + { "SPD1", NULL, "SPD1TX1" }, + { "SPD1", NULL, "SPD1TX2" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + + { "MICBIAS1A", NULL, "MICBIAS1" }, + { "MICBIAS1B", NULL, "MICBIAS1" }, + { "MICBIAS1C", NULL, "MICBIAS1" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + { "AIF2 Capture", NULL, "AIF2TX3" }, + { "AIF2 Capture", NULL, "AIF2TX4" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + { "AIF2RX3", NULL, "AIF2 Playback" }, + { "AIF2RX4", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + { "Audio Trace DSP", NULL, "DSP1" }, + + { "IN1L Analog Mux", "A", "IN1ALN" }, + { "IN1L Analog Mux", "A", "IN1ALP" }, + { "IN1L Analog Mux", "B", "IN1BLN" }, + { "IN1L Analog Mux", "B", "IN1BLP" }, + { "IN1R Analog Mux", "A", "IN1ARN" }, + { "IN1R Analog Mux", "A", "IN1ARP" }, + { "IN1R Analog Mux", "B", "IN1BRN" }, + { "IN1R Analog Mux", "B", "IN1BRP" }, + + { "IN1L Mode", "Analog", "IN1L Analog Mux" }, + { "IN1R Mode", "Analog", "IN1R Analog Mux" }, + + { "IN1L Mode", "Digital", "IN1ALN" }, + { "IN1L Mode", "Digital", "IN1ALP" }, + { "IN1R Mode", "Digital", "IN1ALN" }, + { "IN1R Mode", "Digital", "IN1ALP" }, + + { "IN1L", NULL, "IN1L Mode" }, + { "IN1R", NULL, "IN1R Mode" }, + + { "IN2L Mode", "Analog", "IN2N" }, + { "IN2L Mode", "Analog", "IN2P" }, + + { "IN2L Mode", "Digital", "SPKRXDAT" }, + { "IN2R Mode", "Digital", "SPKRXDAT" }, + + { "IN2L", NULL, "IN2L Mode" }, + { "IN2R", NULL, "IN2R Mode" }, + + MADERA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + MADERA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + MADERA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + MADERA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + MADERA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + MADERA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + MADERA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + MADERA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + MADERA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + MADERA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + MADERA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + MADERA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + MADERA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + + MADERA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + MADERA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + MADERA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"), + MADERA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"), + + MADERA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + MADERA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + MADERA_MUX_ROUTES("SPD1TX1", "SPDIF1TX1"), + MADERA_MUX_ROUTES("SPD1TX2", "SPDIF1TX2"), + + MADERA_MIXER_ROUTES("EQ1", "EQ1"), + MADERA_MIXER_ROUTES("EQ2", "EQ2"), + MADERA_MIXER_ROUTES("EQ3", "EQ3"), + MADERA_MIXER_ROUTES("EQ4", "EQ4"), + + MADERA_MIXER_ROUTES("DRC1L", "DRC1L"), + MADERA_MIXER_ROUTES("DRC1R", "DRC1R"), + MADERA_MIXER_ROUTES("DRC2L", "DRC2L"), + MADERA_MIXER_ROUTES("DRC2R", "DRC2R"), + + MADERA_MIXER_ROUTES("LHPF1", "LHPF1"), + MADERA_MIXER_ROUTES("LHPF2", "LHPF2"), + MADERA_MIXER_ROUTES("LHPF3", "LHPF3"), + MADERA_MIXER_ROUTES("LHPF4", "LHPF4"), + + MADERA_DSP_ROUTES("DSP1"), + + { "DSP Trigger Out", NULL, "DSP1 Trigger Output" }, + + { "DSP1 Trigger Output", "Switch", "DSP1" }, + + MADERA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + MADERA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), + MADERA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"), + MADERA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"), + + MADERA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + MADERA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + MADERA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"), + MADERA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"), + + MADERA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + MADERA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + MADERA_MUX_ROUTES("ISRC2INT3", "ISRC2INT3"), + MADERA_MUX_ROUTES("ISRC2INT4", "ISRC2INT4"), + + MADERA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + MADERA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + MADERA_MUX_ROUTES("ISRC2DEC3", "ISRC2DEC3"), + MADERA_MUX_ROUTES("ISRC2DEC4", "ISRC2DEC4"), + + { "AEC1 Loopback", "HPOUT1L", "OUT1L" }, + { "AEC1 Loopback", "HPOUT1R", "OUT1R" }, + { "AEC2 Loopback", "HPOUT1L", "OUT1L" }, + { "AEC2 Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1 Demux", NULL, "OUT1L" }, + { "HPOUT1 Demux", NULL, "OUT1R" }, + { "HPOUTL", "HPOUT", "HPOUT1 Demux" }, + { "HPOUTR", "HPOUT", "HPOUT1 Demux" }, + { "EPOUTP", "EPOUT", "HPOUT1 Demux" }, + { "EPOUTN", "EPOUT", "HPOUT1 Demux" }, + + { "AEC1 Loopback", "SPKOUTL", "OUT4L" }, + { "AEC2 Loopback", "SPKOUTL", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC1 Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC1 Loopback", "SPKDAT1R", "OUT5R" }, + { "AEC2 Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC2 Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "SPDIF1", NULL, "SPD1" }, + + { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1 Activity Output" }, + { "DRC2 Signal Activity", NULL, "DRC2 Activity Output" }, + { "DRC1 Activity Output", "Switch", "DRC1L" }, + { "DRC1 Activity Output", "Switch", "DRC1R" }, + { "DRC2 Activity Output", "Switch", "DRC2L" }, + { "DRC2 Activity Output", "Switch", "DRC2R" }, +}; + +static int cs47l15_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + + switch (fll_id) { + case MADERA_FLL1_REFCLK: + return madera_set_fll_refclk(&cs47l15->fll[0], source, fref, + fout); + case MADERA_FLLAO_REFCLK: + return madera_set_fll_ao_refclk(&cs47l15->fll[1], source, fref, + fout); + case MADERA_FLL1_SYNCCLK: + return madera_set_fll_syncclk(&cs47l15->fll[0], source, fref, + fout); + default: + return -EINVAL; + } +} + +static struct snd_soc_dai_driver cs47l15_dai[] = { + { + .name = "cs47l15-aif1", + .id = 1, + .base = MADERA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .ops = &madera_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l15-aif2", + .id = 2, + .base = MADERA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .ops = &madera_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l15-aif3", + .id = 3, + .base = MADERA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .ops = &madera_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l15-cpu-trace", + .capture = { + .stream_name = "Audio Trace CPU", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .compress_new = snd_soc_new_compress, + }, + { + .name = "cs47l15-dsp-trace", + .capture = { + .stream_name = "Audio Trace DSP", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + }, +}; + +static int cs47l15_open(struct snd_compr_stream *stream) +{ + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera_priv *priv = &cs47l15->core; + struct madera *madera = priv->madera; + int n_adsp; + + if (strcmp(rtd->codec_dai->name, "cs47l15-dsp-trace") == 0) { + n_adsp = 0; + } else { + dev_err(madera->dev, + "No suitable compressed stream for DAI '%s'\n", + rtd->codec_dai->name); + return -EINVAL; + } + + return wm_adsp_compr_open(&priv->adsp[n_adsp], stream); +} + +static irqreturn_t cs47l15_adsp2_irq(int irq, void *data) +{ + struct cs47l15 *cs47l15 = data; + struct madera_priv *priv = &cs47l15->core; + struct madera *madera = priv->madera; + int ret; + + ret = wm_adsp_compr_handle_irq(&priv->adsp[0]); + if (ret == -ENODEV) { + dev_err(madera->dev, "Spurious compressed data IRQ\n"); + return IRQ_NONE; + } + + return IRQ_HANDLED; +} + +static irqreturn_t cs47l15_dsp_bus_error(int irq, void *data) +{ + struct wm_adsp *dsp = (struct wm_adsp *)data; + + return wm_adsp2_bus_error(dsp); +} + +static int cs47l15_component_probe(struct snd_soc_component *component) +{ + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera *madera = cs47l15->core.madera; + int ret; + + snd_soc_component_init_regmap(component, madera->regmap); + + mutex_lock(&madera->dapm_ptr_lock); + madera->dapm = snd_soc_component_get_dapm(component); + mutex_unlock(&madera->dapm_ptr_lock); + + ret = madera_init_inputs(component); + if (ret) + return ret; + + ret = madera_init_outputs(component, CS47L15_MONO_OUTPUTS); + if (ret) + return ret; + + snd_soc_component_disable_pin(component, "HAPTICS"); + + ret = snd_soc_add_component_controls(component, + madera_adsp_rate_controls, + CS47L15_NUM_ADSP); + if (ret) + return ret; + + wm_adsp2_component_probe(&cs47l15->core.adsp[0], component); + + return 0; +} + +static void cs47l15_component_remove(struct snd_soc_component *component) +{ + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera *madera = cs47l15->core.madera; + + mutex_lock(&madera->dapm_ptr_lock); + madera->dapm = NULL; + mutex_unlock(&madera->dapm_ptr_lock); + + wm_adsp2_component_remove(&cs47l15->core.adsp[0], component); +} + +#define CS47L15_DIG_VU 0x0200 + +static unsigned int cs47l15_digital_vu[] = { + MADERA_DAC_DIGITAL_VOLUME_1L, + MADERA_DAC_DIGITAL_VOLUME_1R, + MADERA_DAC_DIGITAL_VOLUME_4L, + MADERA_DAC_DIGITAL_VOLUME_5L, + MADERA_DAC_DIGITAL_VOLUME_5R, +}; + +static const struct snd_compr_ops cs47l15_compr_ops = { + .open = &cs47l15_open, + .free = &wm_adsp_compr_free, + .set_params = &wm_adsp_compr_set_params, + .get_caps = &wm_adsp_compr_get_caps, + .trigger = &wm_adsp_compr_trigger, + .pointer = &wm_adsp_compr_pointer, + .copy = &wm_adsp_compr_copy, +}; + +static const struct snd_soc_component_driver soc_component_dev_cs47l15 = { + .probe = &cs47l15_component_probe, + .remove = &cs47l15_component_remove, + .set_sysclk = &madera_set_sysclk, + .set_pll = &cs47l15_set_fll, + .name = DRV_NAME, + .compr_ops = &cs47l15_compr_ops, + .controls = cs47l15_snd_controls, + .num_controls = ARRAY_SIZE(cs47l15_snd_controls), + .dapm_widgets = cs47l15_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs47l15_dapm_widgets), + .dapm_routes = cs47l15_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs47l15_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int cs47l15_probe(struct platform_device *pdev) +{ + struct madera *madera = dev_get_drvdata(pdev->dev.parent); + struct cs47l15 *cs47l15; + int i, ret; + + BUILD_BUG_ON(ARRAY_SIZE(cs47l15_dai) > MADERA_MAX_DAI); + + /* quick exit if Madera irqchip driver hasn't completed probe */ + if (!madera->irq_dev) { + dev_dbg(&pdev->dev, "irqchip driver not ready\n"); + return -EPROBE_DEFER; + } + + cs47l15 = devm_kzalloc(&pdev->dev, sizeof(struct cs47l15), + GFP_KERNEL); + if (!cs47l15) + return -ENOMEM; + + platform_set_drvdata(pdev, cs47l15); + + cs47l15->core.madera = madera; + cs47l15->core.dev = &pdev->dev; + cs47l15->core.num_inputs = 4; + + ret = madera_core_init(&cs47l15->core); + if (ret) + return ret; + + ret = madera_init_overheat(&cs47l15->core); + if (ret) + goto error_core; + + ret = madera_request_irq(madera, MADERA_IRQ_DSP_IRQ1, + "ADSP2 Compressed IRQ", cs47l15_adsp2_irq, + cs47l15); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to request DSP IRQ: %d\n", ret); + goto error_overheat; + } + + ret = madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 1); + if (ret) + dev_warn(&pdev->dev, "Failed to set DSP IRQ wake: %d\n", ret); + + cs47l15->core.adsp[0].part = "cs47l15"; + cs47l15->core.adsp[0].num = 1; + cs47l15->core.adsp[0].type = WMFW_ADSP2; + cs47l15->core.adsp[0].rev = 2; + cs47l15->core.adsp[0].dev = madera->dev; + cs47l15->core.adsp[0].regmap = madera->regmap_32bit; + + cs47l15->core.adsp[0].base = MADERA_DSP1_CONFIG_1; + cs47l15->core.adsp[0].mem = cs47l15_dsp1_regions; + cs47l15->core.adsp[0].num_mems = ARRAY_SIZE(cs47l15_dsp1_regions); + + cs47l15->core.adsp[0].lock_regions = + WM_ADSP2_REGION_1 | WM_ADSP2_REGION_2 | WM_ADSP2_REGION_3; + + ret = wm_adsp2_init(&cs47l15->core.adsp[0]); + if (ret != 0) + goto error_dsp_irq; + + ret = madera_init_bus_error_irq(&cs47l15->core, 0, + cs47l15_dsp_bus_error); + if (ret) + goto error_adsp; + + madera_init_fll(madera, 1, MADERA_FLL1_CONTROL_1 - 1, + &cs47l15->fll[0]); + madera_init_fll(madera, 4, MADERA_FLLAO_CONTROL_1 - 1, + &cs47l15->fll[1]); + + for (i = 0; i < ARRAY_SIZE(cs47l15_dai); i++) + madera_init_dai(&cs47l15->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(cs47l15_digital_vu); i++) + regmap_update_bits(madera->regmap, cs47l15_digital_vu[i], + CS47L15_DIG_VU, CS47L15_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &soc_component_dev_cs47l15, + cs47l15_dai, + ARRAY_SIZE(cs47l15_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register component: %d\n", ret); + goto error_pm_runtime; + } + + return ret; + +error_pm_runtime: + pm_runtime_disable(&pdev->dev); + madera_free_bus_error_irq(&cs47l15->core, 0); +error_adsp: + wm_adsp2_remove(&cs47l15->core.adsp[0]); +error_dsp_irq: + madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 0); + madera_free_irq(madera, MADERA_IRQ_DSP_IRQ1, cs47l15); +error_overheat: + madera_free_overheat(&cs47l15->core); +error_core: + madera_core_free(&cs47l15->core); + + return ret; +} + +static int cs47l15_remove(struct platform_device *pdev) +{ + struct cs47l15 *cs47l15 = platform_get_drvdata(pdev); + + pm_runtime_disable(&pdev->dev); + + madera_free_bus_error_irq(&cs47l15->core, 0); + + wm_adsp2_remove(&cs47l15->core.adsp[0]); + + madera_set_irq_wake(cs47l15->core.madera, MADERA_IRQ_DSP_IRQ1, 0); + madera_free_irq(cs47l15->core.madera, MADERA_IRQ_DSP_IRQ1, cs47l15); + madera_free_overheat(&cs47l15->core); + madera_core_free(&cs47l15->core); + + return 0; +} + +static struct platform_driver cs47l15_codec_driver = { + .driver = { + .name = "cs47l15-codec", + }, + .probe = &cs47l15_probe, + .remove = &cs47l15_remove, +}; + +module_platform_driver(cs47l15_codec_driver); + +MODULE_SOFTDEP("pre: madera irq-madera arizona-micsupp"); +MODULE_DESCRIPTION("ASoC CS47L15 driver"); +MODULE_AUTHOR("Richard Fitzgerald <rf(a)opensource.cirrus.com>"); +MODULE_AUTHOR("Jaswinder Jassal <jjassal(a)opensource.cirrus.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cs47l15-codec"); diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 5f1e32a5a8556..1c00e2a115bcd 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -1041,6 +1041,10 @@ static void madera_configure_input_mode(struct madera *madera) int max_analogue_inputs, max_dmic_sup, i; switch (madera->type) { + case CS47L15: + max_analogue_inputs = 1; + max_dmic_sup = 2; + break; case CS47L35: max_analogue_inputs = 2; max_dmic_sup = 2; -- 2.11.0

2 3

[alsa-devel] [PATCH v2 1/3] treewide: add "WITH Linux-syscall-note" to SPDX tag of uapi headers
by Masahiro Yamada 25 Jul '19

25 Jul '19

UAPI headers licensed under GPL are supposed to have exception "WITH Linux-syscall-note" so that they can be included into non-GPL user space application code. The exception note is missing in some UAPI headers. Some of them slipped in by the treewide conversion commit b24413180f56 ("License cleanup: add SPDX GPL-2.0 license identifier to files with no license"). Just run: $ git show --oneline b24413180f56 -- arch/x86/include/uapi/asm/ I believe they are not intentional, and should be fixed too. This patch was generated by the following script: git grep -l --not -e Linux-syscall-note --and -e SPDX-License-Identifier \ -- :arch/*/include/uapi/asm/*.h :include/uapi/ :^*/Kbuild | while read file do sed -i -e '/[[:space:]]OR[[:space:]]/s/$GPL-[^[:space:]]*$/(\1 WITH Linux-syscall-note)/g' \ -e '/[[:space:]]or[[:space:]]/s/$GPL-[^[:space:]]*$/(\1 WITH Linux-syscall-note)/g' \ -e '/[[:space:]]OR[[:space:]]/!{/[[:space:]]or[[:space:]]/!s/$GPL-[^[:space:]]*$/\1 WITH Linux-syscall-note/g}' $file done After this patch is applied, there are 5 UAPI headers that do not contain "WITH Linux-syscall-note". They are kept untouched since this exception applies only to GPL variants. $ git grep --not -e Linux-syscall-note --and -e SPDX-License-Identifier \ -- :arch/*/include/uapi/asm/*.h :include/uapi/ :^*/Kbuild include/uapi/drm/panfrost_drm.h:/* SPDX-License-Identifier: MIT */ include/uapi/linux/batman_adv.h:/* SPDX-License-Identifier: MIT */ include/uapi/linux/qemu_fw_cfg.h:/* SPDX-License-Identifier: BSD-3-Clause */ include/uapi/linux/vbox_err.h:/* SPDX-License-Identifier: MIT */ include/uapi/linux/virtio_iommu.h:/* SPDX-License-Identifier: BSD-3-Clause */ Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- Changes in v2: - Fix missing parentheses arch/arm64/include/uapi/asm/bpf_perf_event.h | 2 +- arch/csky/include/uapi/asm/byteorder.h | 2 +- arch/csky/include/uapi/asm/cachectl.h | 2 +- arch/csky/include/uapi/asm/perf_regs.h | 2 +- arch/csky/include/uapi/asm/ptrace.h | 2 +- arch/csky/include/uapi/asm/sigcontext.h | 2 +- arch/csky/include/uapi/asm/unistd.h | 2 +- arch/nds32/include/uapi/asm/auxvec.h | 2 +- arch/nds32/include/uapi/asm/byteorder.h | 2 +- arch/nds32/include/uapi/asm/cachectl.h | 2 +- arch/nds32/include/uapi/asm/fp_udfiex_crtl.h | 2 +- arch/nds32/include/uapi/asm/param.h | 2 +- arch/nds32/include/uapi/asm/ptrace.h | 2 +- arch/nds32/include/uapi/asm/sigcontext.h | 2 +- arch/nds32/include/uapi/asm/unistd.h | 2 +- arch/powerpc/include/uapi/asm/bpf_perf_event.h | 2 +- arch/riscv/include/uapi/asm/auxvec.h | 2 +- arch/riscv/include/uapi/asm/bitsperlong.h | 2 +- arch/riscv/include/uapi/asm/byteorder.h | 2 +- arch/riscv/include/uapi/asm/hwcap.h | 2 +- arch/riscv/include/uapi/asm/ptrace.h | 2 +- arch/riscv/include/uapi/asm/sigcontext.h | 2 +- arch/riscv/include/uapi/asm/ucontext.h | 2 +- arch/s390/include/uapi/asm/bpf_perf_event.h | 2 +- arch/s390/include/uapi/asm/ipl.h | 2 +- arch/sh/include/uapi/asm/setup.h | 2 +- arch/sh/include/uapi/asm/types.h | 2 +- arch/sparc/include/uapi/asm/oradax.h | 2 +- arch/x86/include/uapi/asm/byteorder.h | 2 +- arch/x86/include/uapi/asm/hwcap2.h | 2 +- arch/x86/include/uapi/asm/sigcontext32.h | 2 +- arch/x86/include/uapi/asm/types.h | 2 +- include/uapi/linux/bpfilter.h | 2 +- include/uapi/linux/ipmi_bmc.h | 2 +- include/uapi/linux/isst_if.h | 2 +- include/uapi/linux/netfilter/nf_synproxy.h | 2 +- include/uapi/linux/psp-sev.h | 2 +- include/uapi/linux/rxrpc.h | 2 +- include/uapi/linux/usb/g_uvc.h | 2 +- include/uapi/linux/vbox_vmmdev_types.h | 2 +- include/uapi/linux/vboxguest.h | 2 +- include/uapi/linux/virtio_pmem.h | 2 +- include/uapi/linux/vmcore.h | 2 +- include/uapi/linux/wmi.h | 2 +- include/uapi/misc/fastrpc.h | 2 +- include/uapi/rdma/rvt-abi.h | 2 +- include/uapi/rdma/siw-abi.h | 2 +- include/uapi/scsi/scsi_bsg_ufs.h | 2 +- include/uapi/sound/skl-tplg-interface.h | 2 +- 49 files changed, 49 insertions(+), 49 deletions(-) diff --git a/arch/arm64/include/uapi/asm/bpf_perf_event.h b/arch/arm64/include/uapi/asm/bpf_perf_event.h index b551b741653d..5e1e648aeec4 100644 --- a/arch/arm64/include/uapi/asm/bpf_perf_event.h +++ b/arch/arm64/include/uapi/asm/bpf_perf_event.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI__ASM_BPF_PERF_EVENT_H__ #define _UAPI__ASM_BPF_PERF_EVENT_H__ diff --git a/arch/csky/include/uapi/asm/byteorder.h b/arch/csky/include/uapi/asm/byteorder.h index b079ec715cdf..d150cd664873 100644 --- a/arch/csky/include/uapi/asm/byteorder.h +++ b/arch/csky/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #ifndef __ASM_CSKY_BYTEORDER_H diff --git a/arch/csky/include/uapi/asm/cachectl.h b/arch/csky/include/uapi/asm/cachectl.h index ddf2f39aa925..ed7fad1ea20d 100644 --- a/arch/csky/include/uapi/asm/cachectl.h +++ b/arch/csky/include/uapi/asm/cachectl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef __ASM_CSKY_CACHECTL_H #define __ASM_CSKY_CACHECTL_H diff --git a/arch/csky/include/uapi/asm/perf_regs.h b/arch/csky/include/uapi/asm/perf_regs.h index ee323d818592..49d4e147a559 100644 --- a/arch/csky/include/uapi/asm/perf_regs.h +++ b/arch/csky/include/uapi/asm/perf_regs.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2019 Hangzhou C-SKY Microsystems co.,ltd. #ifndef _ASM_CSKY_PERF_REGS_H diff --git a/arch/csky/include/uapi/asm/ptrace.h b/arch/csky/include/uapi/asm/ptrace.h index 4e248d5b86ef..66b2268e324e 100644 --- a/arch/csky/include/uapi/asm/ptrace.h +++ b/arch/csky/include/uapi/asm/ptrace.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #ifndef _CSKY_PTRACE_H diff --git a/arch/csky/include/uapi/asm/sigcontext.h b/arch/csky/include/uapi/asm/sigcontext.h index e81e7ff11e36..670c020f2cb8 100644 --- a/arch/csky/include/uapi/asm/sigcontext.h +++ b/arch/csky/include/uapi/asm/sigcontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #ifndef __ASM_CSKY_SIGCONTEXT_H diff --git a/arch/csky/include/uapi/asm/unistd.h b/arch/csky/include/uapi/asm/unistd.h index ec60e49cea66..211c983c7282 100644 --- a/arch/csky/include/uapi/asm/unistd.h +++ b/arch/csky/include/uapi/asm/unistd.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #define __ARCH_WANT_SYS_CLONE diff --git a/arch/nds32/include/uapi/asm/auxvec.h b/arch/nds32/include/uapi/asm/auxvec.h index b5d58ea8decb..bc0b92ab8c15 100644 --- a/arch/nds32/include/uapi/asm/auxvec.h +++ b/arch/nds32/include/uapi/asm/auxvec.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __ASM_AUXVEC_H diff --git a/arch/nds32/include/uapi/asm/byteorder.h b/arch/nds32/include/uapi/asm/byteorder.h index 511e653c709d..c264ef12c49c 100644 --- a/arch/nds32/include/uapi/asm/byteorder.h +++ b/arch/nds32/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __NDS32_BYTEORDER_H__ diff --git a/arch/nds32/include/uapi/asm/cachectl.h b/arch/nds32/include/uapi/asm/cachectl.h index 73793662815c..31b9b439d819 100644 --- a/arch/nds32/include/uapi/asm/cachectl.h +++ b/arch/nds32/include/uapi/asm/cachectl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 1994, 1995, 1996 by Ralf Baechle // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef _ASM_CACHECTL diff --git a/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h b/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h index d54a5d6c6538..f17396db16ec 100644 --- a/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h +++ b/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* Copyright (C) 2005-2019 Andes Technology Corporation */ #ifndef _FP_UDF_IEX_CRTL_H #define _FP_UDF_IEX_CRTL_H diff --git a/arch/nds32/include/uapi/asm/param.h b/arch/nds32/include/uapi/asm/param.h index 2977534a6bd3..48d00328d328 100644 --- a/arch/nds32/include/uapi/asm/param.h +++ b/arch/nds32/include/uapi/asm/param.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __ASM_NDS32_PARAM_H diff --git a/arch/nds32/include/uapi/asm/ptrace.h b/arch/nds32/include/uapi/asm/ptrace.h index 1a6e01c00e6f..d76217c7c010 100644 --- a/arch/nds32/include/uapi/asm/ptrace.h +++ b/arch/nds32/include/uapi/asm/ptrace.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __UAPI_ASM_NDS32_PTRACE_H diff --git a/arch/nds32/include/uapi/asm/sigcontext.h b/arch/nds32/include/uapi/asm/sigcontext.h index dc89af7ddcc3..6c1e6648878f 100644 --- a/arch/nds32/include/uapi/asm/sigcontext.h +++ b/arch/nds32/include/uapi/asm/sigcontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef _ASMNDS32_SIGCONTEXT_H diff --git a/arch/nds32/include/uapi/asm/unistd.h b/arch/nds32/include/uapi/asm/unistd.h index a0b2f7b9c0f2..410795e280fe 100644 --- a/arch/nds32/include/uapi/asm/unistd.h +++ b/arch/nds32/include/uapi/asm/unistd.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #define __ARCH_WANT_STAT64 diff --git a/arch/powerpc/include/uapi/asm/bpf_perf_event.h b/arch/powerpc/include/uapi/asm/bpf_perf_event.h index b551b741653d..5e1e648aeec4 100644 --- a/arch/powerpc/include/uapi/asm/bpf_perf_event.h +++ b/arch/powerpc/include/uapi/asm/bpf_perf_event.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI__ASM_BPF_PERF_EVENT_H__ #define _UAPI__ASM_BPF_PERF_EVENT_H__ diff --git a/arch/riscv/include/uapi/asm/auxvec.h b/arch/riscv/include/uapi/asm/auxvec.h index 62716653554b..d86cb17bbabe 100644 --- a/arch/riscv/include/uapi/asm/auxvec.h +++ b/arch/riscv/include/uapi/asm/auxvec.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2015 Regents of the University of California diff --git a/arch/riscv/include/uapi/asm/bitsperlong.h b/arch/riscv/include/uapi/asm/bitsperlong.h index 0b9b58b57ff6..7d0b32e3b701 100644 --- a/arch/riscv/include/uapi/asm/bitsperlong.h +++ b/arch/riscv/include/uapi/asm/bitsperlong.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2015 Regents of the University of California diff --git a/arch/riscv/include/uapi/asm/byteorder.h b/arch/riscv/include/uapi/asm/byteorder.h index 1920debc09c0..f671e16bf6af 100644 --- a/arch/riscv/include/uapi/asm/byteorder.h +++ b/arch/riscv/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2015 Regents of the University of California diff --git a/arch/riscv/include/uapi/asm/hwcap.h b/arch/riscv/include/uapi/asm/hwcap.h index 7d786145183b..4e7646077056 100644 --- a/arch/riscv/include/uapi/asm/hwcap.h +++ b/arch/riscv/include/uapi/asm/hwcap.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copied from arch/arm64/include/asm/hwcap.h * diff --git a/arch/riscv/include/uapi/asm/ptrace.h b/arch/riscv/include/uapi/asm/ptrace.h index 92d8f7cd8f84..882547f6bd5c 100644 --- a/arch/riscv/include/uapi/asm/ptrace.h +++ b/arch/riscv/include/uapi/asm/ptrace.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 Regents of the University of California */ diff --git a/arch/riscv/include/uapi/asm/sigcontext.h b/arch/riscv/include/uapi/asm/sigcontext.h index 053f809e52ce..84f2dfcfdbce 100644 --- a/arch/riscv/include/uapi/asm/sigcontext.h +++ b/arch/riscv/include/uapi/asm/sigcontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 Regents of the University of California */ diff --git a/arch/riscv/include/uapi/asm/ucontext.h b/arch/riscv/include/uapi/asm/ucontext.h index b58e00cee2ec..411dd7b52ed6 100644 --- a/arch/riscv/include/uapi/asm/ucontext.h +++ b/arch/riscv/include/uapi/asm/ucontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2017 SiFive, Inc. diff --git a/arch/s390/include/uapi/asm/bpf_perf_event.h b/arch/s390/include/uapi/asm/bpf_perf_event.h index cefe7c7cd4f6..3ed42ff6da94 100644 --- a/arch/s390/include/uapi/asm/bpf_perf_event.h +++ b/arch/s390/include/uapi/asm/bpf_perf_event.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI__ASM_BPF_PERF_EVENT_H__ #define _UAPI__ASM_BPF_PERF_EVENT_H__ diff --git a/arch/s390/include/uapi/asm/ipl.h b/arch/s390/include/uapi/asm/ipl.h index fd32b1cd80d2..451ba7d08905 100644 --- a/arch/s390/include/uapi/asm/ipl.h +++ b/arch/s390/include/uapi/asm/ipl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_S390_UAPI_IPL_H #define _ASM_S390_UAPI_IPL_H diff --git a/arch/sh/include/uapi/asm/setup.h b/arch/sh/include/uapi/asm/setup.h index 1170dd2fb998..4bd19f80f9b0 100644 --- a/arch/sh/include/uapi/asm/setup.h +++ b/arch/sh/include/uapi/asm/setup.h @@ -1,2 +1,2 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #include <asm-generic/setup.h> diff --git a/arch/sh/include/uapi/asm/types.h b/arch/sh/include/uapi/asm/types.h index f83795fdc0da..68100e108ea6 100644 --- a/arch/sh/include/uapi/asm/types.h +++ b/arch/sh/include/uapi/asm/types.h @@ -1,2 +1,2 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #include <asm-generic/types.h> diff --git a/arch/sparc/include/uapi/asm/oradax.h b/arch/sparc/include/uapi/asm/oradax.h index 64c67f2ea33f..0dace69058ab 100644 --- a/arch/sparc/include/uapi/asm/oradax.h +++ b/arch/sparc/include/uapi/asm/oradax.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-or-later */ +/* SPDX-License-Identifier: GPL-2.0-or-later WITH Linux-syscall-note */ /* * Copyright (c) 2017, Oracle and/or its affiliates. All rights reserved. */ diff --git a/arch/x86/include/uapi/asm/byteorder.h b/arch/x86/include/uapi/asm/byteorder.h index 484e3cfd7ef2..149143cab9ff 100644 --- a/arch/x86/include/uapi/asm/byteorder.h +++ b/arch/x86/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_BYTEORDER_H #define _ASM_X86_BYTEORDER_H diff --git a/arch/x86/include/uapi/asm/hwcap2.h b/arch/x86/include/uapi/asm/hwcap2.h index 6ebaae90e207..8b2effe6efb8 100644 --- a/arch/x86/include/uapi/asm/hwcap2.h +++ b/arch/x86/include/uapi/asm/hwcap2.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_HWCAP2_H #define _ASM_X86_HWCAP2_H diff --git a/arch/x86/include/uapi/asm/sigcontext32.h b/arch/x86/include/uapi/asm/sigcontext32.h index 6b18e88de8a6..7114801d0499 100644 --- a/arch/x86/include/uapi/asm/sigcontext32.h +++ b/arch/x86/include/uapi/asm/sigcontext32.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_SIGCONTEXT32_H #define _ASM_X86_SIGCONTEXT32_H diff --git a/arch/x86/include/uapi/asm/types.h b/arch/x86/include/uapi/asm/types.h index df55e1ddb0c9..9d5c11a24279 100644 --- a/arch/x86/include/uapi/asm/types.h +++ b/arch/x86/include/uapi/asm/types.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_TYPES_H #define _ASM_X86_TYPES_H diff --git a/include/uapi/linux/bpfilter.h b/include/uapi/linux/bpfilter.h index 2ec3cc99ea4c..cbc1f5813f50 100644 --- a/include/uapi/linux/bpfilter.h +++ b/include/uapi/linux/bpfilter.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI_LINUX_BPFILTER_H #define _UAPI_LINUX_BPFILTER_H diff --git a/include/uapi/linux/ipmi_bmc.h b/include/uapi/linux/ipmi_bmc.h index 1670f0944227..782a03eb1086 100644 --- a/include/uapi/linux/ipmi_bmc.h +++ b/include/uapi/linux/ipmi_bmc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * Copyright (c) 2015-2018, Intel Corporation. */ diff --git a/include/uapi/linux/isst_if.h b/include/uapi/linux/isst_if.h index d10b832c58c5..0a52b7b093d3 100644 --- a/include/uapi/linux/isst_if.h +++ b/include/uapi/linux/isst_if.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * Intel Speed Select Interface: OS to hardware Interface * Copyright (c) 2019, Intel Corporation. diff --git a/include/uapi/linux/netfilter/nf_synproxy.h b/include/uapi/linux/netfilter/nf_synproxy.h index 6f3791c8946f..00d787f0260e 100644 --- a/include/uapi/linux/netfilter/nf_synproxy.h +++ b/include/uapi/linux/netfilter/nf_synproxy.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _NF_SYNPROXY_H #define _NF_SYNPROXY_H diff --git a/include/uapi/linux/psp-sev.h b/include/uapi/linux/psp-sev.h index 8654b2442f6a..592a0c1b77c9 100644 --- a/include/uapi/linux/psp-sev.h +++ b/include/uapi/linux/psp-sev.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Userspace interface for AMD Secure Encrypted Virtualization (SEV) * platform management commands. diff --git a/include/uapi/linux/rxrpc.h b/include/uapi/linux/rxrpc.h index 782069dcf607..4accfa7e266d 100644 --- a/include/uapi/linux/rxrpc.h +++ b/include/uapi/linux/rxrpc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-or-later */ +/* SPDX-License-Identifier: GPL-2.0-or-later WITH Linux-syscall-note */ /* Types and definitions for AF_RXRPC. * * Copyright (C) 2007 Red Hat, Inc. All Rights Reserved. diff --git a/include/uapi/linux/usb/g_uvc.h b/include/uapi/linux/usb/g_uvc.h index 3c9ee3020cbb..652f169a019e 100644 --- a/include/uapi/linux/usb/g_uvc.h +++ b/include/uapi/linux/usb/g_uvc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0+ */ +/* SPDX-License-Identifier: GPL-2.0+ WITH Linux-syscall-note */ /* * g_uvc.h -- USB Video Class Gadget driver API * diff --git a/include/uapi/linux/vbox_vmmdev_types.h b/include/uapi/linux/vbox_vmmdev_types.h index 26f39816af14..c27289fd619a 100644 --- a/include/uapi/linux/vbox_vmmdev_types.h +++ b/include/uapi/linux/vbox_vmmdev_types.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR CDDL-1.0) */ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR CDDL-1.0) */ /* * Virtual Device for Guest <-> VMM/Host communication, type definitions * which are also used for the vboxguest ioctl interface / by vboxsf diff --git a/include/uapi/linux/vboxguest.h b/include/uapi/linux/vboxguest.h index 612f0c7d3558..9cec58a6a5ea 100644 --- a/include/uapi/linux/vboxguest.h +++ b/include/uapi/linux/vboxguest.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR CDDL-1.0) */ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR CDDL-1.0) */ /* * VBoxGuest - VirtualBox Guest Additions Driver Interface. * diff --git a/include/uapi/linux/virtio_pmem.h b/include/uapi/linux/virtio_pmem.h index 9a63ed6d062f..b022787ffb94 100644 --- a/include/uapi/linux/virtio_pmem.h +++ b/include/uapi/linux/virtio_pmem.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause */ +/* SPDX-License-Identifier: (GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause */ /* * Definitions for virtio-pmem devices. * diff --git a/include/uapi/linux/vmcore.h b/include/uapi/linux/vmcore.h index 022619668e0e..3e9da91866ff 100644 --- a/include/uapi/linux/vmcore.h +++ b/include/uapi/linux/vmcore.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI_VMCORE_H #define _UAPI_VMCORE_H diff --git a/include/uapi/linux/wmi.h b/include/uapi/linux/wmi.h index c36f2d7675a4..7085c5dca9fa 100644 --- a/include/uapi/linux/wmi.h +++ b/include/uapi/linux/wmi.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * User API methods for ACPI-WMI mapping driver * diff --git a/include/uapi/misc/fastrpc.h b/include/uapi/misc/fastrpc.h index 6d701af9fc42..fb792e882cef 100644 --- a/include/uapi/misc/fastrpc.h +++ b/include/uapi/misc/fastrpc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef __QCOM_FASTRPC_H__ #define __QCOM_FASTRPC_H__ diff --git a/include/uapi/rdma/rvt-abi.h b/include/uapi/rdma/rvt-abi.h index 7328293c715c..7c05a02d2be5 100644 --- a/include/uapi/rdma/rvt-abi.h +++ b/include/uapi/rdma/rvt-abi.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ /* * This file contains defines, structures, etc. that are used diff --git a/include/uapi/rdma/siw-abi.h b/include/uapi/rdma/siw-abi.h index 3dd8071ace7b..7de68f1dc707 100644 --- a/include/uapi/rdma/siw-abi.h +++ b/include/uapi/rdma/siw-abi.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 or BSD-3-Clause */ +/* SPDX-License-Identifier: (GPL-2.0 WITH Linux-syscall-note) or BSD-3-Clause */ /* Authors: Bernard Metzler <bmt(a)zurich.ibm.com> */ /* Copyright (c) 2008-2019, IBM Corporation */ diff --git a/include/uapi/scsi/scsi_bsg_ufs.h b/include/uapi/scsi/scsi_bsg_ufs.h index 17c7abd0803a..9988db6ad244 100644 --- a/include/uapi/scsi/scsi_bsg_ufs.h +++ b/include/uapi/scsi/scsi_bsg_ufs.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * UFS Transport SGIO v4 BSG Message Support * diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index f39352cef382..9eee32f5e407 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * skl-tplg-interface.h - Intel DSP FW private data interface * -- 2.17.1

3 6

[alsa-devel] [PATCH 1/2] ASoC: cs47l15: Add codec driver for CS47L15
by Charles Keepax 25 Jul '19

25 Jul '19

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Adds the codec driver for the CS47L15 SmartCodec. This is a multi-functional codec based on the Cirrus Logic Madera platform. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Jaswinder Jassal <jjassal(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs47l15.c | 1498 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/madera.c | 4 + 4 files changed, 1510 insertions(+) create mode 100644 sound/soc/codecs/cs47l15.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a5346299d..b027af8e9f39f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 + select SND_SOC_CS47L15 if MFD_CS47L15 select SND_SOC_CS47L35 if MFD_CS47L35 select SND_SOC_CS47L85 if MFD_CS47L85 select SND_SOC_CS47L90 if MFD_CS47L90 @@ -584,6 +585,9 @@ config SND_SOC_CS4349 config SND_SOC_CS47L24 tristate +config SND_SOC_CS47L15 + tristate + config SND_SOC_CS47L35 tristate @@ -722,9 +726,11 @@ config SND_SOC_LOCHNAGAR_SC config SND_SOC_MADERA tristate + default y if SND_SOC_CS47L15=y default y if SND_SOC_CS47L35=y default y if SND_SOC_CS47L85=y default y if SND_SOC_CS47L90=y + default m if SND_SOC_CS47L15=m default m if SND_SOC_CS47L35=m default m if SND_SOC_CS47L85=m default m if SND_SOC_CS47L90=m diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8cf43256..2de423621ba4a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -65,6 +65,7 @@ snd-soc-cs43130-objs := cs43130.o snd-soc-cs4341-objs := cs4341.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o +snd-soc-cs47l15-objs := cs47l15.o snd-soc-cs47l35-objs := cs47l35.o snd-soc-cs47l85-objs := cs47l85.o snd-soc-cs47l90-objs := cs47l90.o @@ -346,6 +347,7 @@ obj-$(CONFIG_SND_SOC_CS43130) += snd-soc-cs43130.o obj-$(CONFIG_SND_SOC_CS4341) += snd-soc-cs4341.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o +obj-$(CONFIG_SND_SOC_CS47L15) += snd-soc-cs47l15.o obj-$(CONFIG_SND_SOC_CS47L35) += snd-soc-cs47l35.o obj-$(CONFIG_SND_SOC_CS47L85) += snd-soc-cs47l85.o obj-$(CONFIG_SND_SOC_CS47L90) += snd-soc-cs47l90.o diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c new file mode 100644 index 0000000000000..919f1c49d7cf1 --- /dev/null +++ b/sound/soc/codecs/cs47l15.c @@ -0,0 +1,1498 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// ALSA SoC Audio driver for CS47L15 codec +// +// Copyright (C) 2016-2019 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <linux/irqchip/irq-madera.h> +#include <linux/mfd/madera/core.h> +#include <linux/mfd/madera/registers.h> + +#include "madera.h" +#include "wm_adsp.h" + +#define CS47L15_NUM_ADSP 1 +#define CS47L15_MONO_OUTPUTS 1 + +/* Mid-mode registers */ +#define CS47L15_ADC_INT_BIAS_MASK 0x3800 +#define CS47L15_ADC_INT_BIAS_SHIFT 11 +#define CS47L15_PGA_BIAS_SEL_MASK 0x03 +#define CS47L15_PGA_BIAS_SEL_SHIFT 0 + +#define DRV_NAME "cs47l15-codec" + +struct cs47l15 { + struct madera_priv core; + struct madera_fll fll[2]; + + bool in1_lp_mode; +}; + +static const struct wm_adsp_region cs47l15_dsp1_regions[] = { + { .type = WMFW_ADSP2_PM, .base = 0x080000 }, + { .type = WMFW_ADSP2_ZM, .base = 0x0e0000 }, + { .type = WMFW_ADSP2_XM, .base = 0x0a0000 }, + { .type = WMFW_ADSP2_YM, .base = 0x0c0000 }, +}; + +static const char * const cs47l15_outdemux_texts[] = { + "HPOUT", + "EPOUT", +}; + +static SOC_ENUM_SINGLE_DECL(cs47l15_outdemux_enum, SND_SOC_NOPM, 0, + cs47l15_outdemux_texts); + +static const struct snd_kcontrol_new cs47l15_outdemux = + SOC_DAPM_ENUM_EXT("HPOUT1 Demux", cs47l15_outdemux_enum, + madera_out1_demux_get, madera_out1_demux_put); + +static int cs47l15_adsp_power_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera_priv *priv = &cs47l15->core; + struct madera *madera = priv->madera; + unsigned int freq; + int ret; + + ret = regmap_read(madera->regmap, MADERA_DSP_CLOCK_2, &freq); + if (ret != 0) { + dev_err(madera->dev, + "Failed to read MADERA_DSP_CLOCK_2: %d\n", ret); + return ret; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = madera_set_adsp_clk(&cs47l15->core, w->shift, freq); + if (ret) + return ret; + break; + default: + break; + } + + return wm_adsp_early_event(w, kcontrol, event); +} + +#define CS47L15_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG SPKOUTL Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static int cs47l15_in1_adc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !!cs47l15->in1_lp_mode; + + return 0; +} + +static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + + switch (ucontrol->value.integer.value[0]) { + case 0: + /* Set IN1 to normal mode */ + snd_soc_component_update_bits(component, MADERA_DMIC1L_CONTROL, + MADERA_IN1_OSR_MASK, + 5 << MADERA_IN1_OSR_SHIFT); + snd_soc_component_update_bits(component, CS47L15_ADC_INT_BIAS, + CS47L15_ADC_INT_BIAS_MASK, + 4 << CS47L15_ADC_INT_BIAS_SHIFT); + snd_soc_component_update_bits(component, CS47L15_PGA_BIAS_SEL, + CS47L15_PGA_BIAS_SEL_MASK, 0); + cs47l15->in1_lp_mode = false; + break; + default: + /* Set IN1 to LP mode */ + snd_soc_component_update_bits(component, MADERA_DMIC1L_CONTROL, + MADERA_IN1_OSR_MASK, + 4 << MADERA_IN1_OSR_SHIFT); + snd_soc_component_update_bits(component, CS47L15_ADC_INT_BIAS, + CS47L15_ADC_INT_BIAS_MASK, + 1 << CS47L15_ADC_INT_BIAS_SHIFT); + snd_soc_component_update_bits(component, CS47L15_PGA_BIAS_SEL, + CS47L15_PGA_BIAS_SEL_MASK, + 3 << CS47L15_PGA_BIAS_SEL_SHIFT); + cs47l15->in1_lp_mode = true; + break; + } + + return 0; +} + +static const struct snd_kcontrol_new cs47l15_snd_controls[] = { +SOC_ENUM("IN1 OSR", madera_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", madera_in_dmic_osr[1]), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", MADERA_IN1L_CONTROL, + MADERA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", MADERA_IN1R_CONTROL, + MADERA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, madera_ana_tlv), + +SOC_ENUM("IN HPF Cutoff Frequency", madera_in_hpf_cut_enum), + +SOC_SINGLE("IN1L HPF Switch", MADERA_IN1L_CONTROL, MADERA_IN1L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN1R HPF Switch", MADERA_IN1R_CONTROL, MADERA_IN1R_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2L HPF Switch", MADERA_IN2L_CONTROL, MADERA_IN2L_HPF_SHIFT, 1, 0), +SOC_SINGLE("IN2R HPF Switch", MADERA_IN2R_CONTROL, MADERA_IN2R_HPF_SHIFT, 1, 0), + +SOC_SINGLE_TLV("IN1L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1L, + MADERA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_1R, + MADERA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2L, + MADERA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", MADERA_ADC_DIGITAL_VOLUME_2R, + MADERA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), + +SOC_ENUM("Input Ramp Up", madera_in_vi_ramp), +SOC_ENUM("Input Ramp Down", madera_in_vd_ramp), + +MADERA_MIXER_CONTROLS("EQ1", MADERA_EQ1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("EQ2", MADERA_EQ2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("EQ3", MADERA_EQ3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("EQ4", MADERA_EQ4MIX_INPUT_1_SOURCE), + +MADERA_EQ_CONTROL("EQ1 Coefficients", MADERA_EQ1_2), +SOC_SINGLE_TLV("EQ1 B1 Volume", MADERA_EQ1_1, MADERA_EQ1_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", MADERA_EQ1_1, MADERA_EQ1_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", MADERA_EQ1_1, MADERA_EQ1_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", MADERA_EQ1_2, MADERA_EQ1_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", MADERA_EQ1_2, MADERA_EQ1_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_EQ_CONTROL("EQ2 Coefficients", MADERA_EQ2_2), +SOC_SINGLE_TLV("EQ2 B1 Volume", MADERA_EQ2_1, MADERA_EQ2_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", MADERA_EQ2_1, MADERA_EQ2_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", MADERA_EQ2_1, MADERA_EQ2_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", MADERA_EQ2_2, MADERA_EQ2_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", MADERA_EQ2_2, MADERA_EQ2_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_EQ_CONTROL("EQ3 Coefficients", MADERA_EQ3_2), +SOC_SINGLE_TLV("EQ3 B1 Volume", MADERA_EQ3_1, MADERA_EQ3_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", MADERA_EQ3_1, MADERA_EQ3_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", MADERA_EQ3_1, MADERA_EQ3_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", MADERA_EQ3_2, MADERA_EQ3_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", MADERA_EQ3_2, MADERA_EQ3_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_EQ_CONTROL("EQ4 Coefficients", MADERA_EQ4_2), +SOC_SINGLE_TLV("EQ4 B1 Volume", MADERA_EQ4_1, MADERA_EQ4_B1_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", MADERA_EQ4_1, MADERA_EQ4_B2_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", MADERA_EQ4_1, MADERA_EQ4_B3_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", MADERA_EQ4_2, MADERA_EQ4_B4_GAIN_SHIFT, + 24, 0, madera_eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", MADERA_EQ4_2, MADERA_EQ4_B5_GAIN_SHIFT, + 24, 0, madera_eq_tlv), + +MADERA_MIXER_CONTROLS("DRC1L", MADERA_DRC1LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DRC1R", MADERA_DRC1RMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DRC2L", MADERA_DRC2LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DRC2R", MADERA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", MADERA_DRC1_CTRL1, 5, + MADERA_DRC1R_ENA | MADERA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", MADERA_DRC2_CTRL1, 5, + MADERA_DRC2R_ENA | MADERA_DRC2L_ENA), + +MADERA_MIXER_CONTROLS("LHPF1", MADERA_HPLP1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("LHPF2", MADERA_HPLP2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("LHPF3", MADERA_HPLP3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("LHPF4", MADERA_HPLP4MIX_INPUT_1_SOURCE), + +MADERA_LHPF_CONTROL("LHPF1 Coefficients", MADERA_HPLPF1_2), +MADERA_LHPF_CONTROL("LHPF2 Coefficients", MADERA_HPLPF2_2), +MADERA_LHPF_CONTROL("LHPF3 Coefficients", MADERA_HPLPF3_2), +MADERA_LHPF_CONTROL("LHPF4 Coefficients", MADERA_HPLPF4_2), + +SOC_ENUM("LHPF1 Mode", madera_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", madera_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", madera_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", madera_lhpf4_mode), + +MADERA_RATE_ENUM("ISRC1 FSL", madera_isrc_fsl[0]), +MADERA_RATE_ENUM("ISRC2 FSL", madera_isrc_fsl[1]), +MADERA_RATE_ENUM("ISRC1 FSH", madera_isrc_fsh[0]), +MADERA_RATE_ENUM("ISRC2 FSH", madera_isrc_fsh[1]), + +WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), + +MADERA_MIXER_CONTROLS("DSP1L", MADERA_DSP1LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("DSP1R", MADERA_DSP1RMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", MADERA_COMFORT_NOISE_GENERATOR, + MADERA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, madera_noise_tlv), + +MADERA_MIXER_CONTROLS("HPOUT1L", MADERA_OUT1LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("HPOUT1R", MADERA_OUT1RMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("SPKOUTL", MADERA_OUT4LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("SPKDAT1L", MADERA_OUT5LMIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("SPKDAT1R", MADERA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 SC Protect Switch", MADERA_HP1_SHORT_CIRCUIT_CTRL, + MADERA_HP1_SC_ENA_SHIFT, 1, 0), + +SOC_SINGLE("SPKDAT1 High Performance Switch", MADERA_OUTPUT_PATH_CONFIG_5L, + MADERA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_1L, + MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", MADERA_DAC_DIGITAL_VOLUME_4L, + MADERA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", MADERA_DAC_DIGITAL_VOLUME_5L, + MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_1L, + MADERA_DAC_DIGITAL_VOLUME_1R, MADERA_OUT1L_VOL_SHIFT, + 0xbf, 0, madera_digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", MADERA_DAC_DIGITAL_VOLUME_4L, + MADERA_OUT4L_VOL_SHIFT, 0xbf, 0, madera_digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", MADERA_DAC_DIGITAL_VOLUME_5L, + MADERA_DAC_DIGITAL_VOLUME_5R, MADERA_OUT5L_VOL_SHIFT, + 0xbf, 0, madera_digital_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", MADERA_PDM_SPK1_CTRL_1, MADERA_SPK1L_MUTE_SHIFT, + MADERA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_ENUM("Output Ramp Up", madera_out_vi_ramp), +SOC_ENUM("Output Ramp Down", madera_out_vd_ramp), + +SOC_SINGLE("Noise Gate Switch", MADERA_NOISE_GATE_CONTROL, + MADERA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", MADERA_NOISE_GATE_CONTROL, + MADERA_NGATE_THR_SHIFT, 7, 1, madera_ng_tlv), +SOC_ENUM("Noise Gate Hold", madera_ng_hold), + +SOC_SINGLE_BOOL_EXT("IN1 LP Mode Switch", 0, + cs47l15_in1_adc_get, cs47l15_in1_adc_put), + +CS47L15_NG_SRC("HPOUT1L", MADERA_NOISE_GATE_SELECT_1L), +CS47L15_NG_SRC("HPOUT1R", MADERA_NOISE_GATE_SELECT_1R), +CS47L15_NG_SRC("SPKOUTL", MADERA_NOISE_GATE_SELECT_4L), +CS47L15_NG_SRC("SPKDAT1L", MADERA_NOISE_GATE_SELECT_5L), +CS47L15_NG_SRC("SPKDAT1R", MADERA_NOISE_GATE_SELECT_5R), + +MADERA_MIXER_CONTROLS("AIF1TX1", MADERA_AIF1TX1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX2", MADERA_AIF1TX2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX3", MADERA_AIF1TX3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX4", MADERA_AIF1TX4MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX5", MADERA_AIF1TX5MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF1TX6", MADERA_AIF1TX6MIX_INPUT_1_SOURCE), + +MADERA_MIXER_CONTROLS("AIF2TX1", MADERA_AIF2TX1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF2TX2", MADERA_AIF2TX2MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF2TX3", MADERA_AIF2TX3MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF2TX4", MADERA_AIF2TX4MIX_INPUT_1_SOURCE), + +MADERA_MIXER_CONTROLS("AIF3TX1", MADERA_AIF3TX1MIX_INPUT_1_SOURCE), +MADERA_MIXER_CONTROLS("AIF3TX2", MADERA_AIF3TX2MIX_INPUT_1_SOURCE), + +MADERA_GAINMUX_CONTROLS("SPDIF1TX1", MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE), +MADERA_GAINMUX_CONTROLS("SPDIF1TX2", MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), +}; + +MADERA_MIXER_ENUMS(EQ1, MADERA_EQ1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(EQ2, MADERA_EQ2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(EQ3, MADERA_EQ3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(EQ4, MADERA_EQ4MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(DRC1L, MADERA_DRC1LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DRC1R, MADERA_DRC1RMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DRC2L, MADERA_DRC2LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DRC2R, MADERA_DRC2RMIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(LHPF1, MADERA_HPLP1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(LHPF2, MADERA_HPLP2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(LHPF3, MADERA_HPLP3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(LHPF4, MADERA_HPLP4MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(DSP1L, MADERA_DSP1LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(DSP1R, MADERA_DSP1RMIX_INPUT_1_SOURCE); +MADERA_DSP_AUX_ENUMS(DSP1, MADERA_DSP1AUX1MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(PWM1, MADERA_PWM1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(PWM2, MADERA_PWM2MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(OUT1L, MADERA_OUT1LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(OUT1R, MADERA_OUT1RMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(SPKOUTL, MADERA_OUT4LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(SPKDAT1L, MADERA_OUT5LMIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(SPKDAT1R, MADERA_OUT5RMIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(AIF1TX1, MADERA_AIF1TX1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX2, MADERA_AIF1TX2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX3, MADERA_AIF1TX3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX4, MADERA_AIF1TX4MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX5, MADERA_AIF1TX5MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF1TX6, MADERA_AIF1TX6MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(AIF2TX1, MADERA_AIF2TX1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF2TX2, MADERA_AIF2TX2MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF2TX3, MADERA_AIF2TX3MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF2TX4, MADERA_AIF2TX4MIX_INPUT_1_SOURCE); + +MADERA_MIXER_ENUMS(AIF3TX1, MADERA_AIF3TX1MIX_INPUT_1_SOURCE); +MADERA_MIXER_ENUMS(AIF3TX2, MADERA_AIF3TX2MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(SPD1TX1, MADERA_SPDIF1TX1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(SPD1TX2, MADERA_SPDIF1TX2MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC1INT1, MADERA_ISRC1INT1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1INT2, MADERA_ISRC1INT2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1INT3, MADERA_ISRC1INT3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1INT4, MADERA_ISRC1INT4MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC1DEC1, MADERA_ISRC1DEC1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1DEC2, MADERA_ISRC1DEC2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1DEC3, MADERA_ISRC1DEC3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC1DEC4, MADERA_ISRC1DEC4MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC2INT1, MADERA_ISRC2INT1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2INT2, MADERA_ISRC2INT2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2INT3, MADERA_ISRC2INT3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2INT4, MADERA_ISRC2INT4MIX_INPUT_1_SOURCE); + +MADERA_MUX_ENUMS(ISRC2DEC1, MADERA_ISRC2DEC1MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2DEC2, MADERA_ISRC2DEC2MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2DEC3, MADERA_ISRC2DEC3MIX_INPUT_1_SOURCE); +MADERA_MUX_ENUMS(ISRC2DEC4, MADERA_ISRC2DEC4MIX_INPUT_1_SOURCE); + +static const char * const cs47l15_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "SPKOUTL", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int cs47l15_aec_loopback_values[] = { + 0, 1, 6, 8, 9, +}; + +static const struct soc_enum cs47l15_aec1_loopback = + SOC_VALUE_ENUM_SINGLE(MADERA_DAC_AEC_CONTROL_1, + MADERA_AEC1_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(cs47l15_aec_loopback_texts), + cs47l15_aec_loopback_texts, + cs47l15_aec_loopback_values); + +static const struct soc_enum cs47l15_aec2_loopback = + SOC_VALUE_ENUM_SINGLE(MADERA_DAC_AEC_CONTROL_2, + MADERA_AEC2_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(cs47l15_aec_loopback_texts), + cs47l15_aec_loopback_texts, + cs47l15_aec_loopback_values); + +static const struct snd_kcontrol_new cs47l15_aec_loopback_mux[] = { + SOC_DAPM_ENUM("AEC1 Loopback", cs47l15_aec1_loopback), + SOC_DAPM_ENUM("AEC2 Loopback", cs47l15_aec2_loopback), +}; + +static const struct snd_soc_dapm_widget cs47l15_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, + 0, madera_sysclk_ev, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, + MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, + MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD1", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", MADERA_MIC_BIAS_CTRL_1, + MADERA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("MICBIAS1A", MADERA_MIC_BIAS_CTRL_5, + MADERA_MICB1A_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1B", MADERA_MIC_BIAS_CTRL_5, + MADERA_MICB1B_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1C", MADERA_MIC_BIAS_CTRL_5, + MADERA_MICB1C_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_SUPPLY("FXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_FX, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("ISRC1CLK", SND_SOC_NOPM, + MADERA_DOM_GRP_ISRC1, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("ISRC2CLK", SND_SOC_NOPM, + MADERA_DOM_GRP_ISRC2, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("OUTCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_OUT, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("SPDCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_SPD, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, + MADERA_DOM_GRP_DSP1, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF1TXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_AIF1, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF2TXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_AIF2, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("AIF3TXCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_AIF3, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("PWMCLK", SND_SOC_NOPM, + MADERA_DOM_GRP_PWM, 0, + madera_domain_clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1ALN"), +SND_SOC_DAPM_INPUT("IN1ALP"), +SND_SOC_DAPM_INPUT("IN1BLN"), +SND_SOC_DAPM_INPUT("IN1BLP"), +SND_SOC_DAPM_INPUT("IN1ARN"), +SND_SOC_DAPM_INPUT("IN1ARP"), +SND_SOC_DAPM_INPUT("IN1BRN"), +SND_SOC_DAPM_INPUT("IN1BRP"), +SND_SOC_DAPM_INPUT("IN2N"), +SND_SOC_DAPM_INPUT("IN2P"), +SND_SOC_DAPM_INPUT("SPKRXDAT"), + +SND_SOC_DAPM_MUX("IN1L Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[0]), +SND_SOC_DAPM_MUX("IN1R Analog Mux", SND_SOC_NOPM, 0, 0, &madera_inmux[1]), + +SND_SOC_DAPM_MUX("IN1L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]), +SND_SOC_DAPM_MUX("IN1R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[0]), + +SND_SOC_DAPM_MUX("IN2L Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]), +SND_SOC_DAPM_MUX("IN2R Mode", SND_SOC_NOPM, 0, 0, &madera_inmode[1]), + +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), +SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), + +SND_SOC_DAPM_OUTPUT("DSP Trigger Out"), + +SND_SOC_DAPM_DEMUX("HPOUT1 Demux", SND_SOC_NOPM, 0, 0, &cs47l15_outdemux), + +SND_SOC_DAPM_PGA("PWM1 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", MADERA_PWM_DRIVE_1, MADERA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + MADERA_AIF1_TX_ENABLES, MADERA_AIF1TX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0, + MADERA_AIF2_TX_ENABLES, MADERA_AIF2TX4_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + MADERA_AIF3_TX_ENABLES, MADERA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + MADERA_OUT1L_ENA_SHIFT, 0, NULL, 0, madera_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + MADERA_OUT1R_ENA_SHIFT, 0, NULL, 0, madera_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", SND_SOC_NOPM, + MADERA_OUT4L_ENA_SHIFT, 0, NULL, 0, madera_spk_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", MADERA_OUTPUT_ENABLES_1, + MADERA_OUT5L_ENA_SHIFT, 0, NULL, 0, madera_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", MADERA_OUTPUT_ENABLES_1, + MADERA_OUT5R_ENA_SHIFT, 0, NULL, 0, madera_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_PGA("SPD1TX1", MADERA_SPD1_TX_CONTROL, + MADERA_SPD1_VAL1_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPD1TX2", MADERA_SPD1_TX_CONTROL, + MADERA_SPD1_VAL2_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPD1", MADERA_SPD1_TX_CONTROL, + MADERA_SPD1_ENA_SHIFT, 0, NULL, 0), + +/* + * mux_in widgets : arranged in the order of sources + * specified in MADERA_MIXER_INPUT_ROUTES + */ + +SND_SOC_DAPM_PGA("Noise Generator", MADERA_COMFORT_NOISE_GENERATOR, + MADERA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", MADERA_TONE_GENERATOR_1, + MADERA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", MADERA_TONE_GENERATOR_1, + MADERA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_MUX("AEC1 Loopback", MADERA_DAC_AEC_CONTROL_1, + MADERA_AEC1_LOOPBACK_ENA_SHIFT, 0, + &cs47l15_aec_loopback_mux[0]), +SND_SOC_DAPM_MUX("AEC2 Loopback", MADERA_DAC_AEC_CONTROL_2, + MADERA_AEC2_LOOPBACK_ENA_SHIFT, 0, + &cs47l15_aec_loopback_mux[1]), + +SND_SOC_DAPM_PGA_E("IN1L", MADERA_INPUT_ENABLES, MADERA_IN1L_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R", MADERA_INPUT_ENABLES, MADERA_IN1R_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L", MADERA_INPUT_ENABLES, MADERA_IN2L_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R", MADERA_INPUT_ENABLES, MADERA_IN2R_ENA_SHIFT, + 0, NULL, 0, madera_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + MADERA_AIF1_RX_ENABLES, MADERA_AIF1RX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0, + MADERA_AIF2_RX_ENABLES, MADERA_AIF2RX4_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + MADERA_AIF3_RX_ENABLES, MADERA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA("EQ1", MADERA_EQ1_1, MADERA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", MADERA_EQ2_1, MADERA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", MADERA_EQ3_1, MADERA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", MADERA_EQ4_1, MADERA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", MADERA_DRC1_CTRL1, MADERA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", MADERA_DRC1_CTRL1, MADERA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", MADERA_DRC2_CTRL1, MADERA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", MADERA_DRC2_CTRL1, MADERA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", MADERA_HPLPF1_1, MADERA_LHPF1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", MADERA_HPLPF2_1, MADERA_LHPF2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", MADERA_HPLPF3_1, MADERA_LHPF3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", MADERA_HPLPF4_1, MADERA_LHPF4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC3", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC4", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_DEC4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT3", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT4", MADERA_ISRC_1_CTRL_3, + MADERA_ISRC1_INT4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC3", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC4", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_DEC4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT3", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT4", MADERA_ISRC_2_CTRL_3, + MADERA_ISRC2_INT4_ENA_SHIFT, 0, NULL, 0), + +WM_ADSP2("DSP1", 0, cs47l15_adsp_power_ev), + +/* end of ordered widget list */ + +MADERA_MIXER_WIDGETS(EQ1, "EQ1"), +MADERA_MIXER_WIDGETS(EQ2, "EQ2"), +MADERA_MIXER_WIDGETS(EQ3, "EQ3"), +MADERA_MIXER_WIDGETS(EQ4, "EQ4"), + +MADERA_MIXER_WIDGETS(DRC1L, "DRC1L"), +MADERA_MIXER_WIDGETS(DRC1R, "DRC1R"), +MADERA_MIXER_WIDGETS(DRC2L, "DRC2L"), +MADERA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +SND_SOC_DAPM_SWITCH("DRC1 Activity Output", SND_SOC_NOPM, 0, 0, + &madera_drc_activity_output_mux[0]), +SND_SOC_DAPM_SWITCH("DRC2 Activity Output", SND_SOC_NOPM, 0, 0, + &madera_drc_activity_output_mux[1]), + +MADERA_MIXER_WIDGETS(LHPF1, "LHPF1"), +MADERA_MIXER_WIDGETS(LHPF2, "LHPF2"), +MADERA_MIXER_WIDGETS(LHPF3, "LHPF3"), +MADERA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +MADERA_MIXER_WIDGETS(PWM1, "PWM1"), +MADERA_MIXER_WIDGETS(PWM2, "PWM2"), + +MADERA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +MADERA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +MADERA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +MADERA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +MADERA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +MADERA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +MADERA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +MADERA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +MADERA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +MADERA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +MADERA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), + +MADERA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +MADERA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), +MADERA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"), +MADERA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"), + +MADERA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +MADERA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +MADERA_MUX_WIDGETS(SPD1TX1, "SPDIF1TX1"), +MADERA_MUX_WIDGETS(SPD1TX2, "SPDIF1TX2"), + +MADERA_DSP_WIDGETS(DSP1, "DSP1"), + +SND_SOC_DAPM_SWITCH("DSP1 Trigger Output", SND_SOC_NOPM, 0, 0, + &madera_dsp_trigger_output_mux[0]), + +MADERA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +MADERA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), +MADERA_MUX_WIDGETS(ISRC1DEC3, "ISRC1DEC3"), +MADERA_MUX_WIDGETS(ISRC1DEC4, "ISRC1DEC4"), + +MADERA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +MADERA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), +MADERA_MUX_WIDGETS(ISRC1INT3, "ISRC1INT3"), +MADERA_MUX_WIDGETS(ISRC1INT4, "ISRC1INT4"), + +MADERA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +MADERA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), +MADERA_MUX_WIDGETS(ISRC2DEC3, "ISRC2DEC3"), +MADERA_MUX_WIDGETS(ISRC2DEC4, "ISRC2DEC4"), + +MADERA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +MADERA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), +MADERA_MUX_WIDGETS(ISRC2INT3, "ISRC2INT3"), +MADERA_MUX_WIDGETS(ISRC2INT4, "ISRC2INT4"), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +SND_SOC_DAPM_OUTPUT("SPDIF1"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define MADERA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC1", "AEC1 Loopback" }, \ + { name, "AEC2", "AEC2 Loopback" }, \ + { name, "IN1L", "IN1L" }, \ + { name, "IN1R", "IN1R" }, \ + { name, "IN2L", "IN2L" }, \ + { name, "IN2R", "IN2R" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF2RX3", "AIF2RX3" }, \ + { name, "AIF2RX4", "AIF2RX4" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1DEC3", "ISRC1DEC3" }, \ + { name, "ISRC1DEC4", "ISRC1DEC4" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC1INT3", "ISRC1INT3" }, \ + { name, "ISRC1INT4", "ISRC1INT4" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2DEC3", "ISRC2DEC3" }, \ + { name, "ISRC2DEC4", "ISRC2DEC4" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" }, \ + { name, "ISRC2INT3", "ISRC2INT3" }, \ + { name, "ISRC2INT4", "ISRC2INT4" }, \ + { name, "DSP1.1", "DSP1" }, \ + { name, "DSP1.2", "DSP1" }, \ + { name, "DSP1.3", "DSP1" }, \ + { name, "DSP1.4", "DSP1" }, \ + { name, "DSP1.5", "DSP1" }, \ + { name, "DSP1.6", "DSP1" } + +static const struct snd_soc_dapm_route cs47l15_dapm_routes[] = { + /* Internal clock domains */ + { "EQ1", NULL, "FXCLK" }, + { "EQ2", NULL, "FXCLK" }, + { "EQ3", NULL, "FXCLK" }, + { "EQ4", NULL, "FXCLK" }, + { "DRC1L", NULL, "FXCLK" }, + { "DRC1R", NULL, "FXCLK" }, + { "DRC2L", NULL, "FXCLK" }, + { "DRC2R", NULL, "FXCLK" }, + { "LHPF1", NULL, "FXCLK" }, + { "LHPF2", NULL, "FXCLK" }, + { "LHPF3", NULL, "FXCLK" }, + { "LHPF4", NULL, "FXCLK" }, + { "PWM1 Mixer", NULL, "PWMCLK" }, + { "PWM2 Mixer", NULL, "PWMCLK" }, + { "OUT1L", NULL, "OUTCLK" }, + { "OUT1R", NULL, "OUTCLK" }, + { "OUT4L", NULL, "OUTCLK" }, + { "OUT5L", NULL, "OUTCLK" }, + { "OUT5R", NULL, "OUTCLK" }, + { "AIF1TX1", NULL, "AIF1TXCLK" }, + { "AIF1TX2", NULL, "AIF1TXCLK" }, + { "AIF1TX3", NULL, "AIF1TXCLK" }, + { "AIF1TX4", NULL, "AIF1TXCLK" }, + { "AIF1TX5", NULL, "AIF1TXCLK" }, + { "AIF1TX6", NULL, "AIF1TXCLK" }, + { "AIF2TX1", NULL, "AIF2TXCLK" }, + { "AIF2TX2", NULL, "AIF2TXCLK" }, + { "AIF2TX3", NULL, "AIF2TXCLK" }, + { "AIF2TX4", NULL, "AIF2TXCLK" }, + { "AIF3TX1", NULL, "AIF3TXCLK" }, + { "AIF3TX2", NULL, "AIF3TXCLK" }, + { "SPD1TX1", NULL, "SPDCLK" }, + { "SPD1TX2", NULL, "SPDCLK" }, + { "DSP1", NULL, "DSP1CLK" }, + { "ISRC1DEC1", NULL, "ISRC1CLK" }, + { "ISRC1DEC2", NULL, "ISRC1CLK" }, + { "ISRC1DEC3", NULL, "ISRC1CLK" }, + { "ISRC1DEC4", NULL, "ISRC1CLK" }, + { "ISRC1INT1", NULL, "ISRC1CLK" }, + { "ISRC1INT2", NULL, "ISRC1CLK" }, + { "ISRC1INT3", NULL, "ISRC1CLK" }, + { "ISRC1INT4", NULL, "ISRC1CLK" }, + { "ISRC2DEC1", NULL, "ISRC2CLK" }, + { "ISRC2DEC2", NULL, "ISRC2CLK" }, + { "ISRC2DEC3", NULL, "ISRC2CLK" }, + { "ISRC2DEC4", NULL, "ISRC2CLK" }, + { "ISRC2INT1", NULL, "ISRC2CLK" }, + { "ISRC2INT2", NULL, "ISRC2CLK" }, + { "ISRC2INT3", NULL, "ISRC2CLK" }, + { "ISRC2INT4", NULL, "ISRC2CLK" }, + + { "OUT1L", NULL, "CPVDD1" }, + { "OUT1R", NULL, "CPVDD1" }, + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + + { "SPD1", NULL, "SYSCLK" }, + { "SPD1", NULL, "SPD1TX1" }, + { "SPD1", NULL, "SPD1TX2" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + + { "MICBIAS1A", NULL, "MICBIAS1" }, + { "MICBIAS1B", NULL, "MICBIAS1" }, + { "MICBIAS1C", NULL, "MICBIAS1" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + { "AIF2 Capture", NULL, "AIF2TX3" }, + { "AIF2 Capture", NULL, "AIF2TX4" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + { "AIF2RX3", NULL, "AIF2 Playback" }, + { "AIF2RX4", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + { "Audio Trace DSP", NULL, "DSP1" }, + + { "IN1L Analog Mux", "A", "IN1ALN" }, + { "IN1L Analog Mux", "A", "IN1ALP" }, + { "IN1L Analog Mux", "B", "IN1BLN" }, + { "IN1L Analog Mux", "B", "IN1BLP" }, + { "IN1R Analog Mux", "A", "IN1ARN" }, + { "IN1R Analog Mux", "A", "IN1ARP" }, + { "IN1R Analog Mux", "B", "IN1BRN" }, + { "IN1R Analog Mux", "B", "IN1BRP" }, + + { "IN1L Mode", "Analog", "IN1L Analog Mux" }, + { "IN1R Mode", "Analog", "IN1R Analog Mux" }, + + { "IN1L Mode", "Digital", "IN1ALN" }, + { "IN1L Mode", "Digital", "IN1ALP" }, + { "IN1R Mode", "Digital", "IN1ALN" }, + { "IN1R Mode", "Digital", "IN1ALP" }, + + { "IN1L", NULL, "IN1L Mode" }, + { "IN1R", NULL, "IN1R Mode" }, + + { "IN2L Mode", "Analog", "IN2N" }, + { "IN2L Mode", "Analog", "IN2P" }, + + { "IN2L Mode", "Digital", "SPKRXDAT" }, + { "IN2R Mode", "Digital", "SPKRXDAT" }, + + { "IN2L", NULL, "IN2L Mode" }, + { "IN2R", NULL, "IN2R Mode" }, + + MADERA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + MADERA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + MADERA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + MADERA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + MADERA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + MADERA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + MADERA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + MADERA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + MADERA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + MADERA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + MADERA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + MADERA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + MADERA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + + MADERA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + MADERA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + MADERA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"), + MADERA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"), + + MADERA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + MADERA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + MADERA_MUX_ROUTES("SPD1TX1", "SPDIF1TX1"), + MADERA_MUX_ROUTES("SPD1TX2", "SPDIF1TX2"), + + MADERA_MIXER_ROUTES("EQ1", "EQ1"), + MADERA_MIXER_ROUTES("EQ2", "EQ2"), + MADERA_MIXER_ROUTES("EQ3", "EQ3"), + MADERA_MIXER_ROUTES("EQ4", "EQ4"), + + MADERA_MIXER_ROUTES("DRC1L", "DRC1L"), + MADERA_MIXER_ROUTES("DRC1R", "DRC1R"), + MADERA_MIXER_ROUTES("DRC2L", "DRC2L"), + MADERA_MIXER_ROUTES("DRC2R", "DRC2R"), + + MADERA_MIXER_ROUTES("LHPF1", "LHPF1"), + MADERA_MIXER_ROUTES("LHPF2", "LHPF2"), + MADERA_MIXER_ROUTES("LHPF3", "LHPF3"), + MADERA_MIXER_ROUTES("LHPF4", "LHPF4"), + + MADERA_DSP_ROUTES("DSP1"), + + { "DSP Trigger Out", NULL, "DSP1 Trigger Output" }, + + { "DSP1 Trigger Output", "Switch", "DSP1" }, + + MADERA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + MADERA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), + MADERA_MUX_ROUTES("ISRC1INT3", "ISRC1INT3"), + MADERA_MUX_ROUTES("ISRC1INT4", "ISRC1INT4"), + + MADERA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + MADERA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + MADERA_MUX_ROUTES("ISRC1DEC3", "ISRC1DEC3"), + MADERA_MUX_ROUTES("ISRC1DEC4", "ISRC1DEC4"), + + MADERA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + MADERA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + MADERA_MUX_ROUTES("ISRC2INT3", "ISRC2INT3"), + MADERA_MUX_ROUTES("ISRC2INT4", "ISRC2INT4"), + + MADERA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + MADERA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + MADERA_MUX_ROUTES("ISRC2DEC3", "ISRC2DEC3"), + MADERA_MUX_ROUTES("ISRC2DEC4", "ISRC2DEC4"), + + { "AEC1 Loopback", "HPOUT1L", "OUT1L" }, + { "AEC1 Loopback", "HPOUT1R", "OUT1R" }, + { "AEC2 Loopback", "HPOUT1L", "OUT1L" }, + { "AEC2 Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1 Demux", NULL, "OUT1L" }, + { "HPOUT1 Demux", NULL, "OUT1R" }, + { "HPOUTL", "HPOUT", "HPOUT1 Demux" }, + { "HPOUTR", "HPOUT", "HPOUT1 Demux" }, + { "EPOUTP", "EPOUT", "HPOUT1 Demux" }, + { "EPOUTN", "EPOUT", "HPOUT1 Demux" }, + + { "AEC1 Loopback", "SPKOUTL", "OUT4L" }, + { "AEC2 Loopback", "SPKOUTL", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC1 Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC1 Loopback", "SPKDAT1R", "OUT5R" }, + { "AEC2 Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC2 Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "SPDIF1", NULL, "SPD1" }, + + { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1 Activity Output" }, + { "DRC2 Signal Activity", NULL, "DRC2 Activity Output" }, + { "DRC1 Activity Output", "Switch", "DRC1L" }, + { "DRC1 Activity Output", "Switch", "DRC1R" }, + { "DRC2 Activity Output", "Switch", "DRC2L" }, + { "DRC2 Activity Output", "Switch", "DRC2R" }, +}; + +static int cs47l15_set_fll(struct snd_soc_component *component, int fll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + + switch (fll_id) { + case MADERA_FLL1_REFCLK: + return madera_set_fll_refclk(&cs47l15->fll[0], source, fref, + fout); + case MADERA_FLLAO_REFCLK: + return madera_set_fll_ao_refclk(&cs47l15->fll[1], source, fref, + fout); + case MADERA_FLL1_SYNCCLK: + return madera_set_fll_syncclk(&cs47l15->fll[0], source, fref, + fout); + default: + return -EINVAL; + } +} + +static struct snd_soc_dai_driver cs47l15_dai[] = { + { + .name = "cs47l15-aif1", + .id = 1, + .base = MADERA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .ops = &madera_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l15-aif2", + .id = 2, + .base = MADERA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .ops = &madera_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l15-aif3", + .id = 3, + .base = MADERA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .ops = &madera_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + }, + { + .name = "cs47l15-cpu-trace", + .capture = { + .stream_name = "Audio Trace CPU", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + .compress_new = snd_soc_new_compress, + }, + { + .name = "cs47l15-dsp-trace", + .capture = { + .stream_name = "Audio Trace DSP", + .channels_min = 1, + .channels_max = 6, + .rates = MADERA_RATES, + .formats = MADERA_FORMATS, + }, + }, +}; + +static int cs47l15_open(struct snd_compr_stream *stream) +{ + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera_priv *priv = &cs47l15->core; + struct madera *madera = priv->madera; + int n_adsp; + + if (strcmp(rtd->codec_dai->name, "cs47l15-dsp-trace") == 0) { + n_adsp = 0; + } else { + dev_err(madera->dev, + "No suitable compressed stream for DAI '%s'\n", + rtd->codec_dai->name); + return -EINVAL; + } + + return wm_adsp_compr_open(&priv->adsp[n_adsp], stream); +} + +static irqreturn_t cs47l15_adsp2_irq(int irq, void *data) +{ + struct cs47l15 *cs47l15 = data; + struct madera_priv *priv = &cs47l15->core; + struct madera *madera = priv->madera; + int ret; + + ret = wm_adsp_compr_handle_irq(&priv->adsp[0]); + if (ret == -ENODEV) { + dev_err(madera->dev, "Spurious compressed data IRQ\n"); + return IRQ_NONE; + } + + return IRQ_HANDLED; +} + +static irqreturn_t cs47l15_dsp_bus_error(int irq, void *data) +{ + struct wm_adsp *dsp = (struct wm_adsp *)data; + + return wm_adsp2_bus_error(dsp); +} + +static int cs47l15_component_probe(struct snd_soc_component *component) +{ + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera *madera = cs47l15->core.madera; + int ret; + + snd_soc_component_init_regmap(component, madera->regmap); + + mutex_lock(&madera->dapm_ptr_lock); + madera->dapm = snd_soc_component_get_dapm(component); + mutex_unlock(&madera->dapm_ptr_lock); + + ret = madera_init_inputs(component); + if (ret) + return ret; + + ret = madera_init_outputs(component, CS47L15_MONO_OUTPUTS); + if (ret) + return ret; + + snd_soc_component_disable_pin(component, "HAPTICS"); + + ret = snd_soc_add_component_controls(component, + madera_adsp_rate_controls, + CS47L15_NUM_ADSP); + if (ret) + return ret; + + wm_adsp2_component_probe(&cs47l15->core.adsp[0], component); + + return 0; +} + +static void cs47l15_component_remove(struct snd_soc_component *component) +{ + struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + struct madera *madera = cs47l15->core.madera; + + mutex_lock(&madera->dapm_ptr_lock); + madera->dapm = NULL; + mutex_unlock(&madera->dapm_ptr_lock); + + wm_adsp2_component_remove(&cs47l15->core.adsp[0], component); +} + +#define CS47L15_DIG_VU 0x0200 + +static unsigned int cs47l15_digital_vu[] = { + MADERA_DAC_DIGITAL_VOLUME_1L, + MADERA_DAC_DIGITAL_VOLUME_1R, + MADERA_DAC_DIGITAL_VOLUME_4L, + MADERA_DAC_DIGITAL_VOLUME_5L, + MADERA_DAC_DIGITAL_VOLUME_5R, +}; + +static const struct snd_compr_ops cs47l15_compr_ops = { + .open = &cs47l15_open, + .free = &wm_adsp_compr_free, + .set_params = &wm_adsp_compr_set_params, + .get_caps = &wm_adsp_compr_get_caps, + .trigger = &wm_adsp_compr_trigger, + .pointer = &wm_adsp_compr_pointer, + .copy = &wm_adsp_compr_copy, +}; + +static const struct snd_soc_component_driver soc_component_dev_cs47l15 = { + .probe = &cs47l15_component_probe, + .remove = &cs47l15_component_remove, + .set_sysclk = &madera_set_sysclk, + .set_pll = &cs47l15_set_fll, + .name = DRV_NAME, + .compr_ops = &cs47l15_compr_ops, + .controls = cs47l15_snd_controls, + .num_controls = ARRAY_SIZE(cs47l15_snd_controls), + .dapm_widgets = cs47l15_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs47l15_dapm_widgets), + .dapm_routes = cs47l15_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs47l15_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int cs47l15_probe(struct platform_device *pdev) +{ + struct madera *madera = dev_get_drvdata(pdev->dev.parent); + struct cs47l15 *cs47l15; + int i, ret; + + BUILD_BUG_ON(ARRAY_SIZE(cs47l15_dai) > MADERA_MAX_DAI); + + /* quick exit if Madera irqchip driver hasn't completed probe */ + if (!madera->irq_dev) { + dev_dbg(&pdev->dev, "irqchip driver not ready\n"); + return -EPROBE_DEFER; + } + + cs47l15 = devm_kzalloc(&pdev->dev, sizeof(struct cs47l15), + GFP_KERNEL); + if (!cs47l15) + return -ENOMEM; + + platform_set_drvdata(pdev, cs47l15); + + cs47l15->core.madera = madera; + cs47l15->core.dev = &pdev->dev; + cs47l15->core.num_inputs = 4; + + ret = madera_core_init(&cs47l15->core); + if (ret) + return ret; + + ret = madera_init_overheat(&cs47l15->core); + if (ret) + goto error_core; + + ret = madera_request_irq(madera, MADERA_IRQ_DSP_IRQ1, + "ADSP2 Compressed IRQ", cs47l15_adsp2_irq, + cs47l15); + if (ret != 0) { + dev_err(&pdev->dev, "Failed to request DSP IRQ: %d\n", ret); + goto error_overheat; + } + + ret = madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 1); + if (ret) + dev_warn(&pdev->dev, "Failed to set DSP IRQ wake: %d\n", ret); + + cs47l15->core.adsp[0].part = "cs47l15"; + cs47l15->core.adsp[0].num = 1; + cs47l15->core.adsp[0].type = WMFW_ADSP2; + cs47l15->core.adsp[0].rev = 2; + cs47l15->core.adsp[0].dev = madera->dev; + cs47l15->core.adsp[0].regmap = madera->regmap_32bit; + + cs47l15->core.adsp[0].base = MADERA_DSP1_CONFIG_1; + cs47l15->core.adsp[0].mem = cs47l15_dsp1_regions; + cs47l15->core.adsp[0].num_mems = ARRAY_SIZE(cs47l15_dsp1_regions); + + cs47l15->core.adsp[0].lock_regions = + WM_ADSP2_REGION_1 | WM_ADSP2_REGION_2 | WM_ADSP2_REGION_3; + + ret = wm_adsp2_init(&cs47l15->core.adsp[0]); + if (ret != 0) + goto error_dsp_irq; + + ret = madera_init_bus_error_irq(&cs47l15->core, 0, + cs47l15_dsp_bus_error); + if (ret) + goto error_adsp; + + madera_init_fll(madera, 1, MADERA_FLL1_CONTROL_1 - 1, + &cs47l15->fll[0]); + madera_init_fll(madera, 4, MADERA_FLLAO_CONTROL_1 - 1, + &cs47l15->fll[1]); + + for (i = 0; i < ARRAY_SIZE(cs47l15_dai); i++) + madera_init_dai(&cs47l15->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(cs47l15_digital_vu); i++) + regmap_update_bits(madera->regmap, cs47l15_digital_vu[i], + CS47L15_DIG_VU, CS47L15_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &soc_component_dev_cs47l15, + cs47l15_dai, + ARRAY_SIZE(cs47l15_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register component: %d\n", ret); + goto error_pm_runtime; + } + + return ret; + +error_pm_runtime: + pm_runtime_disable(&pdev->dev); + madera_free_bus_error_irq(&cs47l15->core, 0); +error_adsp: + wm_adsp2_remove(&cs47l15->core.adsp[0]); +error_dsp_irq: + madera_set_irq_wake(madera, MADERA_IRQ_DSP_IRQ1, 0); + madera_free_irq(madera, MADERA_IRQ_DSP_IRQ1, cs47l15); +error_overheat: + madera_free_overheat(&cs47l15->core); +error_core: + madera_core_free(&cs47l15->core); + + return ret; +} + +static int cs47l15_remove(struct platform_device *pdev) +{ + struct cs47l15 *cs47l15 = platform_get_drvdata(pdev); + + pm_runtime_disable(&pdev->dev); + + madera_free_bus_error_irq(&cs47l15->core, 0); + + wm_adsp2_remove(&cs47l15->core.adsp[0]); + + madera_set_irq_wake(cs47l15->core.madera, MADERA_IRQ_DSP_IRQ1, 0); + madera_free_irq(cs47l15->core.madera, MADERA_IRQ_DSP_IRQ1, cs47l15); + madera_free_overheat(&cs47l15->core); + madera_core_free(&cs47l15->core); + + return 0; +} + +static struct platform_driver cs47l15_codec_driver = { + .driver = { + .name = "cs47l15-codec", + }, + .probe = &cs47l15_probe, + .remove = &cs47l15_remove, +}; + +module_platform_driver(cs47l15_codec_driver); + +MODULE_SOFTDEP("pre: madera irq-madera arizona-micsupp"); +MODULE_DESCRIPTION("ASoC CS47L15 driver"); +MODULE_AUTHOR("Richard Fitzgerald <rf(a)opensource.cirrus.com>"); +MODULE_AUTHOR("Jaswinder Jassal <jjassal(a)opensource.cirrus.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cs47l15-codec"); diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 5f1e32a5a8556..1c00e2a115bcd 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -1041,6 +1041,10 @@ static void madera_configure_input_mode(struct madera *madera) int max_analogue_inputs, max_dmic_sup, i; switch (madera->type) { + case CS47L15: + max_analogue_inputs = 1; + max_dmic_sup = 2; + break; case CS47L35: max_analogue_inputs = 2; max_dmic_sup = 2; -- 2.11.0

1 2

[alsa-devel] [PATCH] ASoC: soc-pcm: Use format size instead physical for sample_bits symmetry
by Peter Ujfalusi 25 Jul '19

25 Jul '19

For DAIs the format size in bits matters when it comes to symmetry. The physical width is the in-memory format size which matters mostly for the platform drivers. If for example a DAI requests symmetric_samplebits and the first stream is using S24_LE then the second stream would not be allowed to use S24_3LE despite both S24_LE and S24_3LE would need the same clocking on the bus. But S32_LE would be allowed (S24_LE and S32_LE is both physically 32 bits in memory) which would not be compatible with S24_LE on the bus. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/soc-pcm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1406f935af22..c70722dc1f4f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -243,7 +243,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, rate = params_rate(params); channels = params_channels(params); - sample_bits = snd_pcm_format_physical_width(params_format(params)); + sample_bits = snd_pcm_format_width(params_format(params)); /* reject unmatched parameters when applying symmetry */ symmetry = cpu_dai->driver->symmetric_rates || @@ -932,7 +932,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->rate = params_rate(&codec_params); codec_dai->channels = params_channels(&codec_params); - codec_dai->sample_bits = snd_pcm_format_physical_width( + codec_dai->sample_bits = snd_pcm_format_width( params_format(&codec_params)); snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); @@ -945,8 +945,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); cpu_dai->channels = params_channels(params); - cpu_dai->sample_bits = - snd_pcm_format_physical_width(params_format(params)); + cpu_dai->sample_bits = snd_pcm_format_width(params_format(params)); snd_soc_dapm_update_dai(substream, params, cpu_dai); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 1

[alsa-devel] [PATCH] ALSA: hda - Add a conexant codec entry to let mute led work
by Hui Wang 25 Jul '19

25 Jul '19

This conexant codec isn't in the supported codec list yet, the hda generic driver can drive this codec well, but on a Lenovo machine with mute/mic-mute leds, we need to apply CXT_FIXUP_THINKPAD_ACPI to make the leds work. After adding this codec to the list, the driver patch_conexant.c will apply THINKPAD_ACPI to this machine. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f8d0845ee1e..f299f137eaea 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1083,6 +1083,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { + HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), -- 2.17.1

2 1

[alsa-devel] [PATCH 1/2] treewide: add "WITH Linux-syscall-note" to SPDX tag of uapi headers
by Masahiro Yamada 25 Jul '19

25 Jul '19

UAPI headers licensed under GPL are supposed to have exception "WITH Linux-syscall-note" so that they can be included into non-GPL user space application code. The exception note is missing in some UAPI headers. Some of them slipped in by the treewide conversion commit b24413180f56 ("License cleanup: add SPDX GPL-2.0 license identifier to files with no license"). Just run: $ git show --oneline b24413180f56 -- arch/x86/include/uapi/asm/ I believe they are not intentional, and should be fixed too. This patch was generated by the following script: git grep -l --not -e Linux-syscall-note --and -e SPDX-License-Identifier \ -- :arch/*/include/uapi/asm/*.h :include/uapi/ :^*/Kbuild | while read file do sed -i -e 's/$GPL-[^[:space:]]*$/\1 WITH Linux-syscall-note/g' $file done After this patch is applied, there are 5 UAPI headers that do not contain "WITH Linux-syscall-note". They are kept untouched since this exception applies only to GPL variants. $ git grep --not -e Linux-syscall-note --and -e SPDX-License-Identifier \ -- :arch/*/include/uapi/asm/*.h :include/uapi/ :^*/Kbuild include/uapi/drm/panfrost_drm.h:/* SPDX-License-Identifier: MIT */ include/uapi/linux/batman_adv.h:/* SPDX-License-Identifier: MIT */ include/uapi/linux/qemu_fw_cfg.h:/* SPDX-License-Identifier: BSD-3-Clause */ include/uapi/linux/vbox_err.h:/* SPDX-License-Identifier: MIT */ include/uapi/linux/virtio_iommu.h:/* SPDX-License-Identifier: BSD-3-Clause */ Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- arch/arm64/include/uapi/asm/bpf_perf_event.h | 2 +- arch/csky/include/uapi/asm/byteorder.h | 2 +- arch/csky/include/uapi/asm/cachectl.h | 2 +- arch/csky/include/uapi/asm/perf_regs.h | 2 +- arch/csky/include/uapi/asm/ptrace.h | 2 +- arch/csky/include/uapi/asm/sigcontext.h | 2 +- arch/csky/include/uapi/asm/unistd.h | 2 +- arch/nds32/include/uapi/asm/auxvec.h | 2 +- arch/nds32/include/uapi/asm/byteorder.h | 2 +- arch/nds32/include/uapi/asm/cachectl.h | 2 +- arch/nds32/include/uapi/asm/fp_udfiex_crtl.h | 2 +- arch/nds32/include/uapi/asm/param.h | 2 +- arch/nds32/include/uapi/asm/ptrace.h | 2 +- arch/nds32/include/uapi/asm/sigcontext.h | 2 +- arch/nds32/include/uapi/asm/unistd.h | 2 +- arch/powerpc/include/uapi/asm/bpf_perf_event.h | 2 +- arch/riscv/include/uapi/asm/auxvec.h | 2 +- arch/riscv/include/uapi/asm/bitsperlong.h | 2 +- arch/riscv/include/uapi/asm/byteorder.h | 2 +- arch/riscv/include/uapi/asm/hwcap.h | 2 +- arch/riscv/include/uapi/asm/ptrace.h | 2 +- arch/riscv/include/uapi/asm/sigcontext.h | 2 +- arch/riscv/include/uapi/asm/ucontext.h | 2 +- arch/s390/include/uapi/asm/bpf_perf_event.h | 2 +- arch/s390/include/uapi/asm/ipl.h | 2 +- arch/sh/include/uapi/asm/setup.h | 2 +- arch/sh/include/uapi/asm/types.h | 2 +- arch/sparc/include/uapi/asm/oradax.h | 2 +- arch/x86/include/uapi/asm/byteorder.h | 2 +- arch/x86/include/uapi/asm/hwcap2.h | 2 +- arch/x86/include/uapi/asm/sigcontext32.h | 2 +- arch/x86/include/uapi/asm/types.h | 2 +- include/uapi/linux/bpfilter.h | 2 +- include/uapi/linux/ipmi_bmc.h | 2 +- include/uapi/linux/isst_if.h | 2 +- include/uapi/linux/netfilter/nf_synproxy.h | 2 +- include/uapi/linux/psp-sev.h | 2 +- include/uapi/linux/rxrpc.h | 2 +- include/uapi/linux/usb/g_uvc.h | 2 +- include/uapi/linux/vbox_vmmdev_types.h | 2 +- include/uapi/linux/vboxguest.h | 2 +- include/uapi/linux/virtio_pmem.h | 2 +- include/uapi/linux/vmcore.h | 2 +- include/uapi/linux/wmi.h | 2 +- include/uapi/misc/fastrpc.h | 2 +- include/uapi/rdma/rvt-abi.h | 2 +- include/uapi/rdma/siw-abi.h | 2 +- include/uapi/scsi/scsi_bsg_ufs.h | 2 +- include/uapi/sound/skl-tplg-interface.h | 2 +- 49 files changed, 49 insertions(+), 49 deletions(-) diff --git a/arch/arm64/include/uapi/asm/bpf_perf_event.h b/arch/arm64/include/uapi/asm/bpf_perf_event.h index b551b741653d..5e1e648aeec4 100644 --- a/arch/arm64/include/uapi/asm/bpf_perf_event.h +++ b/arch/arm64/include/uapi/asm/bpf_perf_event.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI__ASM_BPF_PERF_EVENT_H__ #define _UAPI__ASM_BPF_PERF_EVENT_H__ diff --git a/arch/csky/include/uapi/asm/byteorder.h b/arch/csky/include/uapi/asm/byteorder.h index b079ec715cdf..d150cd664873 100644 --- a/arch/csky/include/uapi/asm/byteorder.h +++ b/arch/csky/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #ifndef __ASM_CSKY_BYTEORDER_H diff --git a/arch/csky/include/uapi/asm/cachectl.h b/arch/csky/include/uapi/asm/cachectl.h index ddf2f39aa925..ed7fad1ea20d 100644 --- a/arch/csky/include/uapi/asm/cachectl.h +++ b/arch/csky/include/uapi/asm/cachectl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef __ASM_CSKY_CACHECTL_H #define __ASM_CSKY_CACHECTL_H diff --git a/arch/csky/include/uapi/asm/perf_regs.h b/arch/csky/include/uapi/asm/perf_regs.h index ee323d818592..49d4e147a559 100644 --- a/arch/csky/include/uapi/asm/perf_regs.h +++ b/arch/csky/include/uapi/asm/perf_regs.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2019 Hangzhou C-SKY Microsystems co.,ltd. #ifndef _ASM_CSKY_PERF_REGS_H diff --git a/arch/csky/include/uapi/asm/ptrace.h b/arch/csky/include/uapi/asm/ptrace.h index 4e248d5b86ef..66b2268e324e 100644 --- a/arch/csky/include/uapi/asm/ptrace.h +++ b/arch/csky/include/uapi/asm/ptrace.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #ifndef _CSKY_PTRACE_H diff --git a/arch/csky/include/uapi/asm/sigcontext.h b/arch/csky/include/uapi/asm/sigcontext.h index e81e7ff11e36..670c020f2cb8 100644 --- a/arch/csky/include/uapi/asm/sigcontext.h +++ b/arch/csky/include/uapi/asm/sigcontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #ifndef __ASM_CSKY_SIGCONTEXT_H diff --git a/arch/csky/include/uapi/asm/unistd.h b/arch/csky/include/uapi/asm/unistd.h index ec60e49cea66..211c983c7282 100644 --- a/arch/csky/include/uapi/asm/unistd.h +++ b/arch/csky/include/uapi/asm/unistd.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2018 Hangzhou C-SKY Microsystems co.,ltd. #define __ARCH_WANT_SYS_CLONE diff --git a/arch/nds32/include/uapi/asm/auxvec.h b/arch/nds32/include/uapi/asm/auxvec.h index b5d58ea8decb..bc0b92ab8c15 100644 --- a/arch/nds32/include/uapi/asm/auxvec.h +++ b/arch/nds32/include/uapi/asm/auxvec.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __ASM_AUXVEC_H diff --git a/arch/nds32/include/uapi/asm/byteorder.h b/arch/nds32/include/uapi/asm/byteorder.h index 511e653c709d..c264ef12c49c 100644 --- a/arch/nds32/include/uapi/asm/byteorder.h +++ b/arch/nds32/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __NDS32_BYTEORDER_H__ diff --git a/arch/nds32/include/uapi/asm/cachectl.h b/arch/nds32/include/uapi/asm/cachectl.h index 73793662815c..31b9b439d819 100644 --- a/arch/nds32/include/uapi/asm/cachectl.h +++ b/arch/nds32/include/uapi/asm/cachectl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 1994, 1995, 1996 by Ralf Baechle // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef _ASM_CACHECTL diff --git a/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h b/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h index d54a5d6c6538..f17396db16ec 100644 --- a/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h +++ b/arch/nds32/include/uapi/asm/fp_udfiex_crtl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* Copyright (C) 2005-2019 Andes Technology Corporation */ #ifndef _FP_UDF_IEX_CRTL_H #define _FP_UDF_IEX_CRTL_H diff --git a/arch/nds32/include/uapi/asm/param.h b/arch/nds32/include/uapi/asm/param.h index 2977534a6bd3..48d00328d328 100644 --- a/arch/nds32/include/uapi/asm/param.h +++ b/arch/nds32/include/uapi/asm/param.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __ASM_NDS32_PARAM_H diff --git a/arch/nds32/include/uapi/asm/ptrace.h b/arch/nds32/include/uapi/asm/ptrace.h index 1a6e01c00e6f..d76217c7c010 100644 --- a/arch/nds32/include/uapi/asm/ptrace.h +++ b/arch/nds32/include/uapi/asm/ptrace.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef __UAPI_ASM_NDS32_PTRACE_H diff --git a/arch/nds32/include/uapi/asm/sigcontext.h b/arch/nds32/include/uapi/asm/sigcontext.h index dc89af7ddcc3..6c1e6648878f 100644 --- a/arch/nds32/include/uapi/asm/sigcontext.h +++ b/arch/nds32/include/uapi/asm/sigcontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #ifndef _ASMNDS32_SIGCONTEXT_H diff --git a/arch/nds32/include/uapi/asm/unistd.h b/arch/nds32/include/uapi/asm/unistd.h index a0b2f7b9c0f2..410795e280fe 100644 --- a/arch/nds32/include/uapi/asm/unistd.h +++ b/arch/nds32/include/uapi/asm/unistd.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ // Copyright (C) 2005-2017 Andes Technology Corporation #define __ARCH_WANT_STAT64 diff --git a/arch/powerpc/include/uapi/asm/bpf_perf_event.h b/arch/powerpc/include/uapi/asm/bpf_perf_event.h index b551b741653d..5e1e648aeec4 100644 --- a/arch/powerpc/include/uapi/asm/bpf_perf_event.h +++ b/arch/powerpc/include/uapi/asm/bpf_perf_event.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI__ASM_BPF_PERF_EVENT_H__ #define _UAPI__ASM_BPF_PERF_EVENT_H__ diff --git a/arch/riscv/include/uapi/asm/auxvec.h b/arch/riscv/include/uapi/asm/auxvec.h index 62716653554b..d86cb17bbabe 100644 --- a/arch/riscv/include/uapi/asm/auxvec.h +++ b/arch/riscv/include/uapi/asm/auxvec.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2015 Regents of the University of California diff --git a/arch/riscv/include/uapi/asm/bitsperlong.h b/arch/riscv/include/uapi/asm/bitsperlong.h index 0b9b58b57ff6..7d0b32e3b701 100644 --- a/arch/riscv/include/uapi/asm/bitsperlong.h +++ b/arch/riscv/include/uapi/asm/bitsperlong.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2015 Regents of the University of California diff --git a/arch/riscv/include/uapi/asm/byteorder.h b/arch/riscv/include/uapi/asm/byteorder.h index 1920debc09c0..f671e16bf6af 100644 --- a/arch/riscv/include/uapi/asm/byteorder.h +++ b/arch/riscv/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2015 Regents of the University of California diff --git a/arch/riscv/include/uapi/asm/hwcap.h b/arch/riscv/include/uapi/asm/hwcap.h index 7d786145183b..4e7646077056 100644 --- a/arch/riscv/include/uapi/asm/hwcap.h +++ b/arch/riscv/include/uapi/asm/hwcap.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copied from arch/arm64/include/asm/hwcap.h * diff --git a/arch/riscv/include/uapi/asm/ptrace.h b/arch/riscv/include/uapi/asm/ptrace.h index 92d8f7cd8f84..882547f6bd5c 100644 --- a/arch/riscv/include/uapi/asm/ptrace.h +++ b/arch/riscv/include/uapi/asm/ptrace.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 Regents of the University of California */ diff --git a/arch/riscv/include/uapi/asm/sigcontext.h b/arch/riscv/include/uapi/asm/sigcontext.h index 053f809e52ce..84f2dfcfdbce 100644 --- a/arch/riscv/include/uapi/asm/sigcontext.h +++ b/arch/riscv/include/uapi/asm/sigcontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 Regents of the University of California */ diff --git a/arch/riscv/include/uapi/asm/ucontext.h b/arch/riscv/include/uapi/asm/ucontext.h index b58e00cee2ec..411dd7b52ed6 100644 --- a/arch/riscv/include/uapi/asm/ucontext.h +++ b/arch/riscv/include/uapi/asm/ucontext.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Copyright (C) 2012 ARM Ltd. * Copyright (C) 2017 SiFive, Inc. diff --git a/arch/s390/include/uapi/asm/bpf_perf_event.h b/arch/s390/include/uapi/asm/bpf_perf_event.h index cefe7c7cd4f6..3ed42ff6da94 100644 --- a/arch/s390/include/uapi/asm/bpf_perf_event.h +++ b/arch/s390/include/uapi/asm/bpf_perf_event.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI__ASM_BPF_PERF_EVENT_H__ #define _UAPI__ASM_BPF_PERF_EVENT_H__ diff --git a/arch/s390/include/uapi/asm/ipl.h b/arch/s390/include/uapi/asm/ipl.h index fd32b1cd80d2..451ba7d08905 100644 --- a/arch/s390/include/uapi/asm/ipl.h +++ b/arch/s390/include/uapi/asm/ipl.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_S390_UAPI_IPL_H #define _ASM_S390_UAPI_IPL_H diff --git a/arch/sh/include/uapi/asm/setup.h b/arch/sh/include/uapi/asm/setup.h index 1170dd2fb998..4bd19f80f9b0 100644 --- a/arch/sh/include/uapi/asm/setup.h +++ b/arch/sh/include/uapi/asm/setup.h @@ -1,2 +1,2 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #include <asm-generic/setup.h> diff --git a/arch/sh/include/uapi/asm/types.h b/arch/sh/include/uapi/asm/types.h index f83795fdc0da..68100e108ea6 100644 --- a/arch/sh/include/uapi/asm/types.h +++ b/arch/sh/include/uapi/asm/types.h @@ -1,2 +1,2 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #include <asm-generic/types.h> diff --git a/arch/sparc/include/uapi/asm/oradax.h b/arch/sparc/include/uapi/asm/oradax.h index 64c67f2ea33f..0dace69058ab 100644 --- a/arch/sparc/include/uapi/asm/oradax.h +++ b/arch/sparc/include/uapi/asm/oradax.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-or-later */ +/* SPDX-License-Identifier: GPL-2.0-or-later WITH Linux-syscall-note */ /* * Copyright (c) 2017, Oracle and/or its affiliates. All rights reserved. */ diff --git a/arch/x86/include/uapi/asm/byteorder.h b/arch/x86/include/uapi/asm/byteorder.h index 484e3cfd7ef2..149143cab9ff 100644 --- a/arch/x86/include/uapi/asm/byteorder.h +++ b/arch/x86/include/uapi/asm/byteorder.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_BYTEORDER_H #define _ASM_X86_BYTEORDER_H diff --git a/arch/x86/include/uapi/asm/hwcap2.h b/arch/x86/include/uapi/asm/hwcap2.h index 6ebaae90e207..8b2effe6efb8 100644 --- a/arch/x86/include/uapi/asm/hwcap2.h +++ b/arch/x86/include/uapi/asm/hwcap2.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_HWCAP2_H #define _ASM_X86_HWCAP2_H diff --git a/arch/x86/include/uapi/asm/sigcontext32.h b/arch/x86/include/uapi/asm/sigcontext32.h index 6b18e88de8a6..7114801d0499 100644 --- a/arch/x86/include/uapi/asm/sigcontext32.h +++ b/arch/x86/include/uapi/asm/sigcontext32.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_SIGCONTEXT32_H #define _ASM_X86_SIGCONTEXT32_H diff --git a/arch/x86/include/uapi/asm/types.h b/arch/x86/include/uapi/asm/types.h index df55e1ddb0c9..9d5c11a24279 100644 --- a/arch/x86/include/uapi/asm/types.h +++ b/arch/x86/include/uapi/asm/types.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _ASM_X86_TYPES_H #define _ASM_X86_TYPES_H diff --git a/include/uapi/linux/bpfilter.h b/include/uapi/linux/bpfilter.h index 2ec3cc99ea4c..cbc1f5813f50 100644 --- a/include/uapi/linux/bpfilter.h +++ b/include/uapi/linux/bpfilter.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI_LINUX_BPFILTER_H #define _UAPI_LINUX_BPFILTER_H diff --git a/include/uapi/linux/ipmi_bmc.h b/include/uapi/linux/ipmi_bmc.h index 1670f0944227..782a03eb1086 100644 --- a/include/uapi/linux/ipmi_bmc.h +++ b/include/uapi/linux/ipmi_bmc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * Copyright (c) 2015-2018, Intel Corporation. */ diff --git a/include/uapi/linux/isst_if.h b/include/uapi/linux/isst_if.h index d10b832c58c5..0a52b7b093d3 100644 --- a/include/uapi/linux/isst_if.h +++ b/include/uapi/linux/isst_if.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * Intel Speed Select Interface: OS to hardware Interface * Copyright (c) 2019, Intel Corporation. diff --git a/include/uapi/linux/netfilter/nf_synproxy.h b/include/uapi/linux/netfilter/nf_synproxy.h index 6f3791c8946f..00d787f0260e 100644 --- a/include/uapi/linux/netfilter/nf_synproxy.h +++ b/include/uapi/linux/netfilter/nf_synproxy.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _NF_SYNPROXY_H #define _NF_SYNPROXY_H diff --git a/include/uapi/linux/psp-sev.h b/include/uapi/linux/psp-sev.h index 8654b2442f6a..592a0c1b77c9 100644 --- a/include/uapi/linux/psp-sev.h +++ b/include/uapi/linux/psp-sev.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * Userspace interface for AMD Secure Encrypted Virtualization (SEV) * platform management commands. diff --git a/include/uapi/linux/rxrpc.h b/include/uapi/linux/rxrpc.h index 782069dcf607..4accfa7e266d 100644 --- a/include/uapi/linux/rxrpc.h +++ b/include/uapi/linux/rxrpc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-or-later */ +/* SPDX-License-Identifier: GPL-2.0-or-later WITH Linux-syscall-note */ /* Types and definitions for AF_RXRPC. * * Copyright (C) 2007 Red Hat, Inc. All Rights Reserved. diff --git a/include/uapi/linux/usb/g_uvc.h b/include/uapi/linux/usb/g_uvc.h index 3c9ee3020cbb..652f169a019e 100644 --- a/include/uapi/linux/usb/g_uvc.h +++ b/include/uapi/linux/usb/g_uvc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0+ */ +/* SPDX-License-Identifier: GPL-2.0+ WITH Linux-syscall-note */ /* * g_uvc.h -- USB Video Class Gadget driver API * diff --git a/include/uapi/linux/vbox_vmmdev_types.h b/include/uapi/linux/vbox_vmmdev_types.h index 26f39816af14..83c3aa649b08 100644 --- a/include/uapi/linux/vbox_vmmdev_types.h +++ b/include/uapi/linux/vbox_vmmdev_types.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR CDDL-1.0) */ +/* SPDX-License-Identifier: (GPL-2.0 WITH Linux-syscall-note OR CDDL-1.0) */ /* * Virtual Device for Guest <-> VMM/Host communication, type definitions * which are also used for the vboxguest ioctl interface / by vboxsf diff --git a/include/uapi/linux/vboxguest.h b/include/uapi/linux/vboxguest.h index 612f0c7d3558..17194495b648 100644 --- a/include/uapi/linux/vboxguest.h +++ b/include/uapi/linux/vboxguest.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR CDDL-1.0) */ +/* SPDX-License-Identifier: (GPL-2.0 WITH Linux-syscall-note OR CDDL-1.0) */ /* * VBoxGuest - VirtualBox Guest Additions Driver Interface. * diff --git a/include/uapi/linux/virtio_pmem.h b/include/uapi/linux/virtio_pmem.h index 9a63ed6d062f..c7a82f347e84 100644 --- a/include/uapi/linux/virtio_pmem.h +++ b/include/uapi/linux/virtio_pmem.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 OR BSD-3-Clause */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note OR BSD-3-Clause */ /* * Definitions for virtio-pmem devices. * diff --git a/include/uapi/linux/vmcore.h b/include/uapi/linux/vmcore.h index 022619668e0e..3e9da91866ff 100644 --- a/include/uapi/linux/vmcore.h +++ b/include/uapi/linux/vmcore.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef _UAPI_VMCORE_H #define _UAPI_VMCORE_H diff --git a/include/uapi/linux/wmi.h b/include/uapi/linux/wmi.h index c36f2d7675a4..7085c5dca9fa 100644 --- a/include/uapi/linux/wmi.h +++ b/include/uapi/linux/wmi.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ +/* SPDX-License-Identifier: GPL-2.0-only WITH Linux-syscall-note */ /* * User API methods for ACPI-WMI mapping driver * diff --git a/include/uapi/misc/fastrpc.h b/include/uapi/misc/fastrpc.h index 6d701af9fc42..fb792e882cef 100644 --- a/include/uapi/misc/fastrpc.h +++ b/include/uapi/misc/fastrpc.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ #ifndef __QCOM_FASTRPC_H__ #define __QCOM_FASTRPC_H__ diff --git a/include/uapi/rdma/rvt-abi.h b/include/uapi/rdma/rvt-abi.h index 7328293c715c..81c89b322f1f 100644 --- a/include/uapi/rdma/rvt-abi.h +++ b/include/uapi/rdma/rvt-abi.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* SPDX-License-Identifier: (GPL-2.0 WITH Linux-syscall-note OR BSD-3-Clause) */ /* * This file contains defines, structures, etc. that are used diff --git a/include/uapi/rdma/siw-abi.h b/include/uapi/rdma/siw-abi.h index 3dd8071ace7b..2cc2331cb254 100644 --- a/include/uapi/rdma/siw-abi.h +++ b/include/uapi/rdma/siw-abi.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 or BSD-3-Clause */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note or BSD-3-Clause */ /* Authors: Bernard Metzler <bmt(a)zurich.ibm.com> */ /* Copyright (c) 2008-2019, IBM Corporation */ diff --git a/include/uapi/scsi/scsi_bsg_ufs.h b/include/uapi/scsi/scsi_bsg_ufs.h index 17c7abd0803a..9988db6ad244 100644 --- a/include/uapi/scsi/scsi_bsg_ufs.h +++ b/include/uapi/scsi/scsi_bsg_ufs.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * UFS Transport SGIO v4 BSG Message Support * diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index f39352cef382..9eee32f5e407 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -1,4 +1,4 @@ -/* SPDX-License-Identifier: GPL-2.0 */ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * skl-tplg-interface.h - Intel DSP FW private data interface * -- 2.17.1

1 2

[alsa-devel] [PATCH 5.1 201/371] ASoC: audio-graph-card: fix use-after-free in graph_for_each_link
by Greg Kroah-Hartman 24 Jul '19

24 Jul '19

[ Upstream commit 1bcc1fd64e4dd903f4d868a9e053986e3b102713 ] After calling of_node_put() on the codec_ep and codec_port variables, they are still being used, which may result in use-after-free. We fix this issue by calling of_node_put() after the last usage. Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562229530-8121-1-git-send-email-wen.yang99@zte.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 69bc4848d787..f730830fb36c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -460,9 +460,6 @@ static int graph_for_each_link(struct graph_priv *priv, codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_get_parent(codec_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); graph_get_conversion(dev, codec_ep, &adata); @@ -482,6 +479,9 @@ static int graph_for_each_link(struct graph_priv *priv, else ret = func_noml(priv, cpu_ep, codec_ep, li); + of_node_put(codec_ep); + of_node_put(codec_port); + if (ret < 0) return ret; -- 2.20.1

1 0

[alsa-devel] [PATCH 5.2 224/413] ASoC: audio-graph-card: fix use-after-free in graph_for_each_link
by Greg Kroah-Hartman 24 Jul '19

24 Jul '19

[ Upstream commit 1bcc1fd64e4dd903f4d868a9e053986e3b102713 ] After calling of_node_put() on the codec_ep and codec_port variables, they are still being used, which may result in use-after-free. We fix this issue by calling of_node_put() after the last usage. Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562229530-8121-1-git-send-email-wen.yang99@zte.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index ec7e673ba475..70ed28d97d49 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -435,9 +435,6 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_get_parent(codec_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); graph_parse_convert(dev, codec_ep, &adata); @@ -457,6 +454,9 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, else ret = func_noml(priv, cpu_ep, codec_ep, li); + of_node_put(codec_ep); + of_node_put(codec_port); + if (ret < 0) return ret; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: wm8955: Fix a typo in 'wm8995_pll_factors()' function name
by Christophe JAILLET 24 Jul '19

24 Jul '19

This should be 'wm8955_pll_factors()' instead. Fix it and use it. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/wm8955.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index cd204f79647d..ec82a8fafdf6 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -143,7 +143,7 @@ struct pll_factors { * to allow rounding later */ #define FIXED_FLL_SIZE ((1 << 22) * 10) -static int wm8995_pll_factors(struct device *dev, +static int wm8955_pll_factors(struct device *dev, int Fref, int Fout, struct pll_factors *pll) { u64 Kpart; @@ -282,7 +282,7 @@ static int wm8955_configure_clocking(struct snd_soc_component *component) /* Use the last divider configuration we saw for the * sample rate. */ - ret = wm8995_pll_factors(component->dev, wm8955->mclk_rate, + ret = wm8955_pll_factors(component->dev, wm8955->mclk_rate, clock_cfgs[sr].mclk, &pll); if (ret != 0) { dev_err(component->dev, -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: Fail card instantiation if DAI format setup fails
by Ricard Wanderlof 24 Jul '19

24 Jul '19

If the DAI format setup fails, there is no valid communication format between CPU and CODEC, so fail card instantiation, rather than continue with a card that will most likely not function properly. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- I've had the problem of a sound card coming up even though an incorrect DAI format was selected in the devicetree, apparently caused by not checking the return value of snd_soc_runtime_set_dai_fmt() when the card is instantiated, with the result in this particular case that that the audio data was not transferred correctly (capture data was shifted one bit up). I'm unsure though if this is intentional, I can't think why it should be, since the DAI format is only set up at this time and not for instance later when a stream is enabled, but perhaps I'm missing something here. At any rate, with this patch, the card does not come up under these circumstances. The error message from snd_soc_runtime_set_dai_fmt() is still printed to the kernel log so it's possible to see what the reason is. /Ricard sound/soc/soc-core.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ecfe641ca46..06697b2d96b1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1511,8 +1511,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card, } } - if (dai_link->dai_fmt) - snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); + if (dai_link->dai_fmt) { + ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); + if (ret) + return ret; + } ret = soc_post_component_init(rtd, dai_link->name); if (ret) -- 2.11.0 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips
by Takashi Iwai 24 Jul '19

24 Jul '19

It turned out that the recent Intel HD-audio controller chips show a significant stall during the system PM resume intermittently. It doesn't happen so often and usually it may read back successfully after one or more seconds, but in some rare worst cases the driver went into fallback mode. After trial-and-error, we found out that the communication stall seems covered by issuing the sync after each verb write, as already done for AMD and other chipsets. So this patch enables the write-sync flag for the recent Intel chips, Skylake and onward, as a workaround. Also, since Broxton and co have the very same driver flags as Skylake, refer to the Skylake driver flags instead of defining the same contents again for simplification. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201901 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index cb8b0945547c..1e14d7270adf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -313,11 +313,10 @@ enum { #define AZX_DCAPS_INTEL_SKYLAKE \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SYNC_WRITE |\ AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) -#define AZX_DCAPS_INTEL_BROXTON \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) +#define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ -- 2.16.4

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[alsa-devel] Reminder: 1 open syzbot bug in sound subsystem
by Eric Biggers 24 Jul '19

24 Jul '19

[This email was generated by a script. Let me know if you have any suggestions to make it better, or if you want it re-generated with the latest status.] Of the currently open syzbot reports against the upstream kernel, I've manually marked 1 of them as possibly being a bug in the sound subsystem. If you believe this bug is no longer valid, please close the syzbot report by sending a '#syz fix', '#syz dup', or '#syz invalid' command in reply to the original thread, as explained at https://goo.gl/tpsmEJ#status If you believe I misattributed this bug to the sound subsystem, please let me know, and if possible forward the report to the correct people or mailing list. Here is the bug: -------------------------------------------------------------------------------- Title: KASAN: use-after-free Read in wake_up_if_idle Last occurred: 131 days ago Reported: 267 days ago Branches: Mainline and others Dashboard link: https://syzkaller.appspot.com/bug?id=b1e300cd7b124fc83dd4199d4d1df26310111b… Original thread: https://lkml.kernel.org/lkml/00000000000066ab7105795f245e@google.com/T/#u Unfortunately, this bug does not have a reproducer. No one replied to the original thread for this bug. If you fix this bug, please add the following tag to the commit: Reported-by: syzbot+2c1253bc508adef78a7f(a)syzkaller.appspotmail.com If you send any email or patch for this bug, please consider replying to the original thread. For the git send-email command to use, or tips on how to reply if the thread isn't in your mailbox, see the "Reply instructions" at https://lkml.kernel.org/r/00000000000066ab7105795f245e@google.com

2 2

[alsa-devel] [PATCH 0/2 v2] Open Alsa dshare without channel binding
by Adam Miartus 24 Jul '19

24 Jul '19

The general idea of this patch is to be able to open the device without defined bindings. As a result, it allows starting the audio clock of underlying device without blocking a channel. This might be useful for creating a dummy dsnoop pcm for hardware that requires running clock early at system start. This clock is then present even without application streaming audio data. Patch n.1 is intended to check for null pointer of pcm callbacks, it is not intended to alter any current functionality, properly defined plugins should not suffer from this change Patch n.2 is not intended to alter any current functionality, only extend it V2 of the patch, changes from V1: - instead of creating new "pcm_unsupported.c" with ops and fast_ops callbacks returning -ENOSYS, callback == NULL is checked directly in core implementation and error is returned when pcm has no callback defined Adam Miartus (2): alsa: pcm: return -ENOSYS when ops or fast_ops callback is NULL alsa: dshare: allow missing bindings src/pcm/pcm.c | 134 +++++++++++++++++++++++++++++++++++++++++---------- src/pcm/pcm_dshare.c | 29 ++++++----- src/pcm/pcm_local.h | 20 ++++++++ src/pcm/pcm_mmap.c | 15 ++++-- src/pcm/pcm_params.c | 10 +++- 5 files changed, 166 insertions(+), 42 deletions(-) -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Use dev_get_drvdata where possible
by Chuhong Yuan 24 Jul '19

24 Jul '19

Instead of using to_pci_dev + pci_get_drvdata, use dev_get_drvdata to make code simpler. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/intel/skylake/skl-nhlt.c | 3 +-- sound/soc/intel/skylake/skl.c | 12 ++++-------- 2 files changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 1132109cb992..46d9977224d1 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -241,8 +241,7 @@ int skl_nhlt_update_topology_bin(struct skl *skl) static ssize_t skl_nhlt_platform_id_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); + struct hdac_bus *bus = dev_get_drvdata(dev); struct skl *skl = bus_to_skl(bus); struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; char platform_id[32]; diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 3362e71b4563..1f625fc65307 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -141,8 +141,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) void skl_update_d0i3c(struct device *dev, bool enable) { - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); + struct hdac_bus *bus = dev_get_drvdata(dev); u8 reg; int timeout = 50; @@ -274,8 +273,7 @@ static int skl_acquire_irq(struct hdac_bus *bus, int do_disconnect) static int skl_suspend_late(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); + struct hdac_bus *bus = dev_get_drvdata(dev); struct skl *skl = bus_to_skl(bus); return skl_suspend_late_dsp(skl); @@ -400,8 +398,7 @@ static int skl_resume(struct device *dev) #ifdef CONFIG_PM static int skl_runtime_suspend(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); + struct hdac_bus *bus = dev_get_drvdata(dev); dev_dbg(bus->dev, "in %s\n", __func__); @@ -410,8 +407,7 @@ static int skl_runtime_suspend(struct device *dev) static int skl_runtime_resume(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); + struct hdac_bus *bus = dev_get_drvdata(dev); dev_dbg(bus->dev, "in %s\n", __func__); -- 2.20.1

1 0

[alsa-devel] [PATCH] sound: line6: sizeof (byte) is always 1, use that fact.
by Pavel Machek 24 Jul '19

24 Jul '19

sizeof (byte) is always 1, use that fact and make interesting code explicit. Signed-off-by: Pavel Machek <pavel(a)denx.de> diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index ab2ec89..387c5c2 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -342,7 +342,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, if (address > 0xffff || datalen > 0xff) return -EINVAL; - len = kmalloc(sizeof(*len), GFP_KERNEL); + len = kmalloc(1, GFP_KERNEL); if (!len) return -ENOMEM; @@ -418,7 +421,7 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, if (address > 0xffff || datalen > 0xffff) return -EINVAL; - status = kmalloc(sizeof(*status), GFP_KERNEL); + status = kmalloc(1, GFP_KERNEL); if (!status) return -ENOMEM; -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

2 1

[alsa-devel] [PATCH] ALSA: core: Fix possible null-pointer dereferences in snd_timer_proc_read()
by Jia-Ju Bai 24 Jul '19

24 Jul '19

In snd_timer_proc_read(), there is an if statement on line 1204 to check whether timer->card is NULL: if (timer->card && timer->card->shutdown) When timer->card is NULL, it is used on lines 1212 and 1215: timer->card->number Thus, possible null-pointer dereferences may occur. To fix these bugs, timer->card is checked before being used. These bugs are found by a static analysis tool STCheck written by us. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/core/timer.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 5c9fbf3f4340..967d06a3cdec 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1208,11 +1208,13 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "G%i: ", timer->tmr_device); break; case SNDRV_TIMER_CLASS_CARD: - snd_iprintf(buffer, "C%i-%i: ", - timer->card->number, timer->tmr_device); + snd_iprintf(buffer, "C%i-%i: ", + timer->card ? timer->card->number : -1, + timer->tmr_device); break; case SNDRV_TIMER_CLASS_PCM: - snd_iprintf(buffer, "P%i-%i-%i: ", timer->card->number, + snd_iprintf(buffer, "P%i-%i-%i: ", + timer->card ? timer->card->number : -1, timer->tmr_device, timer->tmr_subdevice); break; default: -- 2.17.0

2 1

[alsa-devel] [PATCH 00/16] ASoC: add soc-dai.c
by Kuninori Morimoto 23 Jul '19

23 Jul '19

Hi Mark I want to cleanup ALSA SoC. It will do nothing from "technical" point of view, but, will makes code readable. I already have many cleanup patches in my local branch. This is first patch-set for it. Kuninori Morimoto (16): ASoC: add soc-dai.c ASoC: soc-dai: mv soc_dai_hw_params() to soc-dai ASoC: soc-dai: add snd_soc_dai_hw_free() ASoC: soc-dai: add snd_soc_dai_startup() ASoC: soc-dai: add snd_soc_dai_shutdown() ASoC: soc-dai: add snd_soc_dai_prepare() ASoC: soc-dai: add snd_soc_dai_trigger() ASoC: soc-dai: add snd_soc_dai_bespoke_trigger() ASoC: soc-dai: add snd_soc_dai_delay() ASoC: soc-dai: add snd_soc_dai_suspend() ASoC: soc-dai: add snd_soc_dai_resume() ASoC: soc-dai: add snd_soc_dai_probe() ASoC: soc-dai: add snd_soc_dai_remove() ASoC: soc-dai: add snd_soc_dai_compress_new() ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c ASoC: soc-core: use existing snd_soc_dai_digital_mute() include/sound/soc-dai.h | 25 +++ include/sound/soc.h | 4 - sound/soc/Makefile | 2 +- sound/soc/soc-compress.c | 9 +- sound/soc/soc-core.c | 316 ++++-------------------------------- sound/soc/soc-dai.c | 407 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-dapm.c | 46 ++---- sound/soc/soc-pcm.c | 200 +++++++---------------- 8 files changed, 548 insertions(+), 461 deletions(-) create mode 100644 sound/soc/soc-dai.c -- 2.7.4

2 33

[alsa-devel] [PATCH 0/5] ASoC: Intel: IPC framework updates
by Cezary Rojewski 23 Jul '19

23 Jul '19

Existing IPC framework omits crucial part of the entire communication: reply header. Some IPCs cannot function at all without the access to said header. Update the sst-ipc with new sst_ipc_message structure to represent both request and reply allowing reply-processing handlers to save received responses. Despite the range of changes required for model to be updated, no functional changes are made for core hanswell, baytrail and skylake message handlers. Reply-processing handlers now save received response header yet no usage is added by default. To allow for future changes, righful kings of IPC kingdom need to be put back on the throne. This update addresses one of them: LARGE_CONFIG_GET. Cezary Rojewski (5): ASoC: Intel: Update request-reply IPC model ASoC: Intel: Haswell: Align with updated request-reply model ASoC: Intel: Baytrail: Align with updated request-reply model ASoC: Intel: Skylake: Align with updated request-reply model ASoC: Intel: Skylake: large_config_get overhaul sound/soc/intel/baytrail/sst-baytrail-ipc.c | 65 ++++---- sound/soc/intel/common/sst-ipc.c | 68 ++++---- sound/soc/intel/common/sst-ipc.h | 27 ++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 164 +++++++++++--------- sound/soc/intel/skylake/cnl-sst.c | 6 +- sound/soc/intel/skylake/skl-messages.c | 3 +- sound/soc/intel/skylake/skl-sst-ipc.c | 152 ++++++++++-------- sound/soc/intel/skylake/skl-sst-ipc.h | 3 +- 8 files changed, 259 insertions(+), 229 deletions(-) -- 2.17.1

3 12

[alsa-devel] [PATCH 00/21] ASoC: SOF: updates for 5.4
by Pierre-Louis Bossart 23 Jul '19

23 Jul '19

Core updates for runtime-pm, loader extensions for i.MX support, increased timeouts for QEMU tests and state machine fixes to make PulseAudio happy. I included in the same set Intel HDaudio patches, mostly about code refactoring, jack detection and robustness improvements for IPC/DMA handling. If Mark/Takashi prefer separate patchsets for core and Intel stuff I can split the two. This is a rather large set mostly due to Summer vacations and a peak of product validation earlier in Q2, I expect more frequent/smaller updates moving forward. We also have quite a few SoundWire-related patches but they aren't strictly related to SOF. Comments and feedback welcome! Daniel Baluta (3): ASoC: SOF: Introduce snd_sof_dsp_get_bar_index ops ASoC: SOF: loader: Use the BAR provided by FW ASoC: SOF: loader: Don't ignore SRAM block types Fred Oh (1): ASoC: SOF: remove unused state variable in suspend function Guennadi Liakhovetski (1): ASoC: SOF: Intel: hda: add a parameter to disable MSI Janusz Jankowski (1): ASoC: SOF: Intel: ssp: BCLK delay parameter Kai Vehmanen (6): ASoC: SOF: reset DMA state in prepare ASoC: SOF: use common code to send PCM_FREE IPC ASoC: SOF: ipc: use timeout configured at probe ASoC: SOF: core: increase default IPC timeouts ASoC: SOF: Intel: hda: reset link DMA state in prepare ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread Keyon Jie (1): ASoC: SOF: Intel: hda: correct ROM state mask Pan Xiuli (1): ASoC: SOF: pci: mark last_busy value at runtime PM init Rander Wang (4): ASoC: SOF: Intel: hda: reduce ifdef usage for hda ASoC: SOF: Intel: hda: Enable jack detection ASoC: SOF: Intel: hda: fix link DMA config ASoC: SOF: Intel: hda: fix stream id setting Zhu Yingjiang (3): ASoC: SOF: Intel: hda: set position buffer in init chip ASoC: SOF: Intel: hda: use SOF defined init chip in resume ASoC: SOF: Intel: hda: remove duplicated clear WAKESTS include/sound/sof/dai-intel.h | 3 + include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/sof/core.c | 4 +- sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-codec.c | 44 +++++++++++++- sound/soc/sof/intel/hda-ctrl.c | 21 ++++--- sound/soc/sof/intel/hda-dai.c | 38 +++++++----- sound/soc/sof/intel/hda-dsp.c | 100 +++++++++++--------------------- sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda.c | 28 +++++---- sound/soc/sof/intel/hda.h | 9 +-- sound/soc/sof/ipc.c | 8 +-- sound/soc/sof/loader.c | 14 ++++- sound/soc/sof/ops.h | 28 +++++++-- sound/soc/sof/pcm.c | 63 ++++++++++---------- sound/soc/sof/pm.c | 6 +- sound/soc/sof/sof-pci-dev.c | 3 + sound/soc/sof/sof-priv.h | 12 ++-- sound/soc/sof/topology.c | 3 + 20 files changed, 229 insertions(+), 166 deletions(-) -- 2.20.1

2 45

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 23 Jul '19

23 Jul '19

When DSP boots with timeout error try to reload firmware to keep best user expeirience. Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 440bca7afbf1..dd0260557530 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -455,13 +455,19 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + dev_warn(ctx->dev,"Reloading FW\n"); + + ret = bxt_sst_init_fw(skl->dev, skl); + if (ret < 0) { + dev_err(ctx->dev, "Reload fw failed: %d\n", ret); + dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); + ret = -EIO; + goto err; + } } } @@ -486,7 +492,6 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) if (core_id == SKL_DSP_CORE0_ID) core_mask |= SKL_DSP_CORE_MASK(1); skl_dsp_disable_core(ctx, core_mask); - return ret; } -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ASoC: madera: Read device tree configuration
by Charles Keepax 23 Jul '19

23 Jul '19

Read the configuration of the Madera ASoC driver from device tree. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Returned to using a helping in the madera driver itself, discussions around adding a generic helper indicated that all the error reporting would still need to be contained in the driver which constitutes the vast majority of the code in the helper anyway. - Updated the code to use the new device_property_count_u32 helper. Thanks, Charles sound/soc/codecs/madera.c | 97 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 97 insertions(+) diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 1b1be19a2f991..5f1e32a5a8556 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -300,6 +300,100 @@ int madera_free_overheat(struct madera_priv *priv) } EXPORT_SYMBOL_GPL(madera_free_overheat); +static int madera_get_variable_u32_array(struct device *dev, + const char *propname, + u32 *dest, int n_max, + int multiple) +{ + int n, ret; + + n = device_property_count_u32(dev, propname); + if (n < 0) { + if (n == -EINVAL) + return 0; /* missing, ignore */ + + dev_warn(dev, "%s malformed (%d)\n", propname, n); + + return n; + } else if ((n % multiple) != 0) { + dev_warn(dev, "%s not a multiple of %d entries\n", + propname, multiple); + + return -EINVAL; + } + + if (n > n_max) + n = n_max; + + ret = device_property_read_u32_array(dev, propname, dest, n); + if (ret < 0) + return ret; + + return n; +} + +static void madera_prop_get_inmode(struct madera_priv *priv) +{ + struct madera *madera = priv->madera; + struct madera_codec_pdata *pdata = &madera->pdata.codec; + u32 tmp[MADERA_MAX_INPUT * MADERA_MAX_MUXED_CHANNELS]; + int n, i, in_idx, ch_idx; + + BUILD_BUG_ON(ARRAY_SIZE(pdata->inmode) != MADERA_MAX_INPUT); + BUILD_BUG_ON(ARRAY_SIZE(pdata->inmode[0]) != MADERA_MAX_MUXED_CHANNELS); + + n = madera_get_variable_u32_array(madera->dev, "cirrus,inmode", + tmp, ARRAY_SIZE(tmp), + MADERA_MAX_MUXED_CHANNELS); + if (n < 0) + return; + + in_idx = 0; + ch_idx = 0; + for (i = 0; i < n; ++i) { + pdata->inmode[in_idx][ch_idx] = tmp[i]; + + if (++ch_idx == MADERA_MAX_MUXED_CHANNELS) { + ch_idx = 0; + ++in_idx; + } + } +} + +static void madera_prop_get_pdata(struct madera_priv *priv) +{ + struct madera *madera = priv->madera; + struct madera_codec_pdata *pdata = &madera->pdata.codec; + u32 out_mono[ARRAY_SIZE(pdata->out_mono)]; + int i, n; + + madera_prop_get_inmode(priv); + + n = madera_get_variable_u32_array(madera->dev, "cirrus,out-mono", + out_mono, ARRAY_SIZE(out_mono), 1); + if (n > 0) + for (i = 0; i < n; ++i) + pdata->out_mono[i] = !!out_mono[i]; + + madera_get_variable_u32_array(madera->dev, + "cirrus,max-channels-clocked", + pdata->max_channels_clocked, + ARRAY_SIZE(pdata->max_channels_clocked), + 1); + + madera_get_variable_u32_array(madera->dev, "cirrus,pdm-fmt", + pdata->pdm_fmt, + ARRAY_SIZE(pdata->pdm_fmt), 1); + + madera_get_variable_u32_array(madera->dev, "cirrus,pdm-mute", + pdata->pdm_mute, + ARRAY_SIZE(pdata->pdm_mute), 1); + + madera_get_variable_u32_array(madera->dev, "cirrus,dmic-ref", + pdata->dmic_ref, + ARRAY_SIZE(pdata->dmic_ref), 1); +} + int madera_core_init(struct madera_priv *priv) { int i; @@ -308,6 +402,9 @@ int madera_core_init(struct madera_priv *priv) BUILD_BUG_ON(!madera_mixer_texts[MADERA_NUM_MIXER_INPUTS - 1]); BUILD_BUG_ON(!madera_mixer_values[MADERA_NUM_MIXER_INPUTS - 1]); + if (!dev_get_platdata(priv->madera->dev)) + madera_prop_get_pdata(priv); + mutex_init(&priv->rate_lock); for (i = 0; i < MADERA_MAX_HP_OUTPUT; i++) -- 2.11.0

3 7

[alsa-devel] [PATCH 0/2] wm8904: adapt driver for use with audio-graph-card
by Michał Mirosław 23 Jul '19

23 Jul '19

This series allows to use WM8904 codec as audio-graph-card component. It is a rewrite of an earlier approach from https://patchwork.kernel.org/patch/10738097/ Series based on tiwai/sound/for-next tree. You can also pull from: https://rere.qmqm.pl/git/linux branch: wm8904 Example DTS: / { sound { compatible = "audio-graph-card"; widgets = "Speaker", "Loudspeaker", "Headphone", "Headphone Jack", "Microphone", "Internal Mic", routing = "Headphone Jack", "HPOUTL", "Headphone Jack", "HPOUTR", "Amplifier", "LINEOUTL", "Amplifier", "LINEOUTR", "Loudspeaker", "Amplifier", "IN1L", "MICBIAS", "IN1L", "Internal Mic"; dais = <&ssc0_port>; }; }; &i2c1 { status = "okay"; wm8904: acodec@1a { compatible = "wlf,wm8904"; reg = <0x1a>; clocks = <&rtc_32k>; clock-names = "mclk"; interrupts-extended = <&pioA PIN_PB3 IRQ_TYPE_LEVEL_HIGH>; #sound-dai-cells = <0>; codec_port: port { codec_ep: endpoint { remote-endpoint = <&ssc0_ep>; dai-format = "left_j"; bitclock-master; frame-master; mclk-fs = <256>; }; }; }; }; &ssc0 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_ssc0_default>; status = "okay"; atmel,shared-fs-pin; #sound-dai-cells = <0>; ssc0_port: port { ssc0_ep: endpoint { remote-endpoint = <&codec_ep>; }; }; }; Michał Mirosław (2): ASoC: wm8904: rework FLL handling ASoC: wm8904: automatically choose clock source .../devicetree/bindings/sound/wm8904.txt | 4 + sound/soc/atmel/atmel_wm8904.c | 11 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8904.c | 494 ++++++------------ sound/soc/codecs/wm8904.h | 5 - sound/soc/codecs/wm_fll.c | 339 ++++++++++++ sound/soc/codecs/wm_fll.h | 26 + 8 files changed, 531 insertions(+), 354 deletions(-) create mode 100644 sound/soc/codecs/wm_fll.c create mode 100644 sound/soc/codecs/wm_fll.h -- 2.20.1

3 7

[alsa-devel] [PATCH] ALSA: isa: gus: Fix a possible null-pointer dereference in snd_gf1_mem_xfree()
by Jia-Ju Bai 23 Jul '19

23 Jul '19

In snd_gf1_mem_xfree(), there is an if statement on line 72 and line 74 to check whether block->next is NULL: if (block->next) When block->next is NULL, block->next is used on line 84: block->next->prev = block->prev; Thus, a possible null-pointer dereference may occur in this case. To fix this possible bug, block->next is checked before using it. This bug is found by a static analysis tool STCheck written by us. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/isa/gus/gus_mem.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index cb02d18dde60..ed6205b88057 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -81,7 +81,8 @@ int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * blo if (block->prev) block->prev->next = NULL; } else { - block->next->prev = block->prev; + if (block->next) + block->next->prev = block->prev; if (block->prev) block->prev->next = block->next; } -- 2.17.0

2 2

[alsa-devel] [PATCH] ALSA: ac97: Fix double free of ac97_codec_device
by Ding Xiang 23 Jul '19

23 Jul '19

put_device will call ac97_codec_release to free ac97_codec_device and other resources, so remove the kfree and other redundant code. Signed-off-by: Ding Xiang <dingxiang(a)cmss.chinamobile.com> --- sound/ac97/bus.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7b977b7..7985dd8 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, vendor_id); ret = device_add(&codec->dev); - if (ret) - goto err_free_codec; + if (ret) { + put_device(&codec->dev); + return ret; + } return 0; -err_free_codec: - of_node_put(codec->dev.of_node); - put_device(&codec->dev); - kfree(codec); - ac97_ctrl->codecs[idx] = NULL; - - return ret; } unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links
by Stephan Gerhold 23 Jul '19

23 Jul '19

apq8016_sbc_parse_of() sets up multiple DAI links, depending on the number of nodes in the device tree. However, at the moment CPU and platform components are only allocated for the first link. This causes an oops when more than one link is defined: Internal error: Oops: 96000044 [#1] SMP CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4 Call trace: apq8016_sbc_platform_probe+0x1a8/0x3f0 platform_drv_probe+0x50/0xa0 ... Move the allocation inside the loop to ensure that each link is properly initialized. Fixes: 98b232ca9e0e ("ASoC: qcom: apq8016_sbc: use modern dai_link style") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/apq8016_sbc.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index f60a71990f66..ac75838bbfab 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -150,17 +150,17 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) link = data->dai_link; - dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); - if (!dlc) - return ERR_PTR(-ENOMEM); + for_each_child_of_node(node, np) { + dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return ERR_PTR(-ENOMEM); - link->cpus = &dlc[0]; - link->platforms = &dlc[1]; + link->cpus = &dlc[0]; + link->platforms = &dlc[1]; - link->num_cpus = 1; - link->num_platforms = 1; + link->num_cpus = 1; + link->num_platforms = 1; - for_each_child_of_node(node, np) { cpu = of_get_child_by_name(np, "cpu"); codec = of_get_child_by_name(np, "codec"); -- 2.22.0

3 2

[alsa-devel] [PATCH] ASoC: pcm3168a: Fix a typo in the name of a constant
by Christophe JAILLET 23 Jul '19

23 Jul '19

There is a typo in PCM1368A_MAX_SYSCLK, it should be PCM3168A_MAX_SYSCLK (1 and 3 switched in 3168) Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/pcm3168a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index ed0b4317c9ae..844602710eee 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -263,7 +263,7 @@ static unsigned int pcm3168a_scki_ratios[] = { #define PCM3168A_NUM_SCKI_RATIOS_DAC ARRAY_SIZE(pcm3168a_scki_ratios) #define PCM3168A_NUM_SCKI_RATIOS_ADC (ARRAY_SIZE(pcm3168a_scki_ratios) - 2) -#define PCM1368A_MAX_SYSCLK 36864000 +#define PCM3168A_MAX_SYSCLK 36864000 static int pcm3168a_reset(struct pcm3168a_priv *pcm3168a) { @@ -296,7 +296,7 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component); int ret; - if (freq > PCM1368A_MAX_SYSCLK) + if (freq > PCM3168A_MAX_SYSCLK) return -EINVAL; ret = clk_set_rate(pcm3168a->scki, freq); -- 2.20.1

2 1

[alsa-devel] [PATCH V4 0/2] recover the channel swap after xrun
by shengjiu.wang＠nxp.com 23 Jul '19

23 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> recover the channel swap after xrun Shengjiu Wang (2): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun sound/soc/fsl/fsl_esai.c | 266 ++++++++++++++++++++++++++++----------- 1 file changed, 193 insertions(+), 73 deletions(-) Changes in v4 -add more comments and remove read saisr Changes in V3 - update code sytle. Changes in v2 - add one patch for wrap operations to functions. - fix some coding style issue -- 2.21.0

3 5

[alsa-devel] [PATCH] ASoC: rt5665: Fix a typo in the name of a function
by Christophe JAILLET 23 Jul '19

23 Jul '19

All function names start with rt5665_, except 'rt5655_set_verf()'. It is likely a typo. Fix it to be consistent. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/rt5665.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index f2ad3a4c3b7f..897e53d4c40a 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2569,7 +2569,7 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static int rt5655_set_verf(struct snd_soc_dapm_widget *w, +static int rt5665_set_verf(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); @@ -2689,11 +2689,11 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5665_PWR_VOL, RT5665_PWR_MIC_DET_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Vref1", RT5665_PWR_ANLG_1, RT5665_PWR_VREF1_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref2", RT5665_PWR_ANLG_1, RT5665_PWR_VREF2_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref3", RT5665_PWR_ANLG_1, RT5665_PWR_VREF3_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5665_ASRC_1, -- 2.20.1

2 1

[alsa-devel] [PATCH v2] ASoC: rt1308: Convert headers to SPDX
by shumingf＠realtek.com 23 Jul '19

23 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> Convert include and source files to SPDX. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308.c | 17 +++++++---------- sound/soc/codecs/rt1308.h | 6 ++---- 2 files changed, 9 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index cf3d72f99705..6f2ee6809dbb 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -1,13 +1,10 @@ -/* - * rt1308.c -- RT1308 ALSA SoC amplifier component driver - * - * Copyright 2019 Realtek Semiconductor Corp. - * Author: Derek Fang <derek.fang(a)realtek.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// rt1308.c -- RT1308 ALSA SoC amplifier component driver +// +// Copyright 2019 Realtek Semiconductor Corp. +// Author: Derek Fang <derek.fang(a)realtek.com> +// #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h index c330aae1d527..ff7c423e879e 100644 --- a/sound/soc/codecs/rt1308.h +++ b/sound/soc/codecs/rt1308.h @@ -1,12 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * RT1308.h -- RT1308 ALSA SoC amplifier component driver + * rt1308.h -- RT1308 ALSA SoC amplifier component driver * * Copyright 2019 Realtek Semiconductor Corp. * Author: Derek Fang <derek.fang(a)realtek.com> * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef _RT1308_H_ -- 2.22.0

2 1

[alsa-devel] [PATCH v2 1/4] ALSA: compress: Fix regression on compressed capture streams
by Charles Keepax 23 Jul '19

23 Jul '19

A previous fix to the stop handling on compressed capture streams causes some knock on issues. The previous fix updated snd_compr_drain_notify to set the state back to PREPARED for capture streams. This causes some issues however as the handling for snd_compr_poll differs between the two states and some user-space applications were relying on the poll failing after the stream had been stopped. To correct this regression whilst still fixing the original problem the patch was addressing, update the capture handling to skip the PREPARED state rather than skipping the SETUP state as it has done until now. Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes since v1. Thanks, Charles include/sound/compress_driver.h | 5 +---- sound/core/compress_offload.c | 16 +++++++++++----- 2 files changed, 12 insertions(+), 9 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index c5188ff724d12..bc88d6f964da9 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -173,10 +173,7 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) if (snd_BUG_ON(!stream)) return; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); } diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99b8821587053..d79aee6b9edd2 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + if (stream->direction != SND_COMPRESS_CAPTURE) + return -EPERM; + break; + case SNDRV_PCM_STATE_PREPARED: + break; + default: return -EPERM; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); if (!retval) stream->runtime->state = SNDRV_PCM_STATE_RUNNING; -- 2.11.0

3 8

[alsa-devel] [PATCH] ASoC: dapm: fix a memory leak bug
by Wenwen Wang 23 Jul '19

23 Jul '19

From: Wenwen Wang <wenwen(a)cs.uga.edu> In snd_soc_dapm_new_control_unlocked(), a kernel buffer is allocated in dapm_cnew_widget() to hold the new dapm widget. Then, different actions are taken according to the id of the widget, i.e., 'w->id'. If any failure occurs during this process, snd_soc_dapm_new_control_unlocked() should be terminated by going to the 'request_failed' label. However, the allocated kernel buffer is not freed on this code path, leading to a memory leak bug. To fix the above issue, free the buffer before returning from snd_soc_dapm_new_control_unlocked() through the 'request_failed' label. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f013b24..23b9b25 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3706,6 +3706,8 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", w->name, ret); + kfree_const(w->sname); + kfree(w); return ERR_PTR(ret); } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/6] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 23 Jul '19

23 Jul '19

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://lwn.net/Articles/790651/ This patchset has been tested on SDM845 based DragonBoard DB845c with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. Thanks, srini Srinivas Kandagatla (6): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings .../bindings/sound/qcom,wcd934x.txt | 188 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5979 +++++++++++++++++ sound/soc/codecs/wcd934x.h | 426 ++ 5 files changed, 6605 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.txt create mode 100644 sound/soc/codecs/wcd934x.c create mode 100644 sound/soc/codecs/wcd934x.h -- 2.21.0

3 16

[alsa-devel] [PATCH] ASoC: intel: Add Broadwell rt5650 machine driver
by Jon Flatley 23 Jul '19

23 Jul '19

From: Ben Zhang <benzh(a)chromium.org> Add machine driver for Broadwell + rt5650. Signed-off-by: Jon Flatley <jflat(a)chromium.org> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 305 ++++++++++++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 5 + 4 files changed, 323 insertions(+) create mode 100644 sound/soc/intel/boards/bdw-rt5650.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149818b5..c111ae177b4a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -31,6 +31,17 @@ endif ## SND_SOC_INTEL_HASWELL if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL +config SND_SOC_INTEL_BDW_RT5650_MACH + tristate "Broadwell with RT5650 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT5645 + help + This adds the ASoC machine driver for Intel Broadwell platforms with + the RT5650 codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6445f90ea542..b5e2619607be 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_ obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c new file mode 100644 index 000000000000..cb875eeab055 --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,305 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * ASoC machine driver for Intel Broadwell platforms with RT5650 codec + * + * Copyright 2019, The Chromium OS Authors. All rights reserved. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt5645.h" + +struct bdw_rt5650_priv { + struct gpio_desc *gpio_hp_en; + struct snd_soc_component *component; +}; + +static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("DMIC Pair1", NULL), + SND_SOC_DAPM_MIC("DMIC Pair2", NULL), +}; + +static const struct snd_soc_dapm_route bdw_rt5650_map[] = { + /* Speakers */ + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + + /* Headset jack connectors */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + + /* Digital MICs + * DMIC Pair1 are the two DMICs connected on the DMICN1 connector. + * DMIC Pair2 are the two DMICs connected on the DMICN2 connector. + * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2 + * are on the right side. + */ + {"DMIC L1", NULL, "DMIC Pair1"}, + {"DMIC R1", NULL, "DMIC Pair1"}, + {"DMIC L2", NULL, "DMIC Pair2"}, + {"DMIC R2", NULL, "DMIC Pair2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static const struct snd_kcontrol_new bdw_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("DMIC Pair1"), + SOC_DAPM_PIN_SWITCH("DMIC Pair2"), +}; + + +static struct snd_soc_jack headphone_jack; +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin headphone_jack_pin = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, max 4-channels */ + rate->min = rate->max = 48000; + channels->min = 2; + channels->max = 4; + + /* set SSP0 to 24 bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Workaround: set codec PLL to 19.2MHz that PLL source is + * from MCLK(24MHz) to conform 2.4MHz DMIC clock. + */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + 24000000, 19200000); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* The actual MCLK freq is 24MHz. The codec is told that MCLK is + * 24.576MHz to satisfy the requirement of rl6231_get_clk_info. + * ASRC is enabled on AD and DA filters to ensure good audio quality. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops bdw_rt5650_ops = { + .hw_params = bdw_rt5650_hw_params, +}; + +static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct sst_pdata *pdata = dev_get_platdata(component->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings + * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz + */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + return 0; +} + +static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) +{ + struct bdw_rt5650_priv *bdw_rt5650 = + snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. + * The ASRC clock source is clk_i2s1_asrc. + */ + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER | + RT5645_AD_MONO_L_FILTER | + RT5645_AD_MONO_R_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* Create and initialize headphone jack */ + if (snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &headphone_jack, + &headphone_jack_pin, 1)) { + dev_err(component->dev, "Can't create headphone jack\n"); + } + + /* Create and initialize mic jack */ + if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &mic_jack, &mic_jack_pin, 1)) { + dev_err(component->dev, "Can't create mic jack\n"); + } + + rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL); + + bdw_rt5650->component = component; + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); +SND_SOC_DAILINK_DEF(fe, DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); +SND_SOC_DAILINK_DEF(be, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); + +static struct snd_soc_dai_link bdw_rt5650_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .dynamic = 1, + .init = bdw_rt5650_rtd_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(fe, dummy, platform), + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &bdw_rt5650_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = bdw_rt5650_init, + SND_SOC_DAILINK_REG(dummy, be, dummy) + }, +}; + +/* ASoC machine driver for Broadwell DSP + RT5650 */ +static struct snd_soc_card bdw_rt5650_card = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + .dai_link = bdw_rt5650_dais, + .num_links = ARRAY_SIZE(bdw_rt5650_dais), + .dapm_widgets = bdw_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets), + .dapm_routes = bdw_rt5650_map, + .num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map), + .controls = bdw_rt5650_controls, + .num_controls = ARRAY_SIZE(bdw_rt5650_controls), + .fully_routed = true, +}; + +static int bdw_rt5650_probe(struct platform_device *pdev) +{ + struct bdw_rt5650_priv *bdw_rt5650; + + bdw_rt5650_card.dev = &pdev->dev; + + /* Allocate driver private struct */ + bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv), + GFP_KERNEL); + if (!bdw_rt5650) + return -ENOMEM; + snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650); + + return snd_soc_register_card(&bdw_rt5650_card); +} + +static int bdw_rt5650_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&bdw_rt5650_card); + return 0; +} + +static struct platform_driver bdw_rt5650_audio = { + .probe = bdw_rt5650_probe, + .remove = bdw_rt5650_remove, + .driver = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bdw_rt5650_audio) + +/* Module information */ +MODULE_AUTHOR("Ben Zhang <benzh(a)chromium.org>"); +MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bdw-rt5650"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index d27853e7a369..ba3d25658436 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -29,6 +29,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .sof_fw_filename = "sof-bdw.ri", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, + { + .id = "10EC5650", + .drv_name = "bdw-rt5650", + .fw_filename = "intel/IntcSST2.bin", + }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", -- 2.22.0.657.g960e92d24f-goog

2 3

[alsa-devel] [PATCH] ASoC: rt5677: remove superfluous set
by Curtis Malainey 22 Jul '19

22 Jul '19

We define this variable with the same value, no need to set it twice Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c779dc3474f9e..deaed5132dc95 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5098,7 +5098,6 @@ static irqreturn_t rt5677_irq(int unused, void *data) goto exit; } - irq_fired = false; for (i = 0; i < RT5677_IRQ_NUM; i++) { if (reg_irq & rt5677_irq_descs[i].status_mask) { irq_fired = true; -- 2.22.0.410.gd8fdbe21b5-goog

3 3

[alsa-devel] [PATCH] ASoC: SOF: use __u32 instead of uint32_t in uapi headers
by Masahiro Yamada 22 Jul '19

22 Jul '19

When CONFIG_UAPI_HEADER_TEST=y, exported headers are compile-tested to make sure they can be included from user-space. Currently, header.h and fw.h are excluded from the test coverage. To make them join the compile-test, we need to fix the build errors attached below. For a case like this, we decided to use __u{8,16,32,64} variable types in this discussion: https://lkml.org/lkml/2019/6/5/18 Build log: CC usr/include/sound/sof/header.h.s CC usr/include/sound/sof/fw.h.s In file included from <command-line>:32:0: ./usr/include/sound/sof/header.h:19:2: error: unknown type name ‘uint32_t’ uint32_t magic; /**< 'S', 'O', 'F', '\0' */ ^~~~~~~~ ./usr/include/sound/sof/header.h:20:2: error: unknown type name ‘uint32_t’ uint32_t type; /**< component specific type */ ^~~~~~~~ ./usr/include/sound/sof/header.h:21:2: error: unknown type name ‘uint32_t’ uint32_t size; /**< size in bytes of data excl. this struct */ ^~~~~~~~ ./usr/include/sound/sof/header.h:22:2: error: unknown type name ‘uint32_t’ uint32_t abi; /**< SOF ABI version */ ^~~~~~~~ ./usr/include/sound/sof/header.h:23:2: error: unknown type name ‘uint32_t’ uint32_t reserved[4]; /**< reserved for future use */ ^~~~~~~~ ./usr/include/sound/sof/header.h:24:2: error: unknown type name ‘uint32_t’ uint32_t data[0]; /**< Component data - opaque to core */ ^~~~~~~~ In file included from <command-line>:32:0: ./usr/include/sound/sof/fw.h:49:2: error: unknown type name ‘uint32_t’ uint32_t size; /* bytes minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:50:2: error: unknown type name ‘uint32_t’ uint32_t offset; /* offset from base */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:64:2: error: unknown type name ‘uint32_t’ uint32_t size; /* bytes minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:65:2: error: unknown type name ‘uint32_t’ uint32_t num_blocks; /* number of blocks */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:73:2: error: unknown type name ‘uint32_t’ uint32_t file_size; /* size of file minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:74:2: error: unknown type name ‘uint32_t’ uint32_t num_modules; /* number of modules */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:75:2: error: unknown type name ‘uint32_t’ uint32_t abi; /* version of header format */ ^~~~~~~~ Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- include/uapi/sound/sof/fw.h | 16 +++++++++------- include/uapi/sound/sof/header.h | 14 ++++++++------ 2 files changed, 17 insertions(+), 13 deletions(-) diff --git a/include/uapi/sound/sof/fw.h b/include/uapi/sound/sof/fw.h index 1afca973eb09..e9f697467a86 100644 --- a/include/uapi/sound/sof/fw.h +++ b/include/uapi/sound/sof/fw.h @@ -13,6 +13,8 @@ #ifndef __INCLUDE_UAPI_SOF_FW_H__ #define __INCLUDE_UAPI_SOF_FW_H__ +#include <linux/types.h> + #define SND_SOF_FW_SIG_SIZE 4 #define SND_SOF_FW_ABI 1 #define SND_SOF_FW_SIG "Reef" @@ -46,8 +48,8 @@ enum snd_sof_fw_blk_type { struct snd_sof_blk_hdr { enum snd_sof_fw_blk_type type; - uint32_t size; /* bytes minus this header */ - uint32_t offset; /* offset from base */ + __u32 size; /* bytes minus this header */ + __u32 offset; /* offset from base */ } __packed; /* @@ -61,8 +63,8 @@ enum snd_sof_fw_mod_type { struct snd_sof_mod_hdr { enum snd_sof_fw_mod_type type; - uint32_t size; /* bytes minus this header */ - uint32_t num_blocks; /* number of blocks */ + __u32 size; /* bytes minus this header */ + __u32 num_blocks; /* number of blocks */ } __packed; /* @@ -70,9 +72,9 @@ struct snd_sof_mod_hdr { */ struct snd_sof_fw_header { unsigned char sig[SND_SOF_FW_SIG_SIZE]; /* "Reef" */ - uint32_t file_size; /* size of file minus this header */ - uint32_t num_modules; /* number of modules */ - uint32_t abi; /* version of header format */ + __u32 file_size; /* size of file minus this header */ + __u32 num_modules; /* number of modules */ + __u32 abi; /* version of header format */ } __packed; #endif diff --git a/include/uapi/sound/sof/header.h b/include/uapi/sound/sof/header.h index 7868990b0d6f..5f4518e7a972 100644 --- a/include/uapi/sound/sof/header.h +++ b/include/uapi/sound/sof/header.h @@ -9,6 +9,8 @@ #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__ #define __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__ +#include <linux/types.h> + /* * Header for all non IPC ABI data. * @@ -16,12 +18,12 @@ * Used by any bespoke component data structures or binary blobs. */ struct sof_abi_hdr { - uint32_t magic; /**< 'S', 'O', 'F', '\0' */ - uint32_t type; /**< component specific type */ - uint32_t size; /**< size in bytes of data excl. this struct */ - uint32_t abi; /**< SOF ABI version */ - uint32_t reserved[4]; /**< reserved for future use */ - uint32_t data[0]; /**< Component data - opaque to core */ + __u32 magic; /**< 'S', 'O', 'F', '\0' */ + __u32 type; /**< component specific type */ + __u32 size; /**< size in bytes of data excl. this struct */ + __u32 abi; /**< SOF ABI version */ + __u32 reserved[4]; /**< reserved for future use */ + __u32 data[0]; /**< Component data - opaque to core */ } __packed; #endif -- 2.17.1

5 10

[alsa-devel] [PATCH 00/20] ALSA: firewire-lib: use packet descriptor to represent sequence of packet
by Takashi Sakamoto 22 Jul '19

22 Jul '19

Hi, This patchset is for Linux kernel v5.4. In current implementation of ALSA IEC 61883-1/6 packet streaming engine, one callback of isochronous context consists three steps to process an isochronous packet: Callback of isochronous context - For each of packets 1.parse context header and decide the amount of PCM frames 2.process data blocks (in each unit driver) 3.queue packet The result of step 1 affects the rest, thus the above steps can be changed for below process: Callback of isochronous context 1.parse context header and decide the amount of PCM frames - For each of packets 2.process data blocks (in each unit driver) - For each of packets 3.queue packet - For each of packets Especially it's convenient to decide the amount of PCM frames multiplexed into each packet as batch calculation before processing payloads of these packets. Additionally it allows each unit driver to process data blocks for these packet by one call. This patchset uses list of 'struct pkt_desc' as an intermediate representation of packet parameters between these three steps. Here, I note my future plan for ALSA firewire stack for your motivation to review this patchset: * v5.4 * Handle several IT/IR context in one callback of IT context. This idea will introduce AMDTP domain structure with relationship of IRQ master/slave between the IT/IR contexts. * As a result, both of playback/capture PCM substreams run on the same hardware interrupt. Thus the drivers just support batch PCM operation and irq-based programming model[1]. * v5.5 * Some (p.s. not all) supported devices don't follow packet sequences transferred by the drivers. They require clock recovery in driver side to reduce playback noise. Enhance the AMDTP domain structure to have clock recovery target and fill list of 'struct pkt_desc' according to the result of clock recovery. My work for libhinoko[2] allows me to sniff actual packet transmission between devices and drivers in Windows/Mac OS for long period (e.g. 1 hour). Then I can classify some cases and some devices for the requirement of clock recovery. (I note that no specification describes this mechanism as long as I know. In the specification, clock recovery is one-way from transmitter to receivers, thus recovered clock is not necessarily used for transmission from the receiver to the transmitter.) [1] https://git.alsa-project.org/?p=alsa-utils.git;a=blob;f=axfer/axfer-transfe… [2] https://github.com/takaswie/libhinoko Takashi Sakamoto (20): ALSA: firewire-lib: obsolete ctx_data.tx.first_dbc with CIP_UNALIGHED_DBC flag ALSA: firewire-lib: pass data block count as an argument to tracepoints event ALSA: firewire-lib: pass data block counter to data block processing layer ALSA: firewire-lib: operate data block counter in top level of processing for IT context ALSA: firewire-lib: operate data block counter in top level of processing for IR context ALSA: firewire-lib: add syt_override member for some protocols ALSA: firewire-lib: pass no syt information to data block processing layer ALSA: firewire-lib: add list of packet descriptor ALSA: firewire-lib: use packet descriptor for IT context ALSA: firewire-lib: use packet descriptor for IR context ALSA: firewire-lib: code refactoring to process PCM substream ALSA: firewire-lib: code refactoring to process context payloads ALSA: firewire-lib: pass packet descriptor to data block processing layer ALSA: firewire-lib: code refactoring for AM824 data block processing layer ALSA: firewire-digi00x: code refactoring for DOT data block processing layer ALSA: firewire-tascam: code refactoring for TASCAM data block processing layer ALSA: firewire-motu: code refactoring for MOTU data block processing layer ALSA: fireface: code refactoring for FF data block processing layer ALSA: firewire-lib: process payload of isoc context according to packet descriptors ALSA: firewire-motu: more code refactoring for MOTU data block processing layer sound/firewire/amdtp-am824.c | 134 ++++++---- sound/firewire/amdtp-stream-trace.h | 6 +- sound/firewire/amdtp-stream.c | 264 ++++++++++++-------- sound/firewire/amdtp-stream.h | 27 +- sound/firewire/digi00x/amdtp-dot.c | 112 +++++---- sound/firewire/fireface/amdtp-ff.c | 105 ++++---- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/amdtp-motu.c | 155 ++++++++---- sound/firewire/tascam/amdtp-tascam.c | 115 +++++---- 9 files changed, 570 insertions(+), 350 deletions(-) -- 2.20.1

2 21

[alsa-devel] [PATCH 0/3] i.MX8M support for FSL SAI
by Lucas Stach 22 Jul '19

22 Jul '19

Hi all, those are the changes that are absolutely needed to support the i.MX8M SAI block. The most noticable difference is the moved register layout, due to the hardware guys adding two version registers at the start of the register map and considerable bigger FIFO sizes. Downstream has a lot more changes to support new features, but those 3 patches at least allows me to play back audio on i.MX8M based systems. Regards, Lucas Lucas Stach (3): ASoC: fsl_sai: add of_match data ASoC: fsl_sai: derive TX FIFO watermark from FIFO depth ASoC: fsl_sai: add i.MX8M support .../devicetree/bindings/sound/fsl-sai.txt | 2 +- sound/soc/fsl/fsl_sai.c | 45 +++++++++++++++---- sound/soc/fsl/fsl_sai.h | 8 +++- 3 files changed, 44 insertions(+), 11 deletions(-) -- 2.20.1

5 19

[alsa-devel] [PATCH] ASoC: cs42xx8: Fix MFREQ selection issue for async mode
by shengjiu.wang＠nxp.com 22 Jul '19

22 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> When sample rate of TX is different with sample rate of RX in async mode, the MFreq selection will be wrong. For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz. Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX and RX instance, the correct value of MFreq is 2 for both TX and RX. But original method will cause MFreq = 0 for TX, MFreq = 2 for RX. If TX is started after RX, RX will be impacted, RX work abnormal with MFreq = 0. This patch is to select proper MFreq value according to TX rate and RX rate. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++++++++++++++++------ 1 file changed, 97 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index b377cddaf2e6..804969495754 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -47,6 +47,7 @@ struct cs42xx8_priv { unsigned long sysclk; u32 tx_channels; struct gpio_desc *gpiod_reset; + u32 rate[2]; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -176,21 +177,27 @@ static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { }; struct cs42xx8_ratios { - unsigned int ratio; - unsigned char speed; - unsigned char mclk; + unsigned int mfreq; + unsigned int min_mclk; + unsigned int max_mclk; + unsigned int ratio[3]; }; +/* + * According to reference mannual, define the cs42xx8_ratio struct + * MFreq2 | MFreq1 | MFreq0 | Description | SSM | DSM | QSM | + * 0 | 0 | 0 |1.029MHz to 12.8MHz | 256 | 128 | 64 | + * 0 | 0 | 1 |1.536MHz to 19.2MHz | 384 | 192 | 96 | + * 0 | 1 | 0 |2.048MHz to 25.6MHz | 512 | 256 | 128 | + * 0 | 1 | 1 |3.072MHz to 38.4MHz | 768 | 384 | 192 | + * 1 | x | x |4.096MHz to 51.2MHz |1024 | 512 | 256 | + */ static const struct cs42xx8_ratios cs42xx8_ratios[] = { - { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, - { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, - { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, - { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, - { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, - { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, - { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, - { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, - { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } + { 0, 1029000, 12800000, {256, 128, 64} }, + { 2, 1536000, 19200000, {384, 192, 96} }, + { 4, 2048000, 25600000, {512, 256, 128} }, + { 6, 3072000, 38400000, {768, 384, 192} }, + { 8, 4096000, 51200000, {1024, 512, 256} }, }; static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -257,14 +264,68 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - u32 ratio = cs42xx8->sysclk / params_rate(params); - u32 i, fm, val, mask; + u32 ratio[2]; + u32 rate[2]; + u32 fm[2]; + u32 i, val, mask; + bool condition1, condition2; if (tx) cs42xx8->tx_channels = params_channels(params); + rate[tx] = params_rate(params); + rate[!tx] = cs42xx8->rate[!tx]; + + ratio[tx] = rate[tx] > 0 ? cs42xx8->sysclk / rate[tx] : 0; + ratio[!tx] = rate[!tx] > 0 ? cs42xx8->sysclk / rate[!tx] : 0; + + /* Get functional mode for tx and rx according to rate */ + for (i = 0; i < 2; i++) { + if (cs42xx8->slave_mode) { + fm[i] = CS42XX8_FM_AUTO; + } else { + if (rate[i] < 50000) { + fm[i] = CS42XX8_FM_SINGLE; + } else if (rate[i] > 50000 && rate[i] < 100000) { + fm[i] = CS42XX8_FM_DOUBLE; + } else if (rate[i] > 100000 && rate[i] < 200000) { + fm[i] = CS42XX8_FM_QUAD; + } else { + dev_err(component->dev, + "unsupported sample rate\n"); + return -EINVAL; + } + } + } + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { - if (cs42xx8_ratios[i].ratio == ratio) + /* Is the ratio[tx] valid ? */ + condition1 = ((fm[tx] == CS42XX8_FM_AUTO) ? + (cs42xx8_ratios[i].ratio[0] == ratio[tx] || + cs42xx8_ratios[i].ratio[1] == ratio[tx] || + cs42xx8_ratios[i].ratio[2] == ratio[tx]) : + (cs42xx8_ratios[i].ratio[fm[tx]] == ratio[tx])) && + cs42xx8->sysclk >= cs42xx8_ratios[i].min_mclk && + cs42xx8->sysclk <= cs42xx8_ratios[i].max_mclk; + + if (!ratio[tx]) + condition1 = true; + + /* Is the ratio[!tx] valid ? */ + condition2 = ((fm[!tx] == CS42XX8_FM_AUTO) ? + (cs42xx8_ratios[i].ratio[0] == ratio[!tx] || + cs42xx8_ratios[i].ratio[1] == ratio[!tx] || + cs42xx8_ratios[i].ratio[2] == ratio[!tx]) : + (cs42xx8_ratios[i].ratio[fm[!tx]] == ratio[!tx])); + + if (!ratio[!tx]) + condition2 = true; + + /* + * Both ratio[tx] and ratio[!tx] is valid, then we get + * a proper MFreq. + */ + if (condition1 && condition2) break; } @@ -273,15 +334,31 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - mask = CS42XX8_FUNCMOD_MFREQ_MASK; - val = cs42xx8_ratios[i].mclk; + cs42xx8->rate[tx] = params_rate(params); - fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mfreq; regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, - CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + CS42XX8_FUNCMOD_xC_FM(tx, fm[tx]) | val); + + return 0; +} + +static int cs42xx8_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + /* Clear stored rate */ + cs42xx8->rate[tx] = 0; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx), + CS42XX8_FUNCMOD_xC_FM(tx, CS42XX8_FM_AUTO)); return 0; } @@ -302,6 +379,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = { .set_fmt = cs42xx8_set_dai_fmt, .set_sysclk = cs42xx8_set_dai_sysclk, .hw_params = cs42xx8_hw_params, + .hw_free = cs42xx8_hw_free, .digital_mute = cs42xx8_digital_mute, }; -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks
by Charles Keepax 22 Jul '19

22 Jul '19

DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a list of the widgets connected to a specific front end DAI so it can search through this list for available back end DAIs. The custom_stop_condition was added to is_connected_ep to facilitate this list not containing more widgets than is necessary. Doing so both speeds up the DPCM handling as less widgets need to be searched and avoids issues with CODEC to CODEC links as these would be confused with back end DAIs if they appeared in the list of available widgets. custom_stop_condition was implemented by aborting the graph walk when the condition is triggered, however there is an issue with this approach. Whilst walking the graph is_connected_ep should update the endpoints cache on each widget, if the walk is aborted the number of attached end points is unknown for that sub-graph. When the stop condition triggered, the original patch ignored the triggering widget and returned zero connected end points; a later patch updated this to set the triggering widget's cache to 1 and return that. Both of these approaches result in inaccurate values being stored in various end point caches as the values propagate back through the graph, which can result in later issues with widgets powering/not powering unexpectedly. As the original goal was to reduce the size of the widget list passed to the DPCM code, the simplest solution is to limit the functionality of the custom_stop_condition to the widget list. This means the rest of the graph will still be processed resulting in correct end point caches, but only widgets up to the stop condition will be added to the returned widget list. Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets") Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links") Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f013b24c050a1..f73882cf0031d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1157,8 +1157,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, list_add_tail(&widget->work_list, list); if (custom_stop_condition && custom_stop_condition(widget, dir)) { - widget->endpoints[dir] = 1; - return widget->endpoints[dir]; + list = NULL; + custom_stop_condition = NULL; } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { @@ -1195,8 +1195,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, * * Optionally, can be supplied with a function acting as a stopping condition. * This function takes the dapm widget currently being examined and the walk - * direction as an arguments, it should return true if the walk should be - * stopped and false otherwise. + * direction as an arguments, it should return true if widgets from that point + * in the graph onwards should not be added to the widget list. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, -- 2.11.0

2 1

[alsa-devel] [PATCH] SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection
by Enric Balletbo i Serra 22 Jul '19

22 Jul '19

The TS3A227E says that the headset keypress detection needs the MICBIAS power in order to report the key events to ensure proper operation The headset keypress detection needs the MICBIAS power in order to report the key events all the time as long as MIC is present. So MICBIAS pin is forced on when a MICROPHONE is detected. On Veyron Minnie I observed that if the MICBIAS power is not present and the key press detection is activated (just because it is enabled when you insert a headset), it randomly reports a keypress on insert. E.g. (KEY_PLAYPAUSE) Event: (SW_HEADPHONE_INSERT), value 1 Event: (SW_MICROPHONE_INSERT), value 1 Event: -------------- SYN_REPORT ------------ Event: (KEY_PLAYPAUSE), value 1 Userspace thinks that KEY_PLAYPAUSE is pressed and produces the annoying effect that the media player starts a play/pause loop. Note that, although most of the time the key reported is the one associated with BTN_0, not always this is true. On my tests I also saw different keys reported Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- Some notes about the patch Steps to test (userspace GNOME3): 1. Play audio using a media player 2. Make sure the Internal MIC is selected as audio input device and doesn't switches when you insert the headset. 3. Insert a headset (with buttons) Audio switches to headphones and you can hear that enters on a loop play/pause if the KEY_PLAYPAUSE was reported. Also you can check that press to any headset button doesn't work. The part where the datasheet says that power must be supplied in order to have keypress detection work is in page 44 of [1] " The TS3A227E can monitor the microphone line of a 4-pole headset to detect up to 4 key presses/releases and report the key press events back to the host. The key press detection must be activated manually by setting the KP Enable bit of the Device Settings 2 register. To ensure proper operation the MICBIAS voltage must be applied to MICP before enabling key press detection. " [1] http://www.ti.com/lit/ds/symlink/ts3a227e.pdf sound/soc/rockchip/rockchip_max98090.c | 32 ++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index c5fc24675a33..782e534d4c0d 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -61,6 +61,37 @@ static const struct snd_kcontrol_new rk_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; +static int rk_jack_event(struct notifier_block *nb, unsigned long event, + void *data) +{ + struct snd_soc_jack *jack = (struct snd_soc_jack *)data; + struct snd_soc_dapm_context *dapm = &jack->card->dapm; + + if (event & SND_JACK_MICROPHONE) + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + else + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct notifier_block rk_jack_nb = { + .notifier_call = rk_jack_event, +}; + +static int rk_init(struct snd_soc_pcm_runtime *runtime) +{ + /* + * The jack has already been created in the rk_98090_headset_init() + * function. + */ + snd_soc_jack_notifier_register(&headset_jack, &rk_jack_nb); + + return 0; +} + static int rk_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -119,6 +150,7 @@ SND_SOC_DAILINK_DEFS(hifi, static struct snd_soc_dai_link rk_dailink = { .name = "max98090", .stream_name = "Audio", + .init = rk_init, .ops = &rk_aif1_ops, /* set max98090 as slave */ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | -- 2.20.1

2 1

[alsa-devel] [PATCH 00/12] treewide: Fix GENMASK misuses
by Joe Perches 22 Jul '19

22 Jul '19

These GENMASK uses are inverted argument order and the actual masks produced are incorrect. Fix them. Add checkpatch tests to help avoid more misuses too. Joe Perches (12): checkpatch: Add GENMASK tests clocksource/drivers/npcm: Fix misuse of GENMASK macro drm: aspeed_gfx: Fix misuse of GENMASK macro iio: adc: max9611: Fix misuse of GENMASK macro irqchip/gic-v3-its: Fix misuse of GENMASK macro mmc: meson-mx-sdio: Fix misuse of GENMASK macro net: ethernet: mediatek: Fix misuses of GENMASK macro net: stmmac: Fix misuses of GENMASK macro rtw88: Fix misuse of GENMASK macro phy: amlogic: G12A: Fix misuse of GENMASK macro staging: media: cedrus: Fix misuse of GENMASK macro ASoC: wcd9335: Fix misuse of GENMASK macro drivers/clocksource/timer-npcm7xx.c | 2 +- drivers/gpu/drm/aspeed/aspeed_gfx.h | 2 +- drivers/iio/adc/max9611.c | 2 +- drivers/irqchip/irq-gic-v3-its.c | 2 +- drivers/mmc/host/meson-mx-sdio.c | 2 +- drivers/net/ethernet/mediatek/mtk_eth_soc.h | 2 +- drivers/net/ethernet/mediatek/mtk_sgmii.c | 2 +- drivers/net/ethernet/stmicro/stmmac/descs.h | 2 +- drivers/net/ethernet/stmicro/stmmac/dwmac-sun8i.c | 4 ++-- drivers/net/wireless/realtek/rtw88/rtw8822b.c | 2 +- drivers/phy/amlogic/phy-meson-g12a-usb2.c | 2 +- drivers/staging/media/sunxi/cedrus/cedrus_regs.h | 2 +- scripts/checkpatch.pl | 15 +++++++++++++++ sound/soc/codecs/wcd-clsh-v2.c | 2 +- 14 files changed, 29 insertions(+), 14 deletions(-) -- 2.15.0

6 8

[alsa-devel] [PATCH] ASoC: codecs: ad193x: Use regmap_multi_reg_write() when initializing
by Codrin Ciubotariu 22 Jul '19

22 Jul '19

Using regmap_multi_reg_write() when we set the default values for our registers makes the code smaller and easier to read. Suggested-by: Tzung-Bi Shih <tzungbi(a)google.com> Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/ad193x.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 80dab5df9633..fb04c9379b71 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -413,15 +413,10 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { .ops = &ad193x_dai_ops, }; -struct ad193x_reg_default { - unsigned int reg; - unsigned int val; -}; - /* codec register values to set after reset */ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { - const struct ad193x_reg_default reg_init[] = { + const struct reg_sequence reg_init[] = { { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ @@ -437,21 +432,17 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { 12, 0x00 }, /* DAC_L4_VOL: no attenuation */ { 13, 0x00 }, /* DAC_R4_VOL: no attenuation */ }; - const struct ad193x_reg_default reg_adc_init[] = { + const struct reg_sequence reg_adc_init[] = { { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ { 16, 0x00 }, /* ADC_CTRL2: reset */ }; - int i; - for (i = 0; i < ARRAY_SIZE(reg_init); i++) - regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val); + regmap_multi_reg_write(ad193x->regmap, reg_init, ARRAY_SIZE(reg_init)); if (ad193x_has_adc(ad193x)) { - for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) { - regmap_write(ad193x->regmap, reg_adc_init[i].reg, - reg_adc_init[i].val); - } + regmap_multi_reg_write(ad193x->regmap, reg_adc_init, + ARRAY_SIZE(reg_adc_init)); } } -- 2.20.1

3 2

[alsa-devel] [PATCH v2] ASoC: SOF: Intel: hda: Make hdac_device device-managed
by Ranjani Sridharan 22 Jul '19

22 Jul '19

snd_hdac_ext_bus_device_exit() has been recently modified to no longer free the hdac device. SOF allocates memory for hdac_device and hda_hda_priv with kzalloc. Make them device-managed instead so that they will be freed when the SOF driver is unloaded. Because of the above change, hda_codec is device-managed and it will be freed when the ASoC device is removed. Freeing the codec in snd_hda_codec_dev_release() leads to kernel panic while unloading and reloading the ASoC driver. So, avoid freeing the hda_codec for ASoC driver. This is done in the same patch to avoid bisect failure. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 +++++++- sound/soc/sof/intel/hda-codec.c | 6 ++---- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6c51b8363f8b..a2e23d7e768f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -846,7 +846,13 @@ static void snd_hda_codec_dev_release(struct device *dev) snd_hda_sysfs_clear(codec); kfree(codec->modelname); kfree(codec->wcaps); - kfree(codec); + + /* + * In the case of ASoC HD-audio, hda_codec is device managed. + * It will be freed when the ASoC device is removed. + */ + if (codec->core.type == HDA_DEV_LEGACY) + kfree(codec); } #define DEV_NAME_LEN 31 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index b8b37f082309..0d8437b080bf 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -62,8 +62,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) address, resp); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hda_priv = kzalloc(sizeof(*hda_priv), GFP_KERNEL); + hda_priv = devm_kzalloc(sdev->dev, sizeof(*hda_priv), GFP_KERNEL); if (!hda_priv) return -ENOMEM; @@ -82,8 +81,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) return 0; #else - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hdev = kzalloc(sizeof(*hdev), GFP_KERNEL); + hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) return -ENOMEM; -- 2.17.1

3 2

[alsa-devel] [PATCH] sound: soc: codecs: mt6358: change return type of mt6358_codec_init_reg
by Hariprasad Kelam 22 Jul '19

22 Jul '19

As mt6358_codec_init_reg function always returns 0 , change return type from int to void. fixes below issue reported by coccicheck sound/soc/codecs/mt6358.c:2260:5-8: Unneeded variable: "ret". Return "0" on line 2289 Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/soc/codecs/mt6358.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 50b3fc5..c17250a 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -2255,10 +2255,8 @@ static struct snd_soc_dai_driver mt6358_dai_driver[] = { }, }; -static int mt6358_codec_init_reg(struct mt6358_priv *priv) +static void mt6358_codec_init_reg(struct mt6358_priv *priv) { - int ret = 0; - /* Disable HeadphoneL/HeadphoneR short circuit protection */ regmap_update_bits(priv->regmap, MT6358_AUDDEC_ANA_CON0, RG_AUDHPLSCDISABLE_VAUDP15_MASK_SFT, @@ -2285,8 +2283,6 @@ static int mt6358_codec_init_reg(struct mt6358_priv *priv) /* set gpio */ playback_gpio_reset(priv); capture_gpio_reset(priv); - - return ret; } static int mt6358_codec_probe(struct snd_soc_component *cmpnt) -- 2.7.4

3 3

[alsa-devel] [PATCH] sound: soc: bcm: cygnus-pcm: Unneeded variable: "ret".
by Hariprasad Kelam 22 Jul '19

22 Jul '19

This patch fixes below issues reported by coccicheck sound/soc/bcm/cygnus-pcm.c:642:5-8: Unneeded variable: "ret". Return "0" on line 650 sound/soc/bcm/cygnus-pcm.c:671:5-8: Unneeded variable: "ret". Return "0" on line 696 We cannot change return type of these functions as they are callback functions of snd_pcm_ops Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/soc/bcm/cygnus-pcm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c index 123ecf5..8966b02 100644 --- a/sound/soc/bcm/cygnus-pcm.c +++ b/sound/soc/bcm/cygnus-pcm.c @@ -639,7 +639,6 @@ static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct cygnus_aio_port *aio; - int ret = 0; aio = cygnus_dai_get_dma_data(substream); dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); @@ -647,7 +646,7 @@ static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); - return ret; + return 0; } static int cygnus_pcm_hw_free(struct snd_pcm_substream *substream) @@ -668,7 +667,6 @@ static int cygnus_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct cygnus_aio_port *aio; unsigned long bufsize, periodsize; - int ret = 0; bool is_play; u32 start; struct ringbuf_regs *p_rbuf = NULL; @@ -693,7 +691,7 @@ static int cygnus_pcm_prepare(struct snd_pcm_substream *substream) ringbuf_set_initial(aio->cygaud->audio, p_rbuf, is_play, start, periodsize, bufsize); - return ret; + return 0; } static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_pcm_substream *substream) -- 2.7.4

2 1

[alsa-devel] [Patch v2] sound: soc: codecs: wcd9335: add irqflag IRQF_ONESHOT flag
by Hariprasad Kelam 22 Jul '19

22 Jul '19

Add IRQF_ONESHOT to ensure "Interrupt is not reenabled after the hardirq handler finished". fixes below issue reported by coccicheck sound/soc/codecs/wcd9335.c:4068:8-33: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- v1 : add IRQF_ONESHOT and introduce irqflags variable v2 : remove irqflags variable sound/soc/codecs/wcd9335.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 1bbbe42..9566027 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4062,7 +4062,8 @@ static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) ret = devm_request_threaded_irq(wcd->dev, irq, NULL, wcd9335_irqs[i].handler, - IRQF_TRIGGER_RISING, + IRQF_TRIGGER_RISING | + IRQF_ONESHOT, wcd9335_irqs[i].name, wcd); if (ret) { dev_err(wcd->dev, "Failed to request %s\n", -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: mediatek: mt8183: make headset codec optional
by Tzung-Bi Shih 22 Jul '19

22 Jul '19

Make headset codec optional because some variant machines may not have an audio jack. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 887c932229d0..4c816c86844b 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -352,8 +352,6 @@ static struct snd_soc_card mt8183_mt6358_ts3a227_max98357_card = { .owner = THIS_MODULE, .dai_link = mt8183_mt6358_ts3a227_max98357_dai_links, .num_links = ARRAY_SIZE(mt8183_mt6358_ts3a227_max98357_dai_links), - .aux_dev = &mt8183_mt6358_ts3a227_max98357_headset_dev, - .num_aux_devs = 1, }; static int @@ -404,10 +402,9 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,headset-codec", 0); - if (!mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) { - dev_err(&pdev->dev, - "Property 'mediatek,headset-codec' missing/invalid\n"); - return -EINVAL; + if (mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) { + card->aux_dev = &mt8183_mt6358_ts3a227_max98357_headset_dev; + card->num_aux_devs = 1; } default_pins = -- 2.22.0.510.g264f2c817a-goog

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: mark regmap as fast_io
by Lucas Stach 22 Jul '19

22 Jul '19

The regmap is only ever used to access MMIO registers, so it's fair to say that register access is fast. Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8593269156bd..812c94fbb160 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -771,6 +771,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, + .fast_io = true, .max_register = FSL_SAI_RMR, .reg_defaults = fsl_sai_reg_defaults, -- 2.20.1

2 3

[alsa-devel] [PATCH] ASoC: max98383: fix i2c probe failure
by chunguo feng 22 Jul '19

22 Jul '19

From: fengchunguo <chunguo.feng(a)amlogic.com> Added reset_gpio configuration for i2c probe successfully. If not,i2c address can't be found rightly. Error information: max98373 3-0031: Failed to read: 0x21FF Fixed: [3.761299@3] max98373 3-0031: MAX98373 revisionID: 0x43 [3.828911@3] asoc-aml-card auge_sound: max98373-aif1 <-> TDM-B mapping ok Signed-off-by: fengchunguo <chunguo.feng(a)amlogic.com> --- sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/codecs/max98373.h | 1 + 2 files changed, 31 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 528695cd6a1c..9a1eb7222357 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -12,6 +12,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/gpio.h> +#include <linux/of.h> #include <linux/of_gpio.h> #include <sound/tlv.h> #include "max98373.h" @@ -895,6 +896,17 @@ static void max98373_slot_config(struct i2c_client *i2c, else max98373->i_slot = 1; + max98373->reset_gpio = of_get_named_gpio(dev->of_node, + "maxim,reset-gpio", 0); + if (!gpio_is_valid(max98373->reset_gpio)) { + dev_err(dev, "Looking up %s property in node %s failed %d\n", + "maxim,reset-gpio", dev->of_node->full_name, + max98373->reset_gpio); + } else { + dev_dbg(dev, "maxim,reset-gpio=%d", + max98373->reset_gpio); + } + if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) max98373->spkfb_slot = value & 0xF; else @@ -923,7 +935,6 @@ static int max98373_i2c_probe(struct i2c_client *i2c, else max98373->interleave_mode = false; - /* regmap initialization */ max98373->regmap = devm_regmap_init_i2c(i2c, &max98373_regmap); @@ -934,6 +945,24 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + /* voltage/current slot & gpio configuration */ + max98373_slot_config(i2c, max98373); + + /* Power on device */ + if (gpio_is_valid(max98373->reset_gpio)) { + ret = gpio_request(max98373->reset_gpio, "MAX98373_RESET"); + if (ret) { + dev_err(&i2c->dev, "%s: Failed to request gpio %d\n", + __func__, max98373->reset_gpio); + gpio_free(max98373->reset_gpio); + return -EINVAL; + } + gpio_direction_output(max98373->reset_gpio, 0); + msleep(50); + gpio_direction_output(max98373->reset_gpio, 1); + msleep(20); + } + /* Check Revision ID */ ret = regmap_read(max98373->regmap, MAX98373_R21FF_REV_ID, &reg); @@ -944,9 +973,6 @@ static int max98373_i2c_probe(struct i2c_client *i2c, } dev_info(&i2c->dev, "MAX98373 revisionID: 0x%02X\n", reg); - /* voltage/current slot configuration */ - max98373_slot_config(i2c, max98373); - /* codec registeration */ ret = devm_snd_soc_register_component(&i2c->dev, &soc_codec_dev_max98373, max98373_dai, ARRAY_SIZE(max98373_dai)); diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index f6a37aa02f26..533d2053f608 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,7 @@ struct max98373_priv { struct regmap *regmap; + int reset_gpio; unsigned int v_slot; unsigned int i_slot; unsigned int spkfb_slot; -- 2.18.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt1308: simplify the EFUSE read function
by shumingf＠realtek.com 22 Jul '19

22 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> The rt1308_efuse function contains many redundant settings. We remove the redundant settings and the function still works. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308.c | 22 ---------------------- 1 file changed, 22 deletions(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index d673506c7c39..19d255e01827 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -808,33 +808,11 @@ static void rt1308_efuse(struct rt1308_priv *rt1308) { regmap_write(rt1308->regmap, RT1308_RESET, 0); - regmap_write(rt1308->regmap, RT1308_POWER, 0xff371600); - regmap_write(rt1308->regmap, RT1308_CLK_1, 0x52100000); - regmap_write(rt1308->regmap, RT1308_I2C_I2S_SDW_SET, 0x01014005); - regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x227f5501); - regmap_write(rt1308->regmap, RT1308_PADS_1, 0x50150505); - regmap_write(rt1308->regmap, RT1308_VREF, 0x18100000); - regmap_write(rt1308->regmap, RT1308_IV_SENSE, 0x87010000); - regmap_write(rt1308->regmap, RT1308_DUMMY_REG, 0x00000200); - regmap_write(rt1308->regmap, RT1308_SIL_DET, 0x61c30000); - regmap_write(rt1308->regmap, RT1308_CLK_DET, 0x03700000); - regmap_write(rt1308->regmap, RT1308_SINE_TONE_GEN_1, 0x50022f00); regmap_write(rt1308->regmap, RT1308_POWER_STATUS, 0x01800000); - regmap_write(rt1308->regmap, RT1308_DC_CAL_2, 0x00ffff00); - regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x607e5501); - - regmap_write(rt1308->regmap, RT1308_CLK_2, 0x0060e000); - regmap_write(rt1308->regmap, RT1308_EFUSE_1, 0x04fe0f00); msleep(100); regmap_write(rt1308->regmap, RT1308_EFUSE_1, 0x44fe0f00); msleep(20); regmap_write(rt1308->regmap, RT1308_PVDD_OFFSET_CTL, 0x10000000); - - regmap_write(rt1308->regmap, RT1308_POWER_STATUS, 0x00800000); - regmap_write(rt1308->regmap, RT1308_POWER, 0x0); - regmap_write(rt1308->regmap, RT1308_CLK_1, 0x52000000); - regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x227f5501); - regmap_write(rt1308->regmap, RT1308_SINE_TONE_GEN_1, 0x10022f00); } static int rt1308_i2c_probe(struct i2c_client *i2c, -- 2.22.0

2 1

[alsa-devel] [PATCH 2/3] ASoC: rt1308: add silence detection and manual PDB control
by shumingf＠realtek.com 22 Jul '19

22 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> We enable the silence detection function in initial settings. PDB control changes to manual mode, hence the driver could fully control the AMP output on/off. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 19d255e01827..cf3d72f99705 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -40,10 +40,10 @@ static const struct reg_sequence init_list[] = { { RT1308_VREF, 0x18100000 }, { RT1308_IV_SENSE, 0x87010000 }, { RT1308_DUMMY_REG, 0x00000200 }, - { RT1308_SIL_DET, 0x61c30000 }, + { RT1308_SIL_DET, 0xe1c30000 }, { RT1308_DC_CAL_2, 0x00ffff00 }, { RT1308_CLK_DET, 0x01000000 }, - { RT1308_POWER_STATUS, 0x00800000 }, + { RT1308_POWER_STATUS, 0x08800000 }, { RT1308_DAC_SET, 0xafaf0700 }, }; @@ -308,12 +308,13 @@ static int rt1308_classd_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: msleep(30); snd_soc_component_update_bits(component, RT1308_POWER_STATUS, - RT1308_POW_PDB_REG_BIT, RT1308_POW_PDB_REG_BIT); + RT1308_POW_PDB_REG_BIT | RT1308_POW_PDB_MN_BIT, + RT1308_POW_PDB_REG_BIT | RT1308_POW_PDB_MN_BIT); msleep(40); break; case SND_SOC_DAPM_PRE_PMD: snd_soc_component_update_bits(component, RT1308_POWER_STATUS, - RT1308_POW_PDB_REG_BIT, 0); + RT1308_POW_PDB_REG_BIT | RT1308_POW_PDB_MN_BIT, 0); usleep_range(150000, 200000); break; -- 2.22.0

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: suppress error message for EPROBE_DEFER
by Lucas Stach 22 Jul '19

22 Jul '19

Both the supplies and reset GPIO might need a probe deferral for the resource to be available. Don't print a error message in that case, as it is a normal operating condition. Signed-off-by: Lucas Stach <l.stach(a)pengutronix.de> --- sound/soc/codecs/tlv320aic31xx.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 9b37e98da0db..26a4f6cd3288 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1553,7 +1553,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx->gpio_reset = devm_gpiod_get_optional(aic31xx->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(aic31xx->gpio_reset)) { - dev_err(aic31xx->dev, "not able to acquire gpio\n"); + if (PTR_ERR(aic31xx->gpio_reset) != -EPROBE_DEFER) + dev_err(aic31xx->dev, "not able to acquire gpio\n"); return PTR_ERR(aic31xx->gpio_reset); } @@ -1564,7 +1565,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, ARRAY_SIZE(aic31xx->supplies), aic31xx->supplies); if (ret) { - dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(aic31xx->dev, + "Failed to request supplies: %d\n", ret); return ret; } -- 2.20.1

3 2

[alsa-devel] [PATCH] SoC: rockchip-max98090: Remove MICBIAS as supply of input pin IN34
by Enric Balletbo i Serra 22 Jul '19

22 Jul '19

Commit ec0d23b295b9 ("ASoC: rockchip-max98090: Fix the Headset Mic route.") moved the MICBIAS widget to supply Headset Mic but forget to remove the MICBIAS widget to supply IN34 which is not really needed, so remove that path so we have: IN34 <==== Headset MIC <==== MICBIAS Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/rockchip/rockchip_max98090.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 782e534d4c0d..1af1147c3da3 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -45,7 +45,6 @@ static const struct snd_soc_dapm_widget rk_dapm_widgets[] = { static const struct snd_soc_dapm_route rk_audio_map[] = { {"IN34", NULL, "Headset Mic"}, - {"IN34", NULL, "MICBIAS"}, {"Headset Mic", NULL, "MICBIAS"}, {"DMICL", NULL, "Int Mic"}, {"Headphone", NULL, "HPL"}, -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: cs47l35: Use define for DSPCLK enable bit
by Charles Keepax 22 Jul '19

22 Jul '19

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs47l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index e3585c1dab3d7..d396a8545d518 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -524,7 +524,7 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", MADERA_SYSTEM_CLOCK_1, MADERA_SYSCLK_ENA_SHIFT, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("OPCLK", MADERA_OUTPUT_SYSTEM_CLOCK, MADERA_OPCLK_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, 6, +SND_SOC_DAPM_SUPPLY("DSPCLK", MADERA_DSP_CLOCK_1, MADERA_DSP_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), -- 2.11.0

2 1

Re: [alsa-devel] [PATCH 3/3] ASoC: rt1308: Convert headers to SPDX
by Mark Brown 22 Jul '19

22 Jul '19

On Mon, Jul 22, 2019 at 02:07:47AM +0000, Shuming [范書銘] wrote: > > On Fri, Jul 19, 2019 at 02:33:02PM +0800, shumingf(a)realtek.com wrote: > > > @@ -1,12 +1,10 @@ > > > +// SPDX-License-Identifier: GPL-2.0 > > > /* > > > * rt1308.c -- RT1308 ALSA SoC amplifier component driver > > > * > > Please convert the entire comment block to a C++ one so this looks more > > intentional. > I convert the comment like below. > -// SPDX-License-Identifier: GPL-2.0 > +/* SPDX-License-Identifier: GPL-2.0 */ That's a C comment, not a C++ comment. I'm saying convert the whole block there to C++ rather than add one random line that's C++ next to some C comments.

2 3

[alsa-devel] [PATCH] ALSA: line6: Fix a typo
by Christophe JAILLET 22 Jul '19

22 Jul '19

s/Vairax/Variax/ Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/usb/line6/variax.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index 0d24c72c155f..ed158f04de80 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -244,5 +244,5 @@ static struct usb_driver variax_driver = { module_usb_driver(variax_driver); -MODULE_DESCRIPTION("Vairax Workbench USB driver"); +MODULE_DESCRIPTION("Variax Workbench USB driver"); MODULE_LICENSE("GPL"); -- 2.20.1

2 1

[alsa-devel] [PATCH] slimbus: fix duplicated argument to ||
by Hariprasad Kelam 21 Jul '19

21 Jul '19

Remove duplicate argument SLIM_MSG_MC_REQUEST_CLEAR_INFORMATION. fix below issue reported by coccicheck ./drivers/slimbus/slimbus.h:440:3-46: duplicated argument to && or || Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- drivers/slimbus/slimbus.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 9be4108..46a6441 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -438,8 +438,7 @@ static inline bool slim_tid_txn(u8 mt, u8 mc) return (mt == SLIM_MSG_MT_CORE && (mc == SLIM_MSG_MC_REQUEST_INFORMATION || mc == SLIM_MSG_MC_REQUEST_CLEAR_INFORMATION || - mc == SLIM_MSG_MC_REQUEST_VALUE || - mc == SLIM_MSG_MC_REQUEST_CLEAR_INFORMATION)); + mc == SLIM_MSG_MC_REQUEST_VALUE)); } static inline bool slim_ec_txn(u8 mt, u8 mc) -- 2.7.4

2 1

[alsa-devel] [PATCH 0/6] ALSA/HDA: abort probe when DMICs are detected
by Pierre-Louis Bossart 19 Jul '19

19 Jul '19

This is the second take on same problem of detecting when the HDaudio legacy driver can be used and when the SST or SOF drivers are required. The previous attempt based on a the PCI Class information was a resounding failure and broke audio on Linus' laptop, so we need additional information to avoid enabling a DSP-based driver without a good reason to do so. This patchset suggests the use of the NHLT information which *in theory* exposes DMIC endpoints. The legacy HDaudio driver cannot handle DMICs and will not provide any capture capabilities. Since it's typically the first one to probe due to the Makefile order, aborting the probe will let the PCI subsystem look for the next driver which hopefully will support this capability. I tested this patch on 5 devices (SKL, KBL, APL, GLK, WHL), three without DMICs and two with, and the detection seems to work as planned. Additional testing by Canonical and Endless folks did not expose any issues. Changes since RFC: Cosmetic code improvements Moved intel-nhlt.h to include/sound Moved intel-nhlt.c to sound/hda Removed SOC prefixes Added full-support for vendor-defined geometries Added Kconfig and kernel module parameter to opt-in Pierre-Louis Bossart (6): ASoC: Intel: Skylake: move NHLT header to common directory ALSA: hda: move parts of NHLT code to new module ALSA: hda: intel-nhlt: remove useless OR operation ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry ASoC: Intel: Skylake: use common NHLT module ALSA: hda/intel: stop probe if DMICS are detected on Skylake+ platforms .../skl-nhlt.h => include/sound/intel-nhlt.h | 51 +++++++-- sound/hda/Kconfig | 3 + sound/hda/Makefile | 3 + sound/hda/intel-nhlt.c | 102 ++++++++++++++++++ sound/pci/hda/Kconfig | 10 ++ sound/pci/hda/hda_intel.c | 34 ++++++ sound/soc/intel/Kconfig | 1 + sound/soc/intel/skylake/skl-nhlt.c | 91 +--------------- sound/soc/intel/skylake/skl-ssp-clk.c | 1 + sound/soc/intel/skylake/skl-topology.c | 1 + sound/soc/intel/skylake/skl.c | 12 ++- sound/soc/intel/skylake/skl.h | 4 - 12 files changed, 205 insertions(+), 108 deletions(-) rename sound/soc/intel/skylake/skl-nhlt.h => include/sound/intel-nhlt.h (65%) create mode 100644 sound/hda/intel-nhlt.c -- 2.20.1

2 13

[alsa-devel] [PATCH 3/3] ASoC: rt1308: Convert headers to SPDX
by shumingf＠realtek.com 19 Jul '19

19 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> Convert include and source files to SPDX. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308.c | 4 +--- sound/soc/codecs/rt1308.h | 6 ++---- 2 files changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index cf3d72f99705..694ea22f25a8 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -1,12 +1,10 @@ +// SPDX-License-Identifier: GPL-2.0 /* * rt1308.c -- RT1308 ALSA SoC amplifier component driver * * Copyright 2019 Realtek Semiconductor Corp. * Author: Derek Fang <derek.fang(a)realtek.com> * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #include <linux/module.h> diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h index c330aae1d527..ff7c423e879e 100644 --- a/sound/soc/codecs/rt1308.h +++ b/sound/soc/codecs/rt1308.h @@ -1,12 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * RT1308.h -- RT1308 ALSA SoC amplifier component driver + * rt1308.h -- RT1308 ALSA SoC amplifier component driver * * Copyright 2019 Realtek Semiconductor Corp. * Author: Derek Fang <derek.fang(a)realtek.com> * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef _RT1308_H_ -- 2.22.0

2 1

[alsa-devel] [PATCH V6 RESEND] ALSA: usb-audio: Scarlett Gen 2 mixer interface
by Geoffrey D. Bennett 19 Jul '19

19 Jul '19

Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2 audio interfaces. Although the interfaces are USB compliant, additional input/output level controls and hardware routing/mixing functionality are available using proprietary USB requests. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- Hi all, This patch adds the following controls for the Scarlett 6i6, 18i8, and 18i20 Gen 2: - Master volume knob indicator (18i20 only) - Mute and dim switches (18i20 only) - Volume controls for the analogue HW outputs - HW/SW volume switches for the 10 analogue HW outputs (18i20 only) - Line Level/Instrument Level and Pad controls (6i6 and 18i8 only) - Output mux (where the sound for the HW outputs comes from; defaults to PCM outputs) - Capture mux (where the sound for PCM recording comes from; defaults to HW inputs) - Matrix mux (where the sound going into the mixer comes from; 18 inputs default off) - Mixer matrix (18 inputs * 10 outputs = 180 controls) - Level meters Changes since v1: - Add support for the Scarlett 18i8 Gen 2 - Save configuration parameters to NVRAM - Implemented feedback from Takashi's email 24/Apr/2019 - Moved private field from struct snd_usb_audio to struct usb_mixer_interface - Added timer and buffer fields to struct usb_mixer_interface - Other small code fixes/cleanups/improvements Changes since v2: - Add support for the Scarlett 6i6 Gen 2 - Add support for controlling the mute/dim switches (18i20 only) - Implemented feedback from Takashi's email 29/Apr/2019: One pointer field private_data and one function pointer private_free. Replaced timer with delayed work (thanks for that suggestion; it makes the code a *lot* simpler!). - The new functionality is disabled by default as there were reports of some Gen 2 devices not responding to the initialisation sequence and hanging. Added module parameter "scarlett_gen2_mixer_enable". If this parameter is not set, logs a message "Focusrite Scarlett Gen 2 Mixer Driver disabled; use options snd_usb_audio scarlett_gen2_mixer_enable=1 to enable and report any issues to g(a)b4.vu". Changes since v3: - No functional changes; fix checkpatch.pl error and typo in declaration. Changes since v4: - Implemented feedback from Takashi's email 9/Jul/2019: cancel delayed work on suspend, drop copyright texts, remove unnecessary blank lines, and run kmalloc outside the mutex. Changes since v5: - Implemented feedback from Takashi's emails 9-10/Jul/2019: use chip->setup value (device_setup option) instead of the scarlett_gen2_mixer_enable option and sync config to NVRAM immediately on suspend. - Fixed issue where the config save delayed work would be cancelled and not restarted if a scarlett2_*_ctl_put() function was called and the parameter value was not changed. - Renaming definitions and functions for consistency and move private free/suspend functions above their use so pre-declaration isn't necessary. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e1ce257ab705..f4b7384caa9d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,6 +11,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c703f8534b07..734846faee7a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3505,6 +3505,8 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) usb_kill_urb(mixer->urb); if (mixer->rc_urb) usb_kill_urb(mixer->rc_urb); + if (mixer->private_free) + mixer->private_free(mixer); mixer->disconnected = true; } @@ -3532,6 +3534,8 @@ static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) { snd_usb_mixer_inactivate(mixer); + if (mixer->private_suspend) + mixer->private_suspend(mixer); return 0; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 394cd9107507..37e1b234c802 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -28,6 +28,10 @@ struct usb_mixer_interface { struct media_mixer_ctl *media_mixer_ctl; bool disconnected; + + void *private_data; + void (*private_free)(struct usb_mixer_interface *mixer); + void (*private_suspend)(struct usb_mixer_interface *mixer); }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..70805c15b0b5 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -32,6 +32,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -2258,6 +2259,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */ + case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..a772e21bddaa --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,2076 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 6i6/18i8/18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu> + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + */ + +/* Mixer Interface for the Focusrite Scarlett 6i6/18i8/18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * Scarlett 18i8 support added in April 2019. + * + * Scarlett 6i6 support added in June 2019 (thanks to Martin Wittmann + * for providing usbmon output and testing). + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * - line/inst level and pad controls + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ Software gain per channel. + * | Master Gain |<-- 18i20 only: Switch per channel + * +------+------+ to select HW or SW gain control. + * | + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +/* device_setup value to enable */ +#define SCARLETT2_ENABLE 0x01 + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT2_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT2_MIXER_MIN_DB -80 +#define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2) +#define SCARLETT2_MIXER_MAX_DB 6 +#define SCARLETT2_MIXER_MAX_VALUE \ + ((SCARLETT2_MIXER_MAX_DB - SCARLETT2_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_ANALOGUE_MAX 10 + +/* Maximum number of level and pad switches */ +#define SCARLETT2_LEVEL_SWITCH_MAX 2 +#define SCARLETT2_PAD_SWITCH_MAX 4 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +/* Hardware buttons on the 18i20 */ +#define SCARLETT2_BUTTON_MAX 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + u8 line_out_hw_vol; /* line out hw volume is sw controlled */ + u8 button_count; /* number of buttons */ + u8 level_input_count; /* inputs with level selectable */ + u8 pad_input_count; /* inputs with pad selectable */ + const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_mixer_data { + struct usb_mixer_interface *mixer; + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + struct delayed_work work; + const struct scarlett2_device_info *info; + int num_mux_srcs; + u16 scarlett2_seq; + u8 vol_updated; + u8 master_vol; + u8 vol[SCARLETT2_ANALOGUE_MAX]; + u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; + u8 buttons[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + u8 mux[SCARLETT2_MUX_MAX]; + u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s6i6_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first two analogue inputs have an optional pad. */ + .pad_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 8 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 4, 4, 4, 4, 4 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 6, 6, 6, 6, 6 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i8_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first four analogue inputs have an optional pad. */ + .pad_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 4 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 6, 6, 6, 6 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 0, 0, 0, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i20_gen2_info = { + /* The analogue line outputs on the 18i20 can be switched + * between software and hardware volume control + */ + .line_out_hw_vol = 1, + + /* Mute and dim buttons */ + .button_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_CONFIG_SAVE 6 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* volume status is read together (matches scarlett2_config_items[]) */ +struct scarlett2_usb_volume_status { + /* mute & dim buttons */ + u8 buttons[SCARLETT2_BUTTON_MAX]; + + u8 pad1; + + /* software volume setting */ + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + + /* actual volume of output inc. dim (-18dB) */ + s16 hw_vol[SCARLETT2_ANALOGUE_MAX]; + + u8 pad2[SCARLETT2_ANALOGUE_MAX]; + + /* sw (0) or hw (1) controlled */ + u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + + u8 pad3[6]; + + /* front panel volume knob */ + s16 master_vol; +} __packed; + +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, + SCARLETT2_CONFIG_SW_HW_SWITCH = 2, + SCARLETT2_CONFIG_LEVEL_SWITCH = 3, + SCARLETT2_CONFIG_PAD_SWITCH = 4, + SCARLETT2_CONFIG_COUNT = 5 +}; + +/* Location, size, and activation command number for the configuration + * parameters + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +static const struct scarlett2_config + scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { + /* Mute/Dim Buttons */ + { + .offset = 0x31, + .size = 1, + .activate = 2 + }, + + /* Line Out Volume */ + { + .offset = 0x34, + .size = 2, + .activate = 1 + }, + + /* SW/HW Volume Switch */ + { + .offset = 0x66, + .size = 1, + .activate = 3 + }, + + /* Level Switch */ + { + .offset = 0x7c, + .size = 1, + .activate = 7 + }, + + /* Pad Switch */ + { + .offset = 0x84, + .size = 1, + .activate = 8 + } +}; + +/* proprietary request/response format */ +struct scarlett2_usb_packet { + u32 cmd; + u16 size; + u16 seq; + u32 error; + u32 pad; + u8 data[]; +}; + +#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) + +static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, + struct scarlett2_usb_packet *req, + u32 cmd, u16 req_size) +{ + /* sequence must go up by 1 for each request */ + u16 seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; +} + +/* Send a proprietary format request to the Scarlett interface */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, u32 cmd, + void *req_data, u16 req_size, void *resp_data, u16 resp_size) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; + u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; + struct scarlett2_usb_packet *req = 0, *resp = 0; + int err = 0; + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + err = -ENOMEM; + goto error; + } + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + err = -ENOMEM; + goto error; + } + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + scarlett2_fill_request_header(private, req, cmd, req_size); + + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + if (err != req_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB request result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + /* send a second message to get the response */ + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB response result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + if (resp->cmd != req->cmd || + resp->seq != req->seq || + resp_size != le16_to_cpu(resp->size) || + resp->error || + resp->pad) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd), + le16_to_cpu(req->seq), le16_to_cpu(resp->seq), + resp_size, le16_to_cpu(resp->size), + le16_to_cpu(resp->error), + le16_to_cpu(resp->pad)); + err = -EINVAL; + goto unlock; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + +unlock: + mutex_unlock(&private->usb_mutex); +error: + kfree(req); + kfree(resp); + return err; +} + +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct usb_mixer_interface *mixer) +{ + u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); + + scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(u32), + NULL, 0); +} + +/* Delayed work to save config */ +static void scarlett2_config_save_work(struct work_struct *work) +{ + struct scarlett2_mixer_data *private = + container_of(work, struct scarlett2_mixer_data, work.work); + + scarlett2_config_save(private->mixer); +} + +/* Send a USB message to set a configuration parameter (volume level, + * sw/hw volume switch, line/inst level switch, or pad switch) + */ +static int scarlett2_usb_set_config( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int value) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + struct { + u32 offset; + u32 bytes; + s32 value; + } __packed req; + u32 req2; + int err; + struct scarlett2_mixer_data *private = mixer->private_data; + + /* Cancel any pending NVRAM save */ + cancel_delayed_work_sync(&private->work); + + /* Send the configuration parameter data */ + req.offset = cpu_to_le32(config_item.offset + index * config_item.size); + req.bytes = cpu_to_le32(config_item.size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + config_item.size, + NULL, 0); + if (err < 0) + return err; + + /* Activate the change */ + req2 = cpu_to_le32(config_item.activate); + err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + /* Schedule the change to be written to NVRAM */ + schedule_delayed_work(&private->work, msecs_to_jiffies(2000)); + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + int offset, void *buf, int size) +{ + struct { + u32 offset; + u32 size; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA, + &req, sizeof(req), buf, size); +} + +/* Send a USB message to get configuration parameters; result placed in *buf */ +static int scarlett2_usb_get_config( + struct usb_mixer_interface *mixer, + int config_item_num, int count, void *buf) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + int size = config_item.size * count; + + return scarlett2_usb_get(mixer, config_item.offset, buf, size); +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + struct { + u16 mix_num; + u16 data[SCARLETT2_INPUT_MIX_MAX]; + } __packed req; + + int i, j; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(u16), + NULL, 0); +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct { + u16 pad; + u16 num; + u32 data[SCARLETT2_MUX_MAX]; + } __packed req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + u16 *levels) +{ + struct { + u16 pad; + u16 num_meters; + u32 magic; + } __packed req; + u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* Update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < private->info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol[index] = val; + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, val - SCARLETT2_VOLUME_BIAS); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett2_gain, -SCARLETT2_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, private->master_vol - SCARLETT2_VOLUME_BIAS); + if (err < 0) + goto unlock; + + /* Notify of RO/RW change */ + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Line Level/Instrument Level Switch Controls ***/ + +static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Line", "Inst" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->level_switch[elem->control]; + return 0; +} + +static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->level_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->level_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_level_enum_ctl_info, + .get = scarlett2_level_enum_ctl_get, + .put = scarlett2_level_enum_ctl_put, +}; + +/*** Pad Switch Controls ***/ + +static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->pad_switch[elem->control]; + return 0; +} + +static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->pad_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pad_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_pad_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_pad_ctl_get, + .put = scarlett2_pad_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->buttons[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->buttons[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get, + .put = scarlett2_button_ctl_put +}; + +/*** Create the analogue output controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), "Master HW Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_master_volume_ctl, + 0, 1, s, &private->master_vol_ctl); + if (err < 0) + return err; + } + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(s, sizeof(s), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(s, sizeof(s), + "Line %02d Playback Volume", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_line_out_volume_ctl, + i, 1, s, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_sw_hw_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + /* Add HW button controls */ + for (i = 0; i < private->info->button_count; i++) { + err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Create the analogue input controls ***/ + +static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add input level (line/inst) controls */ + for (i = 0; i < info->level_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + /* Add input pad controls */ + for (i = 0; i < info->pad_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) + goto unlock; + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(mixer, mix_num); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett2_mixer, + SCARLETT2_MIXER_MIN_DB * 100, + SCARLETT2_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT2_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett2_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(s, sizeof(s), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) + goto unlock; + + private->mux[index] = val; + err = scarlett2_usb_set_mux(mixer); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(s, sizeof(s) - 5, descr, channel + 1); + strcat(s, " Enum"); + + err = scarlett2_add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, + "Level Meter", NULL); +} + +/*** Cleanup/Suspend Callbacks ***/ + +static void scarlett2_private_free(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + cancel_delayed_work_sync(&private->work); + kfree(private); + mixer->private_data = NULL; +} + +static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + if (cancel_delayed_work_sync(&private->work)) + scarlett2_config_save(private->mixer); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_mixer_data *private = + kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + private->mixer = mixer; + mixer->private_data = private; + mixer->private_free = scarlett2_private_free; + mixer->private_suspend = scarlett2_private_suspend; + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, + NULL, 0, NULL, 0); +} + +/* Read line-in config and line-out volume settings on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + if (info->level_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count, + level_switches); + if (err < 0) + return err; + for (i = 0; i < info->level_input_count; i++) + private->level_switch[i] = level_switches[i]; + } + + if (info->pad_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, + pad_switches); + if (err < 0) + return err; + for (i = 0; i < info->pad_input_count; i++) + private->pad_switch[i] = pad_switches[i]; + } + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = + info->line_out_hw_vol + && volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + int i; + + private->vol_updated = 1; + + for (i = 0; i < private->info->button_count; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + usb_audio_err(mixer->chip, + "scarlett mixer interrupt length %d\n", len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + void *transfer_buffer; + + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb( + mixer->urb, + mixer->chip->dev, + usb_rcvintpipe(mixer->chip->dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT), + transfer_buffer, + SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, + mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8203): + info = &s6i6_gen2_info; + break; + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + if (!(mixer->chip->setup & SCARLETT2_ENABLE)) { + usb_audio_err(mixer->chip, + "Focusrite Scarlett Gen 2 Mixer Driver disabled; " + "use options snd_usb_audio device_setup=1 " + "to enable and report any issues to g(a)b4.vu"); + return 0; + } + + /* Initialise private data, routing, sequence number */ + err = scarlett2_init_private(mixer, info); + if (err < 0) + return err; + + /* Read volume levels and controls from the interface */ + err = scarlett2_read_configs(mixer); + if (err < 0) + return err; + + /* Create the analogue output controls */ + err = scarlett2_add_line_out_ctls(mixer); + if (err < 0) + return err; + + /* Create the analogue input controls */ + err = scarlett2_add_line_in_ctls(mixer); + if (err < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + err = scarlett2_add_mux_enums(mixer); + if (err < 0) + return err; + + /* Create the matrix mixer controls */ + err = scarlett2_add_mixer_ctls(mixer); + if (err < 0) + return err; + + /* Create the level meter controls */ + err = scarlett2_add_meter_ctl(mixer); + if (err < 0) + return err; + + /* Set up the interrupt polling if there are hardware buttons */ + if (info->button_count) { + err = scarlett2_mixer_status_create(mixer); + if (err < 0) + return err; + } + + return 0; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */

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[alsa-devel] [PATCH v3 1/2] ASoC: SOF: add flag for position update ipc
by Keyon Jie 19 Jul '19

19 Jul '19

From: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> In some cases, FW might need to use the host_period_bytes field to synchronize the DMA copying (with host side) but the driver does not need any position information returned over the IPC channel by the firmware. The current IPC definition prevents this capability, so add new field. Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- include/sound/sof/stream.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h index 643f175cb479..06af4ecb2584 100644 --- a/include/sound/sof/stream.h +++ b/include/sound/sof/stream.h @@ -83,10 +83,10 @@ struct sof_ipc_stream_params { uint16_t sample_valid_bytes; uint16_t sample_container_bytes; - /* for notifying host period has completed - 0 means no period IRQ */ uint32_t host_period_bytes; + uint32_t no_position_ipc; /* 1 means no IPC for position upadte */ - uint32_t reserved[2]; + uint16_t reserved[3]; uint16_t chmap[SOF_IPC_MAX_CHANNELS]; /**< channel map - SOF_CHMAP_ */ } __packed; -- 2.20.1

3 8

[alsa-devel] [PATCH] ALSA: pcm: Fix refcount_inc() on zero usage
by Takashi Iwai 19 Jul '19

19 Jul '19

The recent rewrite of PCM link lock management introduced the refcount in snd_pcm_group object, managed by the kernel refcount_t API. This caused unexpected kernel warnings when the kernel is built with CONFIG_REFCOUNT_FULL=y. As the warning line indicates, the problem is obviously that we start with refcount=0 and do refcount_inc() for adding each PCM link, while refcount_t API doesn't like refcount_inc() performed on zero. For adapting the proper refcount_t usage, this patch changes the logic slightly: - The initial refcount is 1, assuming the single list entry - The refcount is incremented / decremented at each PCM link addition and deletion - ... which allows us concentrating only on the refcount as a release condition Fixes: f57f3df03a8e ("ALSA: pcm: More fine-grained PCM link locking") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204221 Reported-and-tested-by: Duncan Overbruck <kernel(a)duncano.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 860543a4c840..12dd9b318db1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -77,7 +77,7 @@ void snd_pcm_group_init(struct snd_pcm_group *group) spin_lock_init(&group->lock); mutex_init(&group->mutex); INIT_LIST_HEAD(&group->substreams); - refcount_set(&group->refs, 0); + refcount_set(&group->refs, 1); } /* define group lock helpers */ @@ -1096,8 +1096,7 @@ static void snd_pcm_group_unref(struct snd_pcm_group *group, if (!group) return; - do_free = refcount_dec_and_test(&group->refs) && - list_empty(&group->substreams); + do_free = refcount_dec_and_test(&group->refs); snd_pcm_group_unlock(group, substream->pcm->nonatomic); if (do_free) kfree(group); @@ -2020,6 +2019,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) snd_pcm_group_lock_irq(target_group, nonatomic); snd_pcm_stream_lock(substream1); snd_pcm_group_assign(substream1, target_group); + refcount_inc(&target_group->refs); snd_pcm_stream_unlock(substream1); snd_pcm_group_unlock_irq(target_group, nonatomic); _end: @@ -2056,13 +2056,14 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) snd_pcm_group_lock_irq(group, nonatomic); relink_to_local(substream); + refcount_dec(&group->refs); /* detach the last stream, too */ if (list_is_singular(&group->substreams)) { relink_to_local(list_first_entry(&group->substreams, struct snd_pcm_substream, link_list)); - do_free = !refcount_read(&group->refs); + do_free = refcount_dec_and_test(&group->refs); } snd_pcm_group_unlock_irq(group, nonatomic); -- 2.16.4

1 0

[alsa-devel] [PATCH v6 0/6] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 19 Jul '19

19 Jul '19

VAG power control is improved to fit the manual [1]. This patchset fixes as minimum one bug: if customer muxes Headphone to Line-In right after boot, the VAG power remains off that leads to poor sound quality from line-in. I.e. after boot: - Connect sound source to Line-In jack; - Connect headphone to HP jack; - Run following commands: $ amixer set 'Headphone' 80% $ amixer set 'Headphone Mux' LINE_IN Also this series includes fixes of non-important bugs in sgtl5000 codec driver. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Changes in v6: - Code optimization Changes in v5: - Add explicit stable tag - Improve commit message Changes in v4: - CC the patch to kernel-stable - Code optimization, simplify function signature (thanks to Cezary Rojewski <cezary.rojewski(a)intel.com> for an idea) - Add a Fixes tag Changes in v3: - Add the reference to NXP SGTL5000 data sheet to commit message - Fix multi-line comment format Changes in v2: - Fix patch formatting Oleksandr Suvorov (6): ASoC: Define a set of DAPM pre/post-up events ASoC: sgtl5000: Improve VAG power and mute control ASoC: sgtl5000: Fix definition of VAG Ramp Control ASoC: sgtl5000: add ADC mute control ASoC: sgtl5000: Fix of unmute outputs on probe ASoC: sgtl5000: Fix charge pump source assignment include/sound/soc-dapm.h | 2 + sound/soc/codecs/sgtl5000.c | 250 ++++++++++++++++++++++++++++++------ sound/soc/codecs/sgtl5000.h | 2 +- 3 files changed, 213 insertions(+), 41 deletions(-) -- 2.20.1

2 7

[alsa-devel] [PATCH v5 0/6] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 19 Jul '19

19 Jul '19

VAG power control is improved to fit the manual [1]. This patchset fixes as minimum one bug: if customer muxes Headphone to Line-In right after boot, the VAG power remains off that leads to poor sound quality from line-in. I.e. after boot: - Connect sound source to Line-In jack; - Connect headphone to HP jack; - Run following commands: $ amixer set 'Headphone' 80% $ amixer set 'Headphone Mux' LINE_IN Also this series includes fixes of non-important bugs in sgtl5000 codec driver. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Changes in v5: - Add explicit stable tag - Improve commit message - Add explicit stable tag Changes in v4: - CC the patch to kernel-stable - Code optimization, simplify function signature (thanks to Cezary Rojewski <cezary.rojewski(a)intel.com> for an idea) - CC the patch to kernel-stable - Add a Fixes tag Changes in v3: - Add the reference to NXP SGTL5000 data sheet to commit message - Add the reference to NXP SGTL5000 data sheet to commit message - Fix multi-line comment format Changes in v2: - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting Oleksandr Suvorov (6): ASoC: Define a set of DAPM pre/post-up events ASoC: sgtl5000: Improve VAG power and mute control ASoC: sgtl5000: Fix definition of VAG Ramp Control ASoC: sgtl5000: add ADC mute control ASoC: sgtl5000: Fix of unmute outputs on probe ASoC: sgtl5000: Fix charge pump source assignment include/sound/soc-dapm.h | 2 + sound/soc/codecs/sgtl5000.c | 240 ++++++++++++++++++++++++++++++------ sound/soc/codecs/sgtl5000.h | 2 +- 3 files changed, 203 insertions(+), 41 deletions(-) -- 2.20.1

3 11

[alsa-devel] [RFC PATCH] Docs: move more driver, device related docs into drivers dir follows kernel source
by Alex Shi 19 Jul '19

19 Jul '19

Could I have a proposal to move the following driver/devices dirs which under Documentation/ into Documentation/drivers/, follows kernel source acpi backlight bus cpu-freq device-mapper driver-model gpio hid ide leds md memory mmc nfc nvmem pcmcia scsi sound watchdog auxdisplay blockdev cdrom crypto driver-api fpga gpu i2c infiniband lightnvm media misc mtd nvdimm PCI apidio serial usb More docs and dirs could move into this dir later, that could make Documenation dir much more clear, and don't cause trouble to find them. If it's acceptable, we could do docs pointer redirect further. Signed-off-by: Alex Shi <alex.shi(a)linux.alibaba.com> Cc: linux-pci(a)vger.kernel.org Cc: linux-doc(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: drbd-dev(a)lists.linbit.com Cc: linux-block(a)vger.kernel.org Cc: nbd(a)other.debian.org Cc: linux-ide(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: keyrings(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linaro-mm-sig(a)lists.linaro.org Cc: dmaengine(a)vger.kernel.org Cc: linux-edac(a)vger.kernel.org Cc: linux-fpga(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-gpio(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-s390(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-usb(a)vger.kernel.org Cc: devel(a)driverdev.osuosl.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-amlogic(a)lists.infradead.org Cc: linux-tegra(a)vger.kernel.org Cc: intel-gfx(a)lists.freedesktop.org Cc: xen-devel(a)lists.xenproject.org Cc: linux-input(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-rdma(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: dc395x(a)twibble.org Cc: esc.storagedev(a)microsemi.com Cc: megaraidlinux.pdl(a)broadcom.com Cc: osst-users(a)lists.sourceforge.net Cc: linux-watchdog(a)vger.kernel.org --- Documentation/{ => drivers}/PCI/MSI-HOWTO.txt | 0 Documentation/{ => drivers}/PCI/PCIEBUS-HOWTO.txt | 0 Documentation/{ => drivers}/PCI/acpi-info.txt | 0 .../{ => drivers}/PCI/endpoint/function/binding/pci-test.txt | 0 Documentation/{ => drivers}/PCI/endpoint/pci-endpoint-cfs.txt | 0 Documentation/{ => drivers}/PCI/endpoint/pci-endpoint.txt | 0 Documentation/{ => drivers}/PCI/endpoint/pci-test-function.txt | 0 Documentation/{ => drivers}/PCI/endpoint/pci-test-howto.txt | 0 Documentation/{ => drivers}/PCI/pci-error-recovery.txt | 0 Documentation/{ => drivers}/PCI/pci-iov-howto.txt | 0 Documentation/{ => drivers}/PCI/pci.txt | 0 Documentation/{ => drivers}/PCI/pcieaer-howto.txt | 0 Documentation/{ => drivers}/acpi/dsd/leds.txt | 0 Documentation/{ => drivers}/auxdisplay/cfag12864b | 0 Documentation/{ => drivers}/auxdisplay/ks0108 | 0 Documentation/{ => drivers}/auxdisplay/lcd-panel-cgram.txt | 0 Documentation/{ => drivers}/backlight/lp855x-driver.txt | 0 .../{ => drivers}/blockdev/drbd/DRBD-8.3-data-packets.svg | 0 Documentation/{ => drivers}/blockdev/drbd/DRBD-data-packets.svg | 0 Documentation/{ => drivers}/blockdev/drbd/README.txt | 0 Documentation/{ => drivers}/blockdev/drbd/conn-states-8.dot | 0 Documentation/{ => drivers}/blockdev/drbd/data-structure-v9.txt | 0 Documentation/{ => drivers}/blockdev/drbd/disk-states-8.dot | 0 .../blockdev/drbd/drbd-connection-state-overview.dot | 0 Documentation/{ => drivers}/blockdev/drbd/node-states-8.dot | 0 Documentation/{ => drivers}/blockdev/floppy.txt | 0 Documentation/{ => drivers}/blockdev/nbd.txt | 0 Documentation/{ => drivers}/blockdev/paride.txt | 0 Documentation/{ => drivers}/blockdev/ramdisk.txt | 0 Documentation/{ => drivers}/blockdev/zram.txt | 0 Documentation/{bus-devices => drivers/bus}/ti-gpmc.txt | 0 Documentation/{ => drivers}/cdrom/cdrom-standard.rst | 0 Documentation/{ => drivers}/cdrom/ide-cd.rst | 0 Documentation/{ => drivers}/cdrom/index.rst | 0 Documentation/{ => drivers}/cdrom/packet-writing.rst | 0 Documentation/{ => drivers}/cpu-freq/amd-powernow.txt | 0 Documentation/{ => drivers}/cpu-freq/core.txt | 0 Documentation/{ => drivers}/cpu-freq/cpu-drivers.txt | 0 Documentation/{ => drivers}/cpu-freq/cpufreq-nforce2.txt | 0 Documentation/{ => drivers}/cpu-freq/cpufreq-stats.txt | 0 Documentation/{ => drivers}/cpu-freq/index.txt | 0 Documentation/{ => drivers}/cpu-freq/pcc-cpufreq.txt | 0 Documentation/{ => drivers}/crypto/api-aead.rst | 0 Documentation/{ => drivers}/crypto/api-akcipher.rst | 0 Documentation/{ => drivers}/crypto/api-digest.rst | 0 Documentation/{ => drivers}/crypto/api-intro.txt | 0 Documentation/{ => drivers}/crypto/api-kpp.rst | 0 Documentation/{ => drivers}/crypto/api-rng.rst | 0 Documentation/{ => drivers}/crypto/api-samples.rst | 0 Documentation/{ => drivers}/crypto/api-skcipher.rst | 0 Documentation/{ => drivers}/crypto/api.rst | 0 Documentation/{ => drivers}/crypto/architecture.rst | 0 Documentation/{ => drivers}/crypto/asymmetric-keys.txt | 0 Documentation/{ => drivers}/crypto/async-tx-api.txt | 0 Documentation/{ => drivers}/crypto/conf.py | 0 Documentation/{ => drivers}/crypto/crypto_engine.rst | 0 Documentation/{ => drivers}/crypto/descore-readme.txt | 0 Documentation/{ => drivers}/crypto/devel-algos.rst | 0 Documentation/{ => drivers}/crypto/index.rst | 0 Documentation/{ => drivers}/crypto/intro.rst | 0 Documentation/{ => drivers}/crypto/userspace-if.rst | 0 Documentation/{ => drivers}/device-mapper/cache-policies.rst | 0 Documentation/{ => drivers}/device-mapper/cache.rst | 0 Documentation/{ => drivers}/device-mapper/delay.rst | 0 Documentation/{ => drivers}/device-mapper/dm-crypt.rst | 0 Documentation/{ => drivers}/device-mapper/dm-dust.txt | 0 Documentation/{ => drivers}/device-mapper/dm-flakey.rst | 0 Documentation/{ => drivers}/device-mapper/dm-init.rst | 0 Documentation/{ => drivers}/device-mapper/dm-integrity.rst | 0 Documentation/{ => drivers}/device-mapper/dm-io.rst | 0 Documentation/{ => drivers}/device-mapper/dm-log.rst | 0 Documentation/{ => drivers}/device-mapper/dm-queue-length.rst | 0 Documentation/{ => drivers}/device-mapper/dm-raid.rst | 0 Documentation/{ => drivers}/device-mapper/dm-service-time.rst | 0 Documentation/{ => drivers}/device-mapper/dm-uevent.rst | 0 Documentation/{ => drivers}/device-mapper/dm-zoned.rst | 0 Documentation/{ => drivers}/device-mapper/era.rst | 0 Documentation/{ => drivers}/device-mapper/index.rst | 0 Documentation/{ => drivers}/device-mapper/kcopyd.rst | 0 Documentation/{ => drivers}/device-mapper/linear.rst | 0 Documentation/{ => drivers}/device-mapper/log-writes.rst | 0 Documentation/{ => drivers}/device-mapper/persistent-data.rst | 0 Documentation/{ => drivers}/device-mapper/snapshot.rst | 0 Documentation/{ => drivers}/device-mapper/statistics.rst | 0 Documentation/{ => drivers}/device-mapper/striped.rst | 0 Documentation/{ => drivers}/device-mapper/switch.rst | 0 Documentation/{ => drivers}/device-mapper/thin-provisioning.rst | 0 Documentation/{ => drivers}/device-mapper/unstriped.rst | 0 Documentation/{ => drivers}/device-mapper/verity.rst | 0 Documentation/{ => drivers}/device-mapper/writecache.rst | 0 Documentation/{ => drivers}/device-mapper/zero.rst | 0 Documentation/{ => drivers}/driver-api/80211/cfg80211.rst | 0 Documentation/{ => drivers}/driver-api/80211/conf.py | 0 Documentation/{ => drivers}/driver-api/80211/index.rst | 0 Documentation/{ => drivers}/driver-api/80211/introduction.rst | 0 .../{ => drivers}/driver-api/80211/mac80211-advanced.rst | 0 Documentation/{ => drivers}/driver-api/80211/mac80211.rst | 0 Documentation/{ => drivers}/driver-api/acpi/index.rst | 0 Documentation/{ => drivers}/driver-api/acpi/linuxized-acpica.rst | 0 Documentation/{ => drivers}/driver-api/acpi/scan_handlers.rst | 0 Documentation/{ => drivers}/driver-api/basics.rst | 0 Documentation/{ => drivers}/driver-api/clk.rst | 0 Documentation/{ => drivers}/driver-api/component.rst | 0 Documentation/{ => drivers}/driver-api/conf.py | 0 Documentation/{ => drivers}/driver-api/device-io.rst | 0 Documentation/{ => drivers}/driver-api/device_connection.rst | 0 Documentation/{ => drivers}/driver-api/device_link.rst | 0 Documentation/{ => drivers}/driver-api/dma-buf.rst | 0 Documentation/{ => drivers}/driver-api/dmaengine/client.rst | 0 Documentation/{ => drivers}/driver-api/dmaengine/dmatest.rst | 0 Documentation/{ => drivers}/driver-api/dmaengine/index.rst | 0 Documentation/{ => drivers}/driver-api/dmaengine/provider.rst | 0 Documentation/{ => drivers}/driver-api/dmaengine/pxa_dma.rst | 0 Documentation/{ => drivers}/driver-api/edac.rst | 0 Documentation/{ => drivers}/driver-api/firewire.rst | 0 Documentation/{ => drivers}/driver-api/firmware/built-in-fw.rst | 0 Documentation/{ => drivers}/driver-api/firmware/core.rst | 0 .../{ => drivers}/driver-api/firmware/direct-fs-lookup.rst | 0 .../{ => drivers}/driver-api/firmware/fallback-mechanisms.rst | 0 .../{ => drivers}/driver-api/firmware/firmware_cache.rst | 0 .../{ => drivers}/driver-api/firmware/fw_search_path.rst | 0 Documentation/{ => drivers}/driver-api/firmware/index.rst | 0 Documentation/{ => drivers}/driver-api/firmware/introduction.rst | 0 Documentation/{ => drivers}/driver-api/firmware/lookup-order.rst | 0 .../{ => drivers}/driver-api/firmware/other_interfaces.rst | 0 .../{ => drivers}/driver-api/firmware/request_firmware.rst | 0 Documentation/{ => drivers}/driver-api/fpga/fpga-bridge.rst | 0 Documentation/{ => drivers}/driver-api/fpga/fpga-mgr.rst | 0 Documentation/{ => drivers}/driver-api/fpga/fpga-programming.rst | 0 Documentation/{ => drivers}/driver-api/fpga/fpga-region.rst | 0 Documentation/{ => drivers}/driver-api/fpga/index.rst | 0 Documentation/{ => drivers}/driver-api/fpga/intro.rst | 0 Documentation/{ => drivers}/driver-api/frame-buffer.rst | 0 Documentation/{ => drivers}/driver-api/generic-counter.rst | 0 Documentation/{ => drivers}/driver-api/gpio/board.rst | 0 Documentation/{ => drivers}/driver-api/gpio/consumer.rst | 0 Documentation/{ => drivers}/driver-api/gpio/driver.rst | 0 Documentation/{ => drivers}/driver-api/gpio/drivers-on-gpio.rst | 0 Documentation/{ => drivers}/driver-api/gpio/index.rst | 0 Documentation/{ => drivers}/driver-api/gpio/intro.rst | 0 Documentation/{ => drivers}/driver-api/gpio/legacy.rst | 0 Documentation/{ => drivers}/driver-api/hsi.rst | 0 Documentation/{ => drivers}/driver-api/i2c.rst | 0 Documentation/{ => drivers}/driver-api/i3c/device-driver-api.rst | 0 Documentation/{ => drivers}/driver-api/i3c/index.rst | 0 Documentation/{ => drivers}/driver-api/i3c/master-driver-api.rst | 0 Documentation/{ => drivers}/driver-api/i3c/protocol.rst | 0 Documentation/{ => drivers}/driver-api/iio/buffers.rst | 0 Documentation/{ => drivers}/driver-api/iio/core.rst | 0 Documentation/{ => drivers}/driver-api/iio/hw-consumer.rst | 0 Documentation/{ => drivers}/driver-api/iio/index.rst | 0 Documentation/{ => drivers}/driver-api/iio/intro.rst | 0 Documentation/{ => drivers}/driver-api/iio/triggered-buffers.rst | 0 Documentation/{ => drivers}/driver-api/iio/triggers.rst | 0 Documentation/{ => drivers}/driver-api/index.rst | 0 Documentation/{ => drivers}/driver-api/infrastructure.rst | 0 Documentation/{ => drivers}/driver-api/input.rst | 0 Documentation/{ => drivers}/driver-api/libata.rst | 0 Documentation/{ => drivers}/driver-api/message-based.rst | 0 Documentation/{ => drivers}/driver-api/misc_devices.rst | 0 Documentation/{ => drivers}/driver-api/miscellaneous.rst | 0 Documentation/{ => drivers}/driver-api/mtdnand.rst | 0 Documentation/{ => drivers}/driver-api/pci/index.rst | 0 Documentation/{ => drivers}/driver-api/pci/p2pdma.rst | 0 Documentation/{ => drivers}/driver-api/pci/pci.rst | 0 Documentation/{ => drivers}/driver-api/pinctl.rst | 0 Documentation/{ => drivers}/driver-api/pm/conf.py | 0 Documentation/{ => drivers}/driver-api/pm/cpuidle.rst | 0 Documentation/{ => drivers}/driver-api/pm/devices.rst | 0 Documentation/{ => drivers}/driver-api/pm/index.rst | 0 Documentation/{ => drivers}/driver-api/pm/notifiers.rst | 0 Documentation/{ => drivers}/driver-api/pm/types.rst | 0 Documentation/{ => drivers}/driver-api/pps.rst | 0 Documentation/{ => drivers}/driver-api/ptp.rst | 0 Documentation/{ => drivers}/driver-api/rapidio.rst | 0 Documentation/{ => drivers}/driver-api/regulator.rst | 0 Documentation/{ => drivers}/driver-api/s390-drivers.rst | 0 Documentation/{ => drivers}/driver-api/scsi.rst | 0 Documentation/{ => drivers}/driver-api/slimbus.rst | 0 Documentation/{ => drivers}/driver-api/sound.rst | 0 .../{ => drivers}/driver-api/soundwire/error_handling.rst | 0 Documentation/{ => drivers}/driver-api/soundwire/index.rst | 0 Documentation/{ => drivers}/driver-api/soundwire/locking.rst | 0 Documentation/{ => drivers}/driver-api/soundwire/stream.rst | 0 Documentation/{ => drivers}/driver-api/soundwire/summary.rst | 0 Documentation/{ => drivers}/driver-api/spi.rst | 0 Documentation/{ => drivers}/driver-api/target.rst | 0 Documentation/{ => drivers}/driver-api/uio-howto.rst | 0 Documentation/{ => drivers}/driver-api/usb/URB.rst | 0 Documentation/{ => drivers}/driver-api/usb/anchors.rst | 0 Documentation/{ => drivers}/driver-api/usb/bulk-streams.rst | 0 Documentation/{ => drivers}/driver-api/usb/callbacks.rst | 0 Documentation/{ => drivers}/driver-api/usb/dma.rst | 0 Documentation/{ => drivers}/driver-api/usb/dwc3.rst | 0 Documentation/{ => drivers}/driver-api/usb/error-codes.rst | 0 Documentation/{ => drivers}/driver-api/usb/gadget.rst | 0 Documentation/{ => drivers}/driver-api/usb/hotplug.rst | 0 Documentation/{ => drivers}/driver-api/usb/index.rst | 0 Documentation/{ => drivers}/driver-api/usb/persist.rst | 0 Documentation/{ => drivers}/driver-api/usb/power-management.rst | 0 Documentation/{ => drivers}/driver-api/usb/typec.rst | 0 Documentation/{ => drivers}/driver-api/usb/typec_bus.rst | 0 Documentation/{ => drivers}/driver-api/usb/usb.rst | 0 Documentation/{ => drivers}/driver-api/usb/usb3-debug-port.rst | 0 .../{ => drivers}/driver-api/usb/writing_musb_glue_layer.rst | 0 Documentation/{ => drivers}/driver-api/usb/writing_usb_driver.rst | 0 Documentation/{ => drivers}/driver-api/vme.rst | 0 Documentation/{ => drivers}/driver-api/w1.rst | 0 Documentation/{ => drivers}/driver-model/binding.txt | 0 Documentation/{ => drivers}/driver-model/bus.txt | 0 Documentation/{ => drivers}/driver-model/class.txt | 0 Documentation/{ => drivers}/driver-model/design-patterns.txt | 0 Documentation/{ => drivers}/driver-model/device.txt | 0 Documentation/{ => drivers}/driver-model/devres.txt | 0 Documentation/{ => drivers}/driver-model/driver.txt | 0 Documentation/{ => drivers}/driver-model/overview.txt | 0 Documentation/{ => drivers}/driver-model/platform.txt | 0 Documentation/{ => drivers}/driver-model/porting.txt | 0 Documentation/{ => drivers}/fpga/dfl.rst | 0 Documentation/{ => drivers}/fpga/index.rst | 0 Documentation/{ => drivers}/gpio/index.rst | 0 Documentation/{ => drivers}/gpio/sysfs.rst | 0 Documentation/{ => drivers}/gpu/afbc.rst | 0 Documentation/{ => drivers}/gpu/amdgpu-dc.rst | 0 Documentation/{ => drivers}/gpu/amdgpu.rst | 0 Documentation/{ => drivers}/gpu/bridge/dw-hdmi.rst | 0 Documentation/{ => drivers}/gpu/conf.py | 0 Documentation/{ => drivers}/gpu/dp-mst/topology-figure-1.dot | 0 Documentation/{ => drivers}/gpu/dp-mst/topology-figure-2.dot | 0 Documentation/{ => 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Documentation/{ => drivers}/scsi/lpfc.txt | 0 Documentation/{ => drivers}/scsi/megaraid.txt | 0 Documentation/{ => drivers}/scsi/ncr53c8xx.txt | 0 Documentation/{ => drivers}/scsi/osst.txt | 0 Documentation/{ => drivers}/scsi/ppa.txt | 0 Documentation/{ => drivers}/scsi/qlogicfas.txt | 0 Documentation/{ => drivers}/scsi/scsi-changer.txt | 0 Documentation/{ => drivers}/scsi/scsi-generic.txt | 0 Documentation/{ => drivers}/scsi/scsi-parameters.txt | 0 Documentation/{ => drivers}/scsi/scsi.txt | 0 Documentation/{ => drivers}/scsi/scsi_eh.txt | 0 Documentation/{ => drivers}/scsi/scsi_fc_transport.txt | 0 Documentation/{ => drivers}/scsi/scsi_mid_low_api.txt | 0 Documentation/{ => drivers}/scsi/scsi_transport_srp/Makefile | 0 .../{ => drivers}/scsi/scsi_transport_srp/rport_state_diagram.dot | 0 Documentation/{ => drivers}/scsi/sd-parameters.txt | 0 Documentation/{ => drivers}/scsi/smartpqi.txt | 0 Documentation/{ => drivers}/scsi/st.txt | 0 Documentation/{ => drivers}/scsi/sym53c500_cs.txt | 0 Documentation/{ => drivers}/scsi/sym53c8xx_2.txt | 0 Documentation/{ => drivers}/scsi/tcm_qla2xxx.txt | 0 Documentation/{ => drivers}/scsi/ufs.txt | 0 Documentation/{ => drivers}/scsi/wd719x.txt | 0 Documentation/{ => drivers}/serial/cyclades_z.rst | 0 Documentation/{ => drivers}/serial/driver.rst | 0 Documentation/{ => drivers}/serial/index.rst | 0 Documentation/{ => drivers}/serial/moxa-smartio.rst | 0 Documentation/{ => drivers}/serial/n_gsm.rst | 0 Documentation/{ => drivers}/serial/rocket.rst | 0 Documentation/{ => drivers}/serial/serial-iso7816.rst | 0 Documentation/{ => drivers}/serial/serial-rs485.rst | 0 Documentation/{ => drivers}/serial/tty.rst | 0 Documentation/{ => drivers}/sound/alsa-configuration.rst | 0 Documentation/{ => drivers}/sound/cards/audigy-mixer.rst | 0 Documentation/{ => drivers}/sound/cards/audiophile-usb.rst | 0 Documentation/{ => drivers}/sound/cards/bt87x.rst | 0 Documentation/{ => drivers}/sound/cards/cmipci.rst | 0 Documentation/{ => drivers}/sound/cards/emu10k1-jack.rst | 0 Documentation/{ => drivers}/sound/cards/hdspm.rst | 0 Documentation/{ => drivers}/sound/cards/img-spdif-in.rst | 0 Documentation/{ => drivers}/sound/cards/index.rst | 0 Documentation/{ => drivers}/sound/cards/joystick.rst | 0 Documentation/{ => drivers}/sound/cards/maya44.rst | 0 Documentation/{ => drivers}/sound/cards/mixart.rst | 0 Documentation/{ => drivers}/sound/cards/multisound.sh | 0 Documentation/{ => drivers}/sound/cards/sb-live-mixer.rst | 0 Documentation/{ => drivers}/sound/cards/serial-u16550.rst | 0 Documentation/{ => drivers}/sound/cards/via82xx-mixer.rst | 0 Documentation/{ => drivers}/sound/conf.py | 0 Documentation/{ => drivers}/sound/designs/channel-mapping-api.rst | 0 Documentation/{ => drivers}/sound/designs/compress-offload.rst | 0 Documentation/{ => drivers}/sound/designs/control-names.rst | 0 Documentation/{ => drivers}/sound/designs/index.rst | 0 Documentation/{ => drivers}/sound/designs/jack-controls.rst | 0 Documentation/{ => drivers}/sound/designs/oss-emulation.rst | 0 Documentation/{ => drivers}/sound/designs/powersave.rst | 0 Documentation/{ => drivers}/sound/designs/procfile.rst | 0 Documentation/{ => drivers}/sound/designs/seq-oss.rst | 0 Documentation/{ => drivers}/sound/designs/timestamping.rst | 0 Documentation/{ => drivers}/sound/designs/tracepoints.rst | 0 Documentation/{ => drivers}/sound/hd-audio/controls.rst | 0 Documentation/{ => drivers}/sound/hd-audio/dp-mst.rst | 0 Documentation/{ => drivers}/sound/hd-audio/index.rst | 0 Documentation/{ => drivers}/sound/hd-audio/models.rst | 0 Documentation/{ => drivers}/sound/hd-audio/notes.rst | 0 Documentation/{ => drivers}/sound/index.rst | 0 Documentation/{ => drivers}/sound/kernel-api/alsa-driver-api.rst | 0 Documentation/{ => drivers}/sound/kernel-api/index.rst | 0 .../{ => drivers}/sound/kernel-api/writing-an-alsa-driver.rst | 0 Documentation/{ => drivers}/sound/soc/clocking.rst | 0 Documentation/{ => drivers}/sound/soc/codec-to-codec.rst | 0 Documentation/{ => drivers}/sound/soc/codec.rst | 0 Documentation/{ => drivers}/sound/soc/dai.rst | 0 Documentation/{ => drivers}/sound/soc/dapm.rst | 0 Documentation/{ => drivers}/sound/soc/dpcm.rst | 0 Documentation/{ => drivers}/sound/soc/index.rst | 0 Documentation/{ => drivers}/sound/soc/jack.rst | 0 Documentation/{ => drivers}/sound/soc/machine.rst | 0 Documentation/{ => drivers}/sound/soc/overview.rst | 0 Documentation/{ => drivers}/sound/soc/platform.rst | 0 Documentation/{ => drivers}/sound/soc/pops-clicks.rst | 0 Documentation/{ => drivers}/usb/CREDITS | 0 Documentation/{ => drivers}/usb/WUSB-Design-overview.txt | 0 Documentation/{ => drivers}/usb/acm.txt | 0 Documentation/{ => drivers}/usb/authorization.txt | 0 Documentation/{ => drivers}/usb/chipidea.txt | 0 Documentation/{ => drivers}/usb/dwc3.txt | 0 Documentation/{ => drivers}/usb/ehci.txt | 0 Documentation/{ => drivers}/usb/functionfs.txt | 0 Documentation/{ => drivers}/usb/gadget-testing.txt | 0 Documentation/{ => drivers}/usb/gadget_configfs.txt | 0 Documentation/{ => drivers}/usb/gadget_hid.txt | 0 Documentation/{ => drivers}/usb/gadget_multi.txt | 0 Documentation/{ => drivers}/usb/gadget_printer.txt | 0 Documentation/{ => drivers}/usb/gadget_serial.txt | 0 Documentation/{ => drivers}/usb/iuu_phoenix.txt | 0 Documentation/{ => drivers}/usb/linux-cdc-acm.inf | 0 Documentation/{ => drivers}/usb/linux.inf | 0 Documentation/{ => drivers}/usb/mass-storage.txt | 0 Documentation/{ => drivers}/usb/misc_usbsevseg.txt | 0 Documentation/{ => drivers}/usb/mtouchusb.txt | 0 Documentation/{ => drivers}/usb/ohci.txt | 0 Documentation/{ => drivers}/usb/rio.txt | 0 Documentation/{ => drivers}/usb/usb-help.txt | 0 Documentation/{ => drivers}/usb/usb-serial.txt | 0 Documentation/{ => drivers}/usb/usbdevfs-drop-permissions.c | 0 Documentation/{ => drivers}/usb/usbip_protocol.txt | 0 Documentation/{ => drivers}/usb/usbmon.txt | 0 Documentation/{ => drivers}/usb/wusb-cbaf | 0 .../{ => drivers}/watchdog/convert_drivers_to_kernel_api.rst | 0 Documentation/{ => drivers}/watchdog/hpwdt.rst | 0 Documentation/{ => drivers}/watchdog/index.rst | 0 Documentation/{ => drivers}/watchdog/mlx-wdt.rst | 0 Documentation/{ => drivers}/watchdog/pcwd-watchdog.rst | 0 Documentation/{ => drivers}/watchdog/watchdog-api.rst | 0 Documentation/{ => drivers}/watchdog/watchdog-kernel-api.rst | 0 Documentation/{ => drivers}/watchdog/watchdog-parameters.rst | 0 Documentation/{ => drivers}/watchdog/watchdog-pm.rst | 0 Documentation/{ => drivers}/watchdog/wdt.rst | 0 1079 files changed, 0 insertions(+), 0 deletions(-) rename Documentation/{ => drivers}/PCI/MSI-HOWTO.txt (100%) rename Documentation/{ => drivers}/PCI/PCIEBUS-HOWTO.txt (100%) rename Documentation/{ => drivers}/PCI/acpi-info.txt (100%) rename Documentation/{ => drivers}/PCI/endpoint/function/binding/pci-test.txt (100%) rename Documentation/{ => drivers}/PCI/endpoint/pci-endpoint-cfs.txt (100%) rename Documentation/{ => drivers}/PCI/endpoint/pci-endpoint.txt (100%) rename Documentation/{ => drivers}/PCI/endpoint/pci-test-function.txt (100%) rename Documentation/{ => drivers}/PCI/endpoint/pci-test-howto.txt (100%) rename Documentation/{ => drivers}/PCI/pci-error-recovery.txt (100%) rename Documentation/{ => drivers}/PCI/pci-iov-howto.txt (100%) rename Documentation/{ => drivers}/PCI/pci.txt (100%) rename Documentation/{ => drivers}/PCI/pcieaer-howto.txt (100%) rename Documentation/{ => drivers}/acpi/dsd/leds.txt (100%) rename Documentation/{ => drivers}/auxdisplay/cfag12864b (100%) rename Documentation/{ => drivers}/auxdisplay/ks0108 (100%) rename Documentation/{ => drivers}/auxdisplay/lcd-panel-cgram.txt (100%) rename Documentation/{ => drivers}/backlight/lp855x-driver.txt (100%) rename Documentation/{ => drivers}/blockdev/drbd/DRBD-8.3-data-packets.svg (100%) rename Documentation/{ => drivers}/blockdev/drbd/DRBD-data-packets.svg (100%) rename Documentation/{ => drivers}/blockdev/drbd/README.txt (100%) rename Documentation/{ => drivers}/blockdev/drbd/conn-states-8.dot (100%) rename Documentation/{ => drivers}/blockdev/drbd/data-structure-v9.txt (100%) rename Documentation/{ => drivers}/blockdev/drbd/disk-states-8.dot (100%) rename Documentation/{ => drivers}/blockdev/drbd/drbd-connection-state-overview.dot (100%) rename Documentation/{ => drivers}/blockdev/drbd/node-states-8.dot (100%) rename Documentation/{ => drivers}/blockdev/floppy.txt (100%) rename Documentation/{ => drivers}/blockdev/nbd.txt (100%) rename Documentation/{ => drivers}/blockdev/paride.txt (100%) rename Documentation/{ => drivers}/blockdev/ramdisk.txt (100%) rename Documentation/{ => drivers}/blockdev/zram.txt (100%) rename Documentation/{bus-devices => drivers/bus}/ti-gpmc.txt (100%) rename Documentation/{ => drivers}/cdrom/cdrom-standard.rst (100%) rename Documentation/{ => drivers}/cdrom/ide-cd.rst (100%) rename Documentation/{ => drivers}/cdrom/index.rst (100%) rename 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--git a/Documentation/media/uapi/mediactl/request-api.rst b/Documentation/drivers/media/uapi/mediactl/request-api.rst similarity index 100% rename from Documentation/media/uapi/mediactl/request-api.rst rename to Documentation/drivers/media/uapi/mediactl/request-api.rst diff --git a/Documentation/media/uapi/mediactl/request-func-close.rst b/Documentation/drivers/media/uapi/mediactl/request-func-close.rst similarity index 100% rename from Documentation/media/uapi/mediactl/request-func-close.rst rename to Documentation/drivers/media/uapi/mediactl/request-func-close.rst diff --git a/Documentation/media/uapi/mediactl/request-func-ioctl.rst b/Documentation/drivers/media/uapi/mediactl/request-func-ioctl.rst similarity index 100% rename from Documentation/media/uapi/mediactl/request-func-ioctl.rst rename to Documentation/drivers/media/uapi/mediactl/request-func-ioctl.rst diff --git a/Documentation/media/uapi/mediactl/request-func-poll.rst b/Documentation/drivers/media/uapi/mediactl/request-func-poll.rst similarity index 100% rename from Documentation/media/uapi/mediactl/request-func-poll.rst rename to Documentation/drivers/media/uapi/mediactl/request-func-poll.rst diff --git a/Documentation/media/uapi/rc/keytable.c.rst b/Documentation/drivers/media/uapi/rc/keytable.c.rst similarity index 100% rename from Documentation/media/uapi/rc/keytable.c.rst rename to Documentation/drivers/media/uapi/rc/keytable.c.rst diff --git a/Documentation/media/uapi/rc/lirc-dev-intro.rst b/Documentation/drivers/media/uapi/rc/lirc-dev-intro.rst similarity index 100% rename from Documentation/media/uapi/rc/lirc-dev-intro.rst rename to Documentation/drivers/media/uapi/rc/lirc-dev-intro.rst diff --git a/Documentation/media/uapi/rc/lirc-dev.rst b/Documentation/drivers/media/uapi/rc/lirc-dev.rst similarity index 100% rename from Documentation/media/uapi/rc/lirc-dev.rst rename to Documentation/drivers/media/uapi/rc/lirc-dev.rst diff --git a/Documentation/media/uapi/rc/lirc-func.rst b/Documentation/drivers/media/uapi/rc/lirc-func.rst similarity index 100% rename from Documentation/media/uapi/rc/lirc-func.rst rename to Documentation/drivers/media/uapi/rc/lirc-func.rst diff --git a/Documentation/media/uapi/rc/lirc-get-features.rst b/Documentation/drivers/media/uapi/rc/lirc-get-features.rst similarity index 100% rename from Documentation/media/uapi/rc/lirc-get-features.rst rename to Documentation/drivers/media/uapi/rc/lirc-get-features.rst diff --git a/Documentation/media/uapi/rc/lirc-get-rec-mode.rst b/Documentation/drivers/media/uapi/rc/lirc-get-rec-mode.rst similarity index 100% rename from Documentation/media/uapi/rc/lirc-get-rec-mode.rst rename to Documentation/drivers/media/uapi/rc/lirc-get-rec-mode.rst diff --git a/Documentation/media/uapi/rc/lirc-get-rec-resolution.rst b/Documentation/drivers/media/uapi/rc/lirc-get-rec-resolution.rst similarity index 100% rename from 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a/Documentation/media/uapi/rc/lirc-write.rst b/Documentation/drivers/media/uapi/rc/lirc-write.rst similarity index 100% rename from Documentation/media/uapi/rc/lirc-write.rst rename to Documentation/drivers/media/uapi/rc/lirc-write.rst diff --git a/Documentation/media/uapi/rc/rc-intro.rst b/Documentation/drivers/media/uapi/rc/rc-intro.rst similarity index 100% rename from Documentation/media/uapi/rc/rc-intro.rst rename to Documentation/drivers/media/uapi/rc/rc-intro.rst diff --git a/Documentation/media/uapi/rc/rc-sysfs-nodes.rst b/Documentation/drivers/media/uapi/rc/rc-sysfs-nodes.rst similarity index 100% rename from Documentation/media/uapi/rc/rc-sysfs-nodes.rst rename to Documentation/drivers/media/uapi/rc/rc-sysfs-nodes.rst diff --git a/Documentation/media/uapi/rc/rc-table-change.rst b/Documentation/drivers/media/uapi/rc/rc-table-change.rst similarity index 100% rename from Documentation/media/uapi/rc/rc-table-change.rst rename to 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Documentation/drivers/usb/usbdevfs-drop-permissions.c diff --git a/Documentation/usb/usbip_protocol.txt b/Documentation/drivers/usb/usbip_protocol.txt similarity index 100% rename from Documentation/usb/usbip_protocol.txt rename to Documentation/drivers/usb/usbip_protocol.txt diff --git a/Documentation/usb/usbmon.txt b/Documentation/drivers/usb/usbmon.txt similarity index 100% rename from Documentation/usb/usbmon.txt rename to Documentation/drivers/usb/usbmon.txt diff --git a/Documentation/usb/wusb-cbaf b/Documentation/drivers/usb/wusb-cbaf similarity index 100% rename from Documentation/usb/wusb-cbaf rename to Documentation/drivers/usb/wusb-cbaf diff --git a/Documentation/watchdog/convert_drivers_to_kernel_api.rst b/Documentation/drivers/watchdog/convert_drivers_to_kernel_api.rst similarity index 100% rename from Documentation/watchdog/convert_drivers_to_kernel_api.rst rename to Documentation/drivers/watchdog/convert_drivers_to_kernel_api.rst diff --git a/Documentation/watchdog/hpwdt.rst b/Documentation/drivers/watchdog/hpwdt.rst similarity index 100% rename from Documentation/watchdog/hpwdt.rst rename to Documentation/drivers/watchdog/hpwdt.rst diff --git a/Documentation/watchdog/index.rst b/Documentation/drivers/watchdog/index.rst similarity index 100% rename from Documentation/watchdog/index.rst rename to Documentation/drivers/watchdog/index.rst diff --git a/Documentation/watchdog/mlx-wdt.rst b/Documentation/drivers/watchdog/mlx-wdt.rst similarity index 100% rename from Documentation/watchdog/mlx-wdt.rst rename to Documentation/drivers/watchdog/mlx-wdt.rst diff --git a/Documentation/watchdog/pcwd-watchdog.rst b/Documentation/drivers/watchdog/pcwd-watchdog.rst similarity index 100% rename from Documentation/watchdog/pcwd-watchdog.rst rename to Documentation/drivers/watchdog/pcwd-watchdog.rst diff --git a/Documentation/watchdog/watchdog-api.rst b/Documentation/drivers/watchdog/watchdog-api.rst similarity index 100% rename from Documentation/watchdog/watchdog-api.rst rename to Documentation/drivers/watchdog/watchdog-api.rst diff --git a/Documentation/watchdog/watchdog-kernel-api.rst b/Documentation/drivers/watchdog/watchdog-kernel-api.rst similarity index 100% rename from Documentation/watchdog/watchdog-kernel-api.rst rename to Documentation/drivers/watchdog/watchdog-kernel-api.rst diff --git a/Documentation/watchdog/watchdog-parameters.rst b/Documentation/drivers/watchdog/watchdog-parameters.rst similarity index 100% rename from Documentation/watchdog/watchdog-parameters.rst rename to Documentation/drivers/watchdog/watchdog-parameters.rst diff --git a/Documentation/watchdog/watchdog-pm.rst b/Documentation/drivers/watchdog/watchdog-pm.rst similarity index 100% rename from Documentation/watchdog/watchdog-pm.rst rename to Documentation/drivers/watchdog/watchdog-pm.rst diff --git a/Documentation/watchdog/wdt.rst b/Documentation/drivers/watchdog/wdt.rst similarity index 100% rename from Documentation/watchdog/wdt.rst rename to Documentation/drivers/watchdog/wdt.rst -- 2.19.1.856.g8858448bb

1 1

[alsa-devel] [PATCH 0/2] ALSA: AMD and Nvidia HD-audio component notifier support
by Takashi Iwai 18 Jul '19

18 Jul '19

Hi, this is a revival of my old patchset for adding the support of drm audio component for AMD and Nvidia HDMI codecs. This is only HD-audio side, and we need the support in DRM side as well, but it's a step forward to the better HDMI hotplug handling. I'm going to submit the radeon and nouveau patches later once after merging this for 5.4 kernel. The first patch is somewhat new, a sort of optimization. The second patch has been slightly rewritten frm the previous version and the support for Nvidia was added as well. Lightly tested on a Dell desktop with Nvidia graphics. Takashi === Takashi Iwai (2): ALSA: hda/hdmi - Don't report spurious jack state changes ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI sound/pci/hda/patch_hdmi.c | 265 +++++++++++++++++++++++++++++++++++++++------ 1 file changed, 229 insertions(+), 36 deletions(-) -- 2.16.4

2 4

[alsa-devel] [PATCH] ALSA: hda - Optimize resume for codecs without jack detection
by Takashi Iwai 18 Jul '19

18 Jul '19

The codecs without jack detection also don't have to be resumed forcibly because, obviously, they have no jack. Skip the forced resume in such a case as optimization as well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e30e86ca6b72..51f10ed9bc43 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2942,7 +2942,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume; + bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; int ret; /* The get/put pair below enforces the runtime resume even if the -- 2.16.4

2 1

[alsa-devel] [PATCH] ALSA: line6: Fix wrong altsetting for LINE6_PODHD500_1
by Kai-Heng Feng 18 Jul '19

18 Jul '19

Commit 7b9584fa1c0b ("staging: line6: Move altsetting to properties") set a wrong altsetting for LINE6_PODHD500_1 during refactoring. Set the correct altsetting number to fix the issue. BugLink: https://bugs.launchpad.net/bugs/1790595 Fixes: 7b9584fa1c0b ("staging: line6: Move altsetting to properties") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/line6/podhd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index f0662bd4e50f..27bf61c177c0 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -368,7 +368,7 @@ static const struct line6_properties podhd_properties_table[] = { .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_HWMON, - .altsetting = 1, + .altsetting = 0, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: samsung: odroid: Use common code in odroid_audio_probe()
by Markus Elfring 18 Jul '19

18 Jul '19

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 18 Jul 2019 11:42:29 +0200 Replace a function call and a return statement by a goto statement so that a bit of common code will be reused at the end of this function. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/samsung/odroid.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index f0f5fa9c27d3..d152ef8dfea3 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -316,8 +316,7 @@ static int odroid_audio_probe(struct platform_device *pdev) } of_node_put(cpu_dai); - of_node_put(codec); - return 0; + goto err_put_node; err_put_clk_i2s: clk_put(priv->clk_i2s_bus); -- 2.22.0

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: SOF: add flag for position update ipc
by Keyon Jie 18 Jul '19

18 Jul '19

From: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> In some cases, FW might need use the host_period_bytes even no position update ipc reqiured from driver, here add another flag for position update, and preserve host_period_bytes for FW to use. This might require corresponding FW change and ABI alignment. Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- include/sound/sof/stream.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h index 643f175cb479..44acfa62fa69 100644 --- a/include/sound/sof/stream.h +++ b/include/sound/sof/stream.h @@ -83,10 +83,10 @@ struct sof_ipc_stream_params { uint16_t sample_valid_bytes; uint16_t sample_container_bytes; - /* for notifying host period has completed - 0 means no period IRQ */ uint32_t host_period_bytes; + uint16_t no_period_irq; /* 1 means period IRQ mode OFF */ - uint32_t reserved[2]; + uint16_t reserved[3]; uint16_t chmap[SOF_IPC_MAX_CHANNELS]; /**< channel map - SOF_CHMAP_ */ } __packed; -- 2.20.1

4 8

[alsa-devel] [PATCH v4 0/6] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 18 Jul '19

18 Jul '19

VAG power control is improved to fit the manual [1]. This patchset fixes as minimum one bug: if customer muxes Headphone to Line-In right after boot, the VAG power remains off that leads to poor sound quality from line-in. I.e. after boot: - Connect sound source to Line-In jack; - Connect headphone to HP jack; - Run following commands: $ amixer set 'Headphone' 80% $ amixer set 'Headphone Mux' LINE_IN Also this series includes fixes of non-important bugs in sgtl5000 codec driver. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Changes in v4: - CC the patch to kernel-stable - Code optimization, simplify function signature (thanks to Cezary Rojewski <cezary.rojewski(a)intel.com> for an idea) - CC the patch to kernel-stable - Add a Fixes tag Changes in v3: - Add the reference to NXP SGTL5000 data sheet to commit message - Add the reference to NXP SGTL5000 data sheet to commit message - Fix multi-line comment format Changes in v2: - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting Oleksandr Suvorov (6): ASoC: Define a set of DAPM pre/post-up events ASoC: sgtl5000: Improve VAG power and mute control ASoC: sgtl5000: Fix definition of VAG Ramp Control ASoC: sgtl5000: add ADC mute control ASoC: sgtl5000: Fix of unmute outputs on probe ASoC: sgtl5000: Fix charge pump source assignment include/sound/soc-dapm.h | 2 + sound/soc/codecs/sgtl5000.c | 240 ++++++++++++++++++++++++++++++------ sound/soc/codecs/sgtl5000.h | 2 +- 3 files changed, 203 insertions(+), 41 deletions(-) -- 2.20.1

3 13

[alsa-devel] Is there a way to support TX master mode and RX slave mode?
by S.j. Wang 17 Jul '19

17 Jul '19

Hi Experts Is there a way to support TX master mode but RX slave mode? It seems current .set_fmt only set same mode for TX and RX. Best regards Wang shengjiu

3 2

[alsa-devel] [PATCH v3 0/6] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 17 Jul '19

17 Jul '19

VAG power control is improved to fit the manual [1]. This patchset fixes as minimum one bug: if customer muxes Headphone to Line-In right after boot, the VAG power remains off that leads to poor sound quality from line-in. I.e. after boot: - Connect sound source to Line-In jack; - Connect headphone to HP jack; - Run following commands: $ amixer set 'Headphone' 80% $ amixer set 'Headphone Mux' LINE_IN Also this series includes fixes of non-important bugs in sgtl5000 codec driver. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Changes in v3: - Add the reference to NXP SGTL5000 data sheet to commit message - Fix multi-line comment format - Add the reference to NXP SGTL5000 data sheet to commit message Changes in v2: - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting - Fix patch formatting Oleksandr Suvorov (6): ASoC: sgtl5000: Fix definition of VAG Ramp Control ASoC: sgtl5000: add ADC mute control ASoC: sgtl5000: Fix of unmute outputs on probe ASoC: sgtl5000: Fix charge pump source assignment ASoC: Define a set of DAPM pre/post-up events ASoC: sgtl5000: Improve VAG power and mute control include/sound/soc-dapm.h | 2 + sound/soc/codecs/sgtl5000.c | 251 ++++++++++++++++++++++++++++++------ sound/soc/codecs/sgtl5000.h | 2 +- 3 files changed, 213 insertions(+), 42 deletions(-) -- 2.20.1

3 10

[alsa-devel] [PATCH] ASoC: Intel: Atom: read timestamp moved to period_elapsed
by Alex Levin 17 Jul '19

17 Jul '19

sst_platform_pcm_pointer is called from both snd_pcm_period_elapsed and from snd_pcm_ioctl. Calling read timestamp results in recalculating pcm_delay and buffer_ptr (sst_calc_tstamp) which consumes buffers in a faster rate than intended. In a tested BSW system with chtrt5650, for a rate of 48000, the measured rate was sometimes 10 times more than that. After moving the timestamp read to period elapsed, buffer consumption is as expected. Signed-off-by: Alex Levin <levinale(a)chromium.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 23 +++++++++++++------- 1 file changed, 15 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 0e8b1c5eec88..196af0b30b41 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -265,16 +265,28 @@ static void sst_period_elapsed(void *arg) { struct snd_pcm_substream *substream = arg; struct sst_runtime_stream *stream; - int status; + struct snd_soc_pcm_runtime *rtd; + int status, ret_val; if (!substream || !substream->runtime) return; stream = substream->runtime->private_data; if (!stream) return; + + rtd = substream->private_data; + if (!rtd) + return; + status = sst_get_stream_status(stream); if (status != SST_PLATFORM_RUNNING) return; + + ret_val = stream->ops->stream_read_tstamp(sst->dev, &stream->stream_info); + if (ret_val) { + dev_err(rtd->dev, "stream_read_tstamp err code = %d\n", ret_val); + return; + } snd_pcm_period_elapsed(substream); } @@ -658,20 +670,15 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream; - int ret_val, status; + int status; struct pcm_stream_info *str_info; - struct snd_soc_pcm_runtime *rtd = substream->private_data; stream = substream->runtime->private_data; status = sst_get_stream_status(stream); if (status == SST_PLATFORM_INIT) return 0; + str_info = &stream->stream_info; - ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); - if (ret_val) { - dev_err(rtd->dev, "sst: error code = %d\n", ret_val); - return ret_val; - } substream->runtime->delay = str_info->pcm_delay; return str_info->buffer_ptr; } -- 2.22.0.410.gd8fdbe21b5-goog

4 3

[alsa-devel] [PATCH v4 0/5] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 17 Jul '19

17 Jul '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v3 to v4: - hdmi-codec.h: Modify the hook_plugged_cb ops to take an additional argument, that is, the pointer to struct device for codec device. - dw-hdmi-i2s-audio.c: Simplify the registration of callback so it uses dw_hdmi_set_plugged_cb exported by dw-hdmi.c. - dw-hdmi.c: Simplify the flow to invoke callback since now dw_hdmi has a pointer to codec device as callback argument. There is no need to rely on driver data of other driver. - dw-hdmi.c: Minor change for readability. - synopsys/Kconfig: Fix the dependency of hdmi-codec in Kconfig. - Fixed the incorrect FROMLIST title of patch 5/5. Cheng-Yi Chiang (5): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support drivers/gpu/drm/bridge/synopsys/Kconfig | 2 +- .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + include/sound/hdmi-codec.h | 17 +++ sound/soc/codecs/hdmi-codec.c | 46 +++++++ sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 9 files changed, 217 insertions(+), 28 deletions(-) -- 2.22.0.510.g264f2c817a-goog

2 11

[alsa-devel] [PATCH] ALSA: hda/realtek: apply ALC891 headset fixup to one Dell machine
by Hui Wang 17 Jul '19

17 Jul '19

Without this patch, the headset-mic and headphone-mic don't work. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c84c12b39b3..de224cbea7a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8803,6 +8803,11 @@ static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { {0x18, 0x01a19030}, {0x1a, 0x01813040}, {0x21, 0x01014020}), + SND_HDA_PIN_QUIRK(0x10ec0867, 0x1028, "Dell", ALC891_FIXUP_DELL_MIC_NO_PRESENCE, + {0x16, 0x01813030}, + {0x17, 0x02211010}, + {0x18, 0x01a19040}, + {0x21, 0x01014020}), SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE, {0x14, 0x01014010}, {0x18, 0x01a19020}, -- 2.17.1

3 2

[alsa-devel] [PATCH 0/2] ASoC: samsung: odroid: fix err handling of odroid_audio_probe
by Wen Yang 16 Jul '19

16 Jul '19

We developed a coccinelle SmPL to detect sound/soc/samsung/odroid.c and found some use-after-free problems. This patch series fixes those problems. Wen Yang (2): ASoC: samsung: odroid: fix an use-after-free issue for codec ASoC: samsung: odroid: fix a double-free issue for cpu_dai sound/soc/samsung/odroid.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org -- 2.9.5

5 12

[alsa-devel] [PATCH] ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook
by Hans de Goede 16 Jul '19

16 Jul '19

The Irbis NB41 netbook has its internal mic on IN2, inverted jack-detect and stereo speakers, add a quirk for this. Cc: russianneuromancer(a)ya.ru Reported-and-tested-by: russianneuromancer(a)ya.ru Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcht_es8316.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 2fe1ce879123..c360ebc3ccc7 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -436,6 +436,14 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { /* Please keep this list alphabetically sorted */ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { + { /* Irbis NB41 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"), + DMI_MATCH(DMI_PRODUCT_NAME, "NB41"), + }, + .driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP + | BYT_CHT_ES8316_JD_INVERTED), + }, { /* Teclast X98 Plus II */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"), -- 2.21.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/hdmi - Remove duplicated define
by Takashi Iwai 16 Jul '19

16 Jul '19

INTEL_GET_VENDOR_VERB is defined twice identically. Let's remove a superfluous line. Fixes: b0d8bc50b9f2 ("ALSA: hda: hdmi - add Icelake support") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 40323d91f9e4..1e6c489bca15 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2416,7 +2416,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -#define INTEL_GET_VENDOR_VERB 0xf81 #define INTEL_GET_VENDOR_VERB 0xf81 #define INTEL_SET_VENDOR_VERB 0x781 #define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ -- 2.16.4

2 4

[alsa-devel] [PATCH v2] ALSA: hda/hdmi - Fix i915 reverse port/pin mapping
by Takashi Iwai 16 Jul '19

16 Jul '19

The recent fix for Icelake HDMI codec introduced the mapping from pin NID to the i915 gfx port number. However, it forgot the reverse mapping from the port number to the pin NID that is used in the ELD notifier callback. As a result, it's processed to a wrong widget and gives a warning like snd_hda_codec_hdmi hdaudioC0D2: HDMI: pin nid 5 not registered This patch corrects it with a proper reverse mapping function. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=204133 Fixes: b0d8bc50b9f2 ("ALSA: hda: hdmi - add Icelake support") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Add missing error check from intel_port2pin() sound/pci/hda/patch_hdmi.c | 24 +++++++++++++++++++----- 1 file changed, 19 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1e6c489bca15..0b2a26e2c5f1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2524,18 +2524,32 @@ static int intel_pin2port(void *audio_ptr, int pin_nid) return -1; } +static int intel_port2pin(struct hda_codec *codec, int port) +{ + struct hdmi_spec *spec = codec->spec; + + if (!spec->port_num) { + /* we assume only from port-B to port-D */ + if (port < 1 || port > 3) + return 0; + /* intel port is 1-based */ + return port + intel_base_nid(codec) - 1; + } + + if (port < 1 || port > spec->port_num) + return 0; + return spec->port_map[port - 1]; +} + static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; int pin_nid; int dev_id = pipe; - /* we assume only from port-B to port-D */ - if (port < 1 || port > 3) + pin_nid = intel_port2pin(codec, port); + if (!pin_nid) return; - - pin_nid = port + intel_base_nid(codec) - 1; /* intel port is 1-based */ - /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ -- 2.16.4

1 0

[alsa-devel] [PATCH 00/14] pending doc patches for 5.3-rc
by Mauro Carvalho Chehab 16 Jul '19

16 Jul '19

Those are the pending documentation patches after my pull request for this branch: git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media.git tags/docs/v5.3-1 Patches 1 to 13 were already submitted, but got rebased. Patch 14 is a new fixup one. Patches 1 and 2 weren't submitted before due to merge conflicts that are now solved upstream; Patch 3 fixes a series of random Documentation/* references that are pointing to the wrong places. Patch 4 fix a longstanding issue: every time a new book is added, conf.py need changes, in order to allow generating a PDF file. After the patch, conf.py will automatically recognize new books, saving the trouble of keeping adding documents to it. Patches 5 to 11 are due to fonts support when building translations.pdf. The main focus is to add xeCJK support. While doing it, I discovered some bugs at sphinx-pre-install script after running it with 7 different distributions. Patch 12 improves support for partial doc building. Currently, each subdir needs to have its own conf.py, in order to support partial doc build. After it, any Documentation subdir can be used to roduce html/pdf docs with: make SPHINXDIRS="foo bar" htmldocs (or pdfdocs, latexdocs, epubdocs, ...) Patch 13 is a cleanup patch: it simply get rid of all those extra conf.py files that aren't needed anymore. The only extra config file after it is this one: Documentation/media/conf_nitpick.py With enables some extra optional Sphinx features. Patch 14 adds Documentation/virtual to the main index.rst file and add a new *.rst file that was orphaned there. - After this series, there's just one more patch meant to be applied for 5.3, with is still waiting for some patches to be merged from linux-next: https://git.linuxtv.org/mchehab/experimental.git/commit/?id=b1b5dc7d7bbfbbf… Mauro Carvalho Chehab (14): docs: powerpc: convert docs to ReST and rename to *.rst docs: power: add it to to the main documentation index docs: fix broken doc references due to renames docs: pdf: add all Documentation/*/index.rst to PDF output docs: conf.py: add CJK package needed by translations docs: conf.py: only use CJK if the font is available scripts/sphinx-pre-install: fix script for RHEL/CentOS scripts/sphinx-pre-install: don't use LaTeX with CentOS 7 scripts/sphinx-pre-install: fix latexmk dependencies scripts/sphinx-pre-install: cleanup Gentoo checks scripts/sphinx-pre-install: seek for Noto CJK fonts for pdf output docs: load_config.py: avoid needing a conf.py just due to LaTeX docs docs: remove extra conf.py files docs: virtual: add it to the documentation body Documentation/PCI/pci-error-recovery.rst | 5 +- Documentation/RCU/rculist_nulls.txt | 2 +- Documentation/admin-guide/conf.py | 10 -- Documentation/conf.py | 30 +++- Documentation/core-api/conf.py | 10 -- Documentation/crypto/conf.py | 10 -- Documentation/dev-tools/conf.py | 10 -- .../devicetree/bindings/arm/idle-states.txt | 2 +- Documentation/doc-guide/conf.py | 10 -- Documentation/driver-api/80211/conf.py | 10 -- Documentation/driver-api/conf.py | 10 -- Documentation/driver-api/pm/conf.py | 10 -- Documentation/filesystems/conf.py | 10 -- Documentation/gpu/conf.py | 10 -- Documentation/index.rst | 3 + Documentation/input/conf.py | 10 -- Documentation/kernel-hacking/conf.py | 10 -- Documentation/locking/spinlocks.rst | 4 +- Documentation/maintainer/conf.py | 10 -- Documentation/media/conf.py | 12 -- Documentation/memory-barriers.txt | 2 +- Documentation/networking/conf.py | 10 -- Documentation/power/index.rst | 2 +- .../{bootwrapper.txt => bootwrapper.rst} | 28 +++- .../{cpu_families.txt => cpu_families.rst} | 23 +-- .../{cpu_features.txt => cpu_features.rst} | 6 +- Documentation/powerpc/{cxl.txt => cxl.rst} | 46 ++++-- .../powerpc/{cxlflash.txt => cxlflash.rst} | 10 +- .../{DAWR-POWER9.txt => dawr-power9.rst} | 15 +- Documentation/powerpc/{dscr.txt => dscr.rst} | 18 +- ...ecovery.txt => eeh-pci-error-recovery.rst} | 108 ++++++------ ...ed-dump.txt => firmware-assisted-dump.rst} | 117 +++++++------ Documentation/powerpc/{hvcs.txt => hvcs.rst} | 108 ++++++------ Documentation/powerpc/index.rst | 34 ++++ Documentation/powerpc/isa-versions.rst | 15 +- .../powerpc/{mpc52xx.txt => mpc52xx.rst} | 12 +- ...nv.txt => pci_iov_resource_on_powernv.rst} | 15 +- .../powerpc/{pmu-ebb.txt => pmu-ebb.rst} | 1 + Documentation/powerpc/ptrace.rst | 156 ++++++++++++++++++ Documentation/powerpc/ptrace.txt | 151 ----------------- .../{qe_firmware.txt => qe_firmware.rst} | 37 +++-- .../{syscall64-abi.txt => syscall64-abi.rst} | 29 ++-- ...al_memory.txt => transactional_memory.rst} | 45 ++--- Documentation/process/conf.py | 10 -- Documentation/sh/conf.py | 10 -- Documentation/sound/conf.py | 10 -- Documentation/sphinx/load_config.py | 27 ++- .../translations/ko_KR/memory-barriers.txt | 2 +- Documentation/userspace-api/conf.py | 10 -- Documentation/virtual/kvm/index.rst | 1 + Documentation/vm/conf.py | 10 -- Documentation/watchdog/hpwdt.rst | 2 +- Documentation/x86/conf.py | 10 -- MAINTAINERS | 14 +- arch/powerpc/kernel/exceptions-64s.S | 2 +- drivers/gpu/drm/drm_modes.c | 2 +- drivers/i2c/busses/i2c-nvidia-gpu.c | 2 +- drivers/scsi/hpsa.c | 4 +- drivers/soc/fsl/qe/qe.c | 2 +- drivers/tty/hvc/hvcs.c | 2 +- include/soc/fsl/qe/qe.h | 2 +- scripts/sphinx-pre-install | 118 ++++++++++--- 62 files changed, 738 insertions(+), 678 deletions(-) delete mode 100644 Documentation/admin-guide/conf.py delete mode 100644 Documentation/core-api/conf.py delete mode 100644 Documentation/crypto/conf.py delete mode 100644 Documentation/dev-tools/conf.py delete mode 100644 Documentation/doc-guide/conf.py delete mode 100644 Documentation/driver-api/80211/conf.py delete mode 100644 Documentation/driver-api/conf.py delete mode 100644 Documentation/driver-api/pm/conf.py delete mode 100644 Documentation/filesystems/conf.py delete mode 100644 Documentation/gpu/conf.py delete mode 100644 Documentation/input/conf.py delete mode 100644 Documentation/kernel-hacking/conf.py delete mode 100644 Documentation/maintainer/conf.py delete mode 100644 Documentation/media/conf.py delete mode 100644 Documentation/networking/conf.py rename Documentation/powerpc/{bootwrapper.txt => bootwrapper.rst} (93%) rename Documentation/powerpc/{cpu_families.txt => cpu_families.rst} (95%) rename Documentation/powerpc/{cpu_features.txt => cpu_features.rst} (97%) rename Documentation/powerpc/{cxl.txt => cxl.rst} (95%) rename Documentation/powerpc/{cxlflash.txt => cxlflash.rst} (98%) rename Documentation/powerpc/{DAWR-POWER9.txt => dawr-power9.rst} (95%) rename Documentation/powerpc/{dscr.txt => dscr.rst} (91%) rename Documentation/powerpc/{eeh-pci-error-recovery.txt => eeh-pci-error-recovery.rst} (82%) rename Documentation/powerpc/{firmware-assisted-dump.txt => firmware-assisted-dump.rst} (80%) rename Documentation/powerpc/{hvcs.txt => hvcs.rst} (91%) create mode 100644 Documentation/powerpc/index.rst rename Documentation/powerpc/{mpc52xx.txt => mpc52xx.rst} (91%) rename Documentation/powerpc/{pci_iov_resource_on_powernv.txt => pci_iov_resource_on_powernv.rst} (97%) rename Documentation/powerpc/{pmu-ebb.txt => pmu-ebb.rst} (99%) create mode 100644 Documentation/powerpc/ptrace.rst delete mode 100644 Documentation/powerpc/ptrace.txt rename Documentation/powerpc/{qe_firmware.txt => qe_firmware.rst} (95%) rename Documentation/powerpc/{syscall64-abi.txt => syscall64-abi.rst} (82%) rename Documentation/powerpc/{transactional_memory.txt => transactional_memory.rst} (93%) delete mode 100644 Documentation/process/conf.py delete mode 100644 Documentation/sh/conf.py delete mode 100644 Documentation/sound/conf.py delete mode 100644 Documentation/userspace-api/conf.py delete mode 100644 Documentation/vm/conf.py delete mode 100644 Documentation/x86/conf.py -- 2.21.0

1 1

[alsa-devel] [PATCH v3 0/5] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 16 Jul '19

16 Jul '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v2 to v3: - dw-hdmi-i2s-audio.c: Use fixed ID instead of auto ID. - rk3288_hdmi_analog.c: Use the fixed name hdmi-audio-codec for codec device. - rockchip_max98090: Use the fixed name hdmi-audio-codec for codec device. - rockchip_max98090: Fix the dependency of hdmi-codec in Kconfig. Cheng-Yi Chiang (5): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI FROMLIST: ASoC: rockchip_max98090: Add HDMI jack support .../gpu/drm/bridge/synopsys/dw-hdmi-audio.h | 3 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 12 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 55 ++++++++- include/sound/hdmi-codec.h | 16 +++ sound/soc/codecs/hdmi-codec.c | 45 +++++++ sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 8 files changed, 226 insertions(+), 27 deletions(-) -- 2.22.0.510.g264f2c817a-goog

3 11

[alsa-devel] [PATCH] ALSA: hda - Don't resume forcibly i915 HDMI/DP codec
by Takashi Iwai 16 Jul '19

16 Jul '19

We apply the codec resume forcibly at system resume callback for updating and syncing the jack detection state that may have changed during sleeping. This is, however, superfluous for the codec like Intel HDMI/DP, where the jack detection is managed via the audio component notification; i.e. the jack state change shall be reported sooner or later from the graphics side at mode change. This patch changes the codec resume callback to avoid the forcible resume conditionally with a new flag, codec->relaxed_resume, for reducing the resume time. The flag is set in the codec probe. Although this doesn't fix the entire bug mentioned in the bugzilla entry below, it's still a good optimization and some improvements are seen. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201901 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_codec.h | 2 ++ sound/pci/hda/hda_codec.c | 8 ++++++-- sound/pci/hda/patch_hdmi.c | 6 +++++- 3 files changed, 13 insertions(+), 3 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 8f46ff3449d5..871993696c5f 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -252,6 +252,8 @@ struct hda_codec { unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ unsigned int force_pin_prefix:1; /* Add location prefix */ unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ + unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ + #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5346631df1ec..e30e86ca6b72 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2941,15 +2941,19 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_force_resume(struct device *dev) { + struct hda_codec *codec = dev_to_hda_codec(dev); + bool forced_resume = !codec->relaxed_resume; int ret; /* The get/put pair below enforces the runtime resume even if the * device hasn't been used at suspend time. This trick is needed to * update the jack state change during the sleep. */ - pm_runtime_get_noresume(dev); + if (forced_resume) + pm_runtime_get_noresume(dev); ret = pm_runtime_force_resume(dev); - pm_runtime_put(dev); + if (forced_resume) + pm_runtime_put(dev); return ret; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 40323d91f9e4..72000f6fedca 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2292,8 +2292,10 @@ static void generic_hdmi_free(struct hda_codec *codec) struct hdmi_spec *spec = codec->spec; int pin_idx, pcm_idx; - if (codec_has_acomp(codec)) + if (codec_has_acomp(codec)) { snd_hdac_acomp_register_notifier(&codec->bus->core, NULL); + codec->relaxed_resume = 0; + } for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); @@ -2566,6 +2568,8 @@ static void register_i915_notifier(struct hda_codec *codec) spec->drm_audio_ops.pin_eld_notify = intel_pin_eld_notify; snd_hdac_acomp_register_notifier(&codec->bus->core, &spec->drm_audio_ops); + /* no need for forcible resume for jack check thanks to notifier */ + codec->relaxed_resume = 1; } /* setup_stream ops override for HSW+ */ -- 2.16.4

1 0

[alsa-devel] [PATCH 0/2] expand dshare to allow audio clock when not streaming data
by Adam Miartus 16 Jul '19

16 Jul '19

The general idea of these changes is to be able to open the device without defined bindings. This would start the audio clock without blocking a channel. This might be useful for hardware that requires running clock early at system start and to be available even without application streaming audio data. Andreas Pape (2): alsa: pcm: add unsupported OPS alsa: dshare: allow missing bindings src/pcm/Makefile.am | 4 +- src/pcm/pcm_dshare.c | 72 +++++++++++-- src/pcm/pcm_unsupported.c | 265 ++++++++++++++++++++++++++++++++++++++++++++++ src/pcm/pcm_unsupported.h | 167 +++++++++++++++++++++++++++++ 4 files changed, 495 insertions(+), 13 deletions(-) create mode 100644 src/pcm/pcm_unsupported.c create mode 100644 src/pcm/pcm_unsupported.h -- 2.7.4

3 7

[alsa-devel] [PATCH v2 0/6]
by vanitha.channaiah＠in.bosch.com 16 Jul '19

16 Jul '19

- Updated the patches by incorporating review comments from Takashi Iwai-san [PATCH 1/5] pcm: direct: Add generic hw_ptr_alignment function for dmix, dshare and dsnoop [Takashi Iwai:] > The patch description needs rephrasing. What actually this does is to move the code from pcm_dmix.c to pcm_direct.c > and its header so that the helper code can be re-used by other direct-PCM plugins. - Commit message is rephrased as suggested. - New commit: [PATCH v2 1/6] pcm: direct: Add generic hw_ptr_alignment function for [PATCH 2/5] pcm: dshare: Added "hw_ptr_alignment" option in configuration for alignment of slave pointers [Takashi Iwai:] > Again, this patch description is too ambiguous. > And, if it enables the hw_ptr_alignment option, update the documentation as well. - Commit message is explained in detail for the changes done. - Documentation updated. - New commit: [PATCH v2 2/6] pcm: dshare: Added "hw_ptr_alignment" option in [PATCH 3/5] pcm: dsnoop: Added "hw_ptr_alignment" option in configuration for slave pointer alignment [Takashi Iwai:] > Ditto as patch 2, the description is too ambiguous, and the update of documentation is missing. > It's not good to change the helper function semantics out of sudden, even without any description. - Commit message is explained in detail for the changes done. - Documentation updated. - Divided the patch with commit ("pcm: dsnoop: Add hw_ptr_alignment option in configuration") into additional patch commit ("pcm: direct: Round up of slave_app_ptr pointer if buffer") - Usecase scenario is described for the changes done in helper function. - New commit: [PATCH v2 3/6] pcm: dsnoop: Added "hw_ptr_alignment" option in [PATCH v2 4/6] pcm: direct: Round up of slave_app_ptr pointer if buffer [PATCH 4/5] pcm: restructuring sw params function [Takashi Iwai:] > I see no reason to do that. Please describe. - Commit message is explained in detail why reformating was done. - New commit: [PATCH v2 5/6] pcm: restructuring sw params function [PATCH 5/5] pcm: Update pcm->avail_min with needed_slave_avail_min, after reading unaligned frames [Takashi Iwai:] > This kind of changes in the core code should be avoided as much as possible, especially if it's only relevant with the specific plugins. > Sorry, this isn't convincing enough. If this is a MUST, please clarify better. - Commit message is explained in detail with the generic usecase and specific use case we came across. - New commit: [PATCH v2 6/6] pcm: Update pcm->avail_min with needed_slave_avail_min,

4 20

[alsa-devel] [PATCH] ALSA: seq: Break too long mutex context in the write loop
by Takashi Iwai 16 Jul '19

16 Jul '19

The fix for the racy writes and ioctls to sequencer widened the application of client->ioctl_mutex to the whole write loop. Although it does unlock/relock for the lengthy operation like the event dup, the loop keeps the ioctl_mutex for the whole time in other situations. This may take quite long time if the user-space would give a huge buffer, and this is a likely cause of some weird behavior spotted by syzcaller fuzzer. This patch puts a simple workaround, just adding a mutex break in the loop when a large number of events have been processed. This shouldn't hit any performance drop because the threshold is set high enough for usual operations. Fixes: 7bd800915677 ("ALSA: seq: More protection for concurrent write and ioctl races") Reported-by: syzbot+97aae04ce27e39cbfca9(a)syzkaller.appspotmail.com Reported-by: syzbot+4c595632b98bb8ffcc66(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index a60e7a17f0b8..7737b2670064 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1021,7 +1021,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, { struct snd_seq_client *client = file->private_data; int written = 0, len; - int err; + int err, handled; struct snd_seq_event event; if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT)) @@ -1034,6 +1034,8 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, if (!client->accept_output || client->pool == NULL) return -ENXIO; + repeat: + handled = 0; /* allocate the pool now if the pool is not allocated yet */ mutex_lock(&client->ioctl_mutex); if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) { @@ -1093,12 +1095,19 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, 0, 0, &client->ioctl_mutex); if (err < 0) break; + handled++; __skip_event: /* Update pointers and counts */ count -= len; buf += len; written += len; + + /* let's have a coffee break if too many events are queued */ + if (++handled >= 200) { + mutex_unlock(&client->ioctl_mutex); + goto repeat; + } } out: -- 2.16.4

1 0

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda: Initialize HDA controller after i915 init
by Ranjani Sridharan 15 Jul '19

15 Jul '19

On some platforms, sound card registration fails when a HDMI monitor is not connected. This is caused by a recent commit that switched the order in which the HDA controller and the i915 are initialized. Initializing the i915 before initializing the HDA controller fixes the problem. Fixes: be1b577d01787c ("ASoC: SOF: Intel: hda: fix the hda init chip") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 136f98bf5d7e..82f61c4e74bd 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -321,10 +321,23 @@ static int hda_init_caps(struct snd_sof_dev *sdev) if (bus->ppcap) dev_dbg(sdev->dev, "PP capability, will probe DSP later.\n"); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* init i915 and HDMI codecs */ + ret = hda_codec_i915_init(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: init i915 and HDMI codec failed\n"); + return ret; + } +#endif + + /* Init HDA controller after i915 init */ ret = hda_dsp_ctrl_init_chip(sdev, true); if (ret < 0) { dev_err(bus->dev, "error: init chip failed with ret: %d\n", ret); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + hda_codec_i915_exit(sdev); +#endif return ret; } @@ -332,13 +345,6 @@ static int hda_init_caps(struct snd_sof_dev *sdev) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); - /* init i915 and HDMI codecs */ - ret = hda_codec_i915_init(sdev); - if (ret < 0) { - dev_err(sdev->dev, "error: no HDMI audio devices found\n"); - return ret; - } - /* codec detection */ if (!bus->codec_mask) { dev_info(bus->dev, "no hda codecs found!\n"); -- 2.17.1

1 0

[alsa-devel] [RESEND PATCH v2] ASoC: SOF: Intel: hda: Make hdac_device device-managed
by Ranjani Sridharan 15 Jul '19

15 Jul '19

snd_hdac_ext_bus_device_exit() has been recently modified to no longer free the hdac device. SOF allocates memory for hdac_device and hda_hda_priv with kzalloc. Make them device-managed instead so that they will be freed when the SOF driver is unloaded. Because of the above change, hda_codec is device-managed and it will be freed when the ASoC device is removed. Freeing the codec in snd_hda_codec_dev_release() leads to kernel panic while unloading and reloading the ASoC driver. So, avoid freeing the hda_codec for ASoC driver. This is done in the same patch to avoid bisect failure. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 +++++++- sound/soc/sof/intel/hda-codec.c | 6 ++---- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6c51b8363f8b..a2e23d7e768f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -846,7 +846,13 @@ static void snd_hda_codec_dev_release(struct device *dev) snd_hda_sysfs_clear(codec); kfree(codec->modelname); kfree(codec->wcaps); - kfree(codec); + + /* + * In the case of ASoC HD-audio, hda_codec is device managed. + * It will be freed when the ASoC device is removed. + */ + if (codec->core.type == HDA_DEV_LEGACY) + kfree(codec); } #define DEV_NAME_LEN 31 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index b8b37f082309..0d8437b080bf 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -62,8 +62,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) address, resp); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hda_priv = kzalloc(sizeof(*hda_priv), GFP_KERNEL); + hda_priv = devm_kzalloc(sdev->dev, sizeof(*hda_priv), GFP_KERNEL); if (!hda_priv) return -ENOMEM; @@ -82,8 +81,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) return 0; #else - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hdev = kzalloc(sizeof(*hdev), GFP_KERNEL); + hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) return -ENOMEM; -- 2.17.1

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[alsa-devel] [PATCH AUTOSEL 5.1 201/219] ASoC: audio-graph-card: fix use-after-free in graph_for_each_link
by Sasha Levin 15 Jul '19

15 Jul '19

From: Wen Yang <wen.yang99(a)zte.com.cn> [ Upstream commit 1bcc1fd64e4dd903f4d868a9e053986e3b102713 ] After calling of_node_put() on the codec_ep and codec_port variables, they are still being used, which may result in use-after-free. We fix this issue by calling of_node_put() after the last usage. Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562229530-8121-1-git-send-email-wen.yang99@zte.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 69bc4848d787..f730830fb36c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -460,9 +460,6 @@ static int graph_for_each_link(struct graph_priv *priv, codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_get_parent(codec_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); graph_get_conversion(dev, codec_ep, &adata); @@ -482,6 +479,9 @@ static int graph_for_each_link(struct graph_priv *priv, else ret = func_noml(priv, cpu_ep, codec_ep, li); + of_node_put(codec_ep); + of_node_put(codec_port); + if (ret < 0) return ret; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.2 230/249] ASoC: audio-graph-card: fix use-after-free in graph_for_each_link
by Sasha Levin 15 Jul '19

15 Jul '19

From: Wen Yang <wen.yang99(a)zte.com.cn> [ Upstream commit 1bcc1fd64e4dd903f4d868a9e053986e3b102713 ] After calling of_node_put() on the codec_ep and codec_port variables, they are still being used, which may result in use-after-free. We fix this issue by calling of_node_put() after the last usage. Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562229530-8121-1-git-send-email-wen.yang99@zte.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index ec7e673ba475..70ed28d97d49 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -435,9 +435,6 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_get_parent(codec_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); graph_parse_convert(dev, codec_ep, &adata); @@ -457,6 +454,9 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, else ret = func_noml(priv, cpu_ep, codec_ep, li); + of_node_put(codec_ep); + of_node_put(codec_port); + if (ret < 0) return ret; -- 2.20.1

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[alsa-devel] [PATCH 0/8] docs: some improvements when producing PDF files
by Mauro Carvalho Chehab 15 Jul '19

15 Jul '19

Hi Jon, This series addresses your concerns related to CJK fonts that are needed for translations.pdf. It touches only the documentation build system, not the docs themselves. It ended to be bigger than I originally foreseen, as I found several issues when running "make pdfdocs" for the distros that are recognized by the scripts/sphinx-pre-install script. It also took a lot of time, as I tested it with several VMs (each one updated to latest packages): - Fedora 30, CentOS 7, Mageia 7, ArchLinux, Ubuntu 18.04, Gentoo, OpenSuse Tumbleweed. Patch 1 addresses an issue that could be related to the fact that I don't use openSUSE. Basically, I was unable to find the right package for texlive to use CJK fonts on openSUSE. [1]. So, the first patch on this series adds a workaround: if the needed CJK font is not found on a system, conf.py won't use xeCjk extension. That sounds a good thing to have, as other distros may not package it, or maybe the one building the doc is not that interested on translations.pdf file; [1] I actually found some, but they are not recognized with the font name conf.py is expecting ("Noto Sans CJK SC"). Perhaps SUSE uses a different name for those fonts? Patch 2 fixes the logic with recognizes CentOS/RHEL; Patch 3 is another workaround: CentOS 7 (and similar distros) don't package all texlive packages we need. So, it just ignores PDF when recommending packages on such distros, and point to a URL with explains how to install TexLive outside distro-specific package management (for the brave enough people); Patch 4 fixes latexmk dependency on a few distros; Patch 5 suppreses a Gentoo specific instruction if the user already followed in the past; Patch 6 is the one that actually does what you requested. Patch 7 solves an issue when SPHINXDIRS is used with make pdfdocs: right now, using it will produce a lot of warnings and won't do anything, if a dir-specific conf.py file is not found. With the patch, latex_documents are now properly updated when SPHINXDIRS is used. Patch 8 is a cleanup: with patch 7 applied, we don't need to have anymore any conf.py file due to pdfdocs. With regard to the load_config.py extension, It keeps accepting custom configuration. That's helpful if someone wants, for example, to have something like: Documentation/media/conf_nitpick.py with would enable extra nitpick options if one wants that. - Jon, Please let me know if you prefer if I submit those together with the big pile of doc files I have, or if you prefer adding (some of?) them on your tree after the merge window. Regards, Mauro Mauro Carvalho Chehab (8): docs: conf.py: only use CJK if the font is available scripts/sphinx-pre-install: fix script for RHEL/CentOS scripts/sphinx-pre-install: don't use LaTeX with CentOS 7 scripts/sphinx-pre-install: fix latexmk dependencies scripts/sphinx-pre-install: cleanup Gentoo checks scripts/sphinx-pre-install: seek for Noto CJK fonts for pdf output docs: load_config.py: avoid needing a conf.py just due to LaTeX docs docs: remove extra conf.py files Documentation/admin-guide/conf.py | 10 --- Documentation/conf.py | 13 ++- Documentation/core-api/conf.py | 10 --- Documentation/crypto/conf.py | 10 --- Documentation/dev-tools/conf.py | 10 --- Documentation/doc-guide/conf.py | 10 --- Documentation/driver-api/80211/conf.py | 10 --- Documentation/driver-api/conf.py | 10 --- Documentation/driver-api/pm/conf.py | 10 --- Documentation/filesystems/conf.py | 10 --- Documentation/gpu/conf.py | 10 --- Documentation/input/conf.py | 10 --- Documentation/kernel-hacking/conf.py | 10 --- Documentation/maintainer/conf.py | 10 --- Documentation/media/conf.py | 12 --- Documentation/networking/conf.py | 10 --- Documentation/process/conf.py | 10 --- Documentation/sh/conf.py | 10 --- Documentation/sound/conf.py | 10 --- Documentation/sphinx/load_config.py | 25 +++++- Documentation/userspace-api/conf.py | 10 --- Documentation/vm/conf.py | 10 --- Documentation/x86/conf.py | 10 --- scripts/sphinx-pre-install | 118 ++++++++++++++++++++----- 24 files changed, 131 insertions(+), 237 deletions(-) delete mode 100644 Documentation/admin-guide/conf.py delete mode 100644 Documentation/core-api/conf.py delete mode 100644 Documentation/crypto/conf.py delete mode 100644 Documentation/dev-tools/conf.py delete mode 100644 Documentation/doc-guide/conf.py delete mode 100644 Documentation/driver-api/80211/conf.py delete mode 100644 Documentation/driver-api/conf.py delete mode 100644 Documentation/driver-api/pm/conf.py delete mode 100644 Documentation/filesystems/conf.py delete mode 100644 Documentation/gpu/conf.py delete mode 100644 Documentation/input/conf.py delete mode 100644 Documentation/kernel-hacking/conf.py delete mode 100644 Documentation/maintainer/conf.py delete mode 100644 Documentation/media/conf.py delete mode 100644 Documentation/networking/conf.py delete mode 100644 Documentation/process/conf.py delete mode 100644 Documentation/sh/conf.py delete mode 100644 Documentation/sound/conf.py delete mode 100644 Documentation/userspace-api/conf.py delete mode 100644 Documentation/vm/conf.py delete mode 100644 Documentation/x86/conf.py -- 2.21.0

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[alsa-devel] [PATCH] sound: pci: rme9652: Unneeded variable: "result".
by Hariprasad Kelam 15 Jul '19

15 Jul '19

This patch fixes below issue reported by coccicheck sound/pci/rme9652/rme9652.c:2161:5-11: Unneeded variable: "result". Return "0" on line 2167 Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/pci/rme9652/rme9652.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index cb9818a..4c851f8 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2158,13 +2158,12 @@ static int snd_rme9652_prepare(struct snd_pcm_substream *substream) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); unsigned long flags; - int result = 0; spin_lock_irqsave(&rme9652->lock, flags); if (!rme9652->running) rme9652_reset_hw_pointer(rme9652); spin_unlock_irqrestore(&rme9652->lock, flags); - return result; + return 0; } static const struct snd_pcm_hardware snd_rme9652_playback_subinfo = -- 2.7.4

2 1

[alsa-devel] [PATCH] sound: pci: emu10k1: Remove unneeded variable "change"
by Hariprasad Kelam 15 Jul '19

15 Jul '19

fix below issue reported by coccicheck sound/pci/emu10k1/emu10k1x.c:1077:5-11: Unneeded variable: "change". Return "0" on line 1092 Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/pci/emu10k1/emu10k1x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 67d6473..9cf8183 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1074,7 +1074,6 @@ static int snd_emu10k1x_shared_spdif_put(struct snd_kcontrol *kcontrol, { struct emu10k1x *emu = snd_kcontrol_chip(kcontrol); unsigned int val; - int change = 0; val = ucontrol->value.integer.value[0] ; @@ -1089,7 +1088,7 @@ static int snd_emu10k1x_shared_spdif_put(struct snd_kcontrol *kcontrol, snd_emu10k1x_ptr_write(emu, ROUTING, 0, 0x1003F); snd_emu10k1x_gpio_write(emu, 0x1080); } - return change; + return 0; } static const struct snd_kcontrol_new snd_emu10k1x_shared_spdif = -- 2.7.4

3 3

[alsa-devel] [PATCH] sound: pci: au88x0: Remove unneeded variable: "changed"
by Hariprasad Kelam 15 Jul '19

15 Jul '19

Fix below issues reported by coccicheck sound/pci/au88x0/au88x0_a3d.c:821:8-15: Unneeded variable: "changed". Return "1" on line 834 sound/pci/au88x0/au88x0_a3d.c:768:5-12: Unneeded variable: "changed". Return "1" on line 777 sound/pci/au88x0/au88x0_a3d.c:804:5-12: Unneeded variable: "changed". Return "1" on line 813 sound/pci/au88x0/au88x0_a3d.c:786:8-15: Unneeded variable: "changed". Return "1" on line 796 Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/pci/au88x0/au88x0_a3d.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 7347103..2db183f 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -765,7 +765,7 @@ snd_vortex_a3d_hrtf_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { a3dsrc_t *a = kcontrol->private_data; - int changed = 1, i; + int i; int coord[6]; for (i = 0; i < 6; i++) coord[i] = ucontrol->value.integer.value[i]; @@ -774,7 +774,7 @@ snd_vortex_a3d_hrtf_put(struct snd_kcontrol *kcontrol, vortex_a3d_coord2hrtf(a->hrtf[1], coord); a3dsrc_SetHrtfTarget(a, a->hrtf[0], a->hrtf[1]); a3dsrc_SetHrtfCurrent(a, a->hrtf[0], a->hrtf[1]); - return changed; + return 1; } static int @@ -783,7 +783,7 @@ snd_vortex_a3d_itd_put(struct snd_kcontrol *kcontrol, { a3dsrc_t *a = kcontrol->private_data; int coord[6]; - int i, changed = 1; + int i; for (i = 0; i < 6; i++) coord[i] = ucontrol->value.integer.value[i]; /* Translate orientation coordinates to a3d params. */ @@ -793,7 +793,7 @@ snd_vortex_a3d_itd_put(struct snd_kcontrol *kcontrol, a3dsrc_SetItdTarget(a, a->itd[0], a->itd[1]); a3dsrc_SetItdCurrent(a, a->itd[0], a->itd[1]); a3dsrc_SetItdDline(a, a->dline); - return changed; + return 1; } static int @@ -801,7 +801,6 @@ snd_vortex_a3d_ild_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { a3dsrc_t *a = kcontrol->private_data; - int changed = 1; int l, r; /* There may be some scale tranlation needed here. */ l = ucontrol->value.integer.value[0]; @@ -810,7 +809,7 @@ snd_vortex_a3d_ild_put(struct snd_kcontrol *kcontrol, /* Left Right panning. */ a3dsrc_SetGainTarget(a, l, r); a3dsrc_SetGainCurrent(a, l, r); - return changed; + return 1; } static int @@ -818,7 +817,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { a3dsrc_t *a = kcontrol->private_data; - int i, changed = 1; + int i; int params[6]; for (i = 0; i < 6; i++) params[i] = ucontrol->value.integer.value[i]; @@ -831,7 +830,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol, a3dsrc_SetAtmosCurrent(a, a->filter[0], a->filter[1], a->filter[2], a->filter[3], a->filter[4]); - return changed; + return 1; } static const struct snd_kcontrol_new vortex_a3d_kcontrol = { -- 2.7.4

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[alsa-devel] Dell ALC295 Headphone Mic can't record
by Kailang 15 Jul '19

15 Jul '19

Hi Takashi, Attach patch was for Dell ALC295 Headphone Mic can't record. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] sound: soc: codecs: wcd9335: fix "conversion to bool not needed here"
by Hariprasad Kelam 13 Jul '19

13 Jul '19

Fix below issue reported by coccicheck sound/soc/codecs/wcd9335.c:3991:25-30: WARNING: conversion to bool not needed here Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/soc/codecs/wcd9335.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 1bbbe42..85a8d10 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -3988,12 +3988,7 @@ static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) regmap_read(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_RX_SOURCE0 + j, &val); if (val) { - if (!tx) - reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + - (port_id / 8); - else - reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + - (port_id / 8); + reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + (port_id / 8); regmap_read( wcd->if_regmap, reg, &int_val); /* -- 2.7.4

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[alsa-devel] [PATCH] ASoC: audio-graph-card: fix type mismatch warning
by Arnd Bergmann 13 Jul '19

13 Jul '19

The new temporary variable is lacks a 'const' annotation: sound/soc/generic/audio-graph-card.c:87:7: error: assigning to 'u32 *' (aka 'unsigned int *') from 'const void *' discards qualifiers [-Werror,-Wincompatible-pointer-types-discards-qualifiers] Fixes: c152f8491a8d ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/generic/audio-graph-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c8abb86afefa..288df245b2f0 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -63,7 +63,7 @@ static int graph_get_dai_id(struct device_node *ep) struct device_node *endpoint; struct of_endpoint info; int i, id; - u32 *reg; + const u32 *reg; int ret; /* use driver specified DAI ID if exist */ -- 2.20.0

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: add missing const at graph_get_dai_id()
by Kuninori Morimoto 12 Jul '19

12 Jul '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()") fixups use-after-free issue, but, it need to use "const" for reg. This patch adds it. We will have below without this patch LINUX/sound/soc/generic/audio-graph-card.c: In function 'graph_get_dai_id': LINUX/sound/soc/generic/audio-graph-card.c:87:7: warning: assignment discards\ 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] reg = of_get_property(node, "reg", NULL); Fixes: c152f8491a8d9 ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 343ede8..ebf2ca3 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -63,7 +63,7 @@ static int graph_get_dai_id(struct device_node *ep) struct device_node *endpoint; struct of_endpoint info; int i, id; - u32 *reg; + const u32 *reg; int ret; /* use driver specified DAI ID if exist */ -- 2.7.4

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[alsa-devel] How to treat old style platform components
by Daniel Baluta 12 Jul '19

12 Jul '19

HI Morimoto-san, I'm using simple-card with an out of tree PCM DMA driver which basically just does this: static struct snd_soc_component_driver imx_soc_platform = { .ops» » = &imx_pcm_ops, .pcm_new» = imx_pcm_new, }; Then somewhere in the DAI driver we call: devm_snd_soc_register_component(dev, &imx_soc_platform, NULL, 0); I'm using simple-card with the following configuration: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/tree/ar… After latest changes it seems that imx_pcm_ops->open() function is no longer called. Any idea? Please feel free to not waste any time on this, as we are using an out of tree driver. thanks, Daniel.

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[alsa-devel] [PATCH] ASoC: audio-graph-card: Constify reg in graph_get_dai_id
by Nathan Chancellor 12 Jul '19

12 Jul '19

clang errors: sound/soc/generic/audio-graph-card.c:87:7: error: assigning to 'u32 *' (aka 'unsigned int *') from 'const void *' discards qualifiers [-Werror,-Wincompatible-pointer-types-discards-qualifiers] reg = of_get_property(node, "reg", NULL); ^ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 1 error generated. Move the declaration up a bit to keep the reverse christmas tree look. Fixes: c152f8491a8d ("ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id()") Link: https://github.com/ClangBuiltLinux/linux/issues/599 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/generic/audio-graph-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c8abb86afefa..c0d262a2ce2c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -62,8 +62,8 @@ static int graph_get_dai_id(struct device_node *ep) struct device_node *node; struct device_node *endpoint; struct of_endpoint info; + const u32 *reg; int i, id; - u32 *reg; int ret; /* use driver specified DAI ID if exist */ -- 2.22.0

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Re: [alsa-devel] DW-DMA: Probe failures on broadwell
by Andy Shevchenko 11 Jul '19

11 Jul '19

(I think this is okay to go on public here) +Cc: Pierre, Liam, DMA Engine ML, ASoC ML On Mon, Jul 08, 2019 at 01:50:07PM -0700, Curtis Malainey wrote: > Hello Andy, > > I am reaching out to you as you are the main author for > drivers/dma/dw/core.c and we are seeing a failure in there on some of > our samus devices. On certain device we are seeing the following > failure (debugging enabled in this log.) > > [ 3.587515] sst-acpi INT3438:00: DW_PARAMS: 0x00000000 AFAIK, when Synopsys DesignWare DMA IP is in use in Intel, we almost always (yes, I know couple of exceptions), enable auto configuration block. Thus, the *private* DMA controllers used in Broadwell are actually Intel iDMA 32-bit. Nowadays the differences can be seen in drivers/dma/dw/idma32.c. Note, DMA in the driver is used solely for firmware load, simplest workaround is to disable DMA. Though, personally, for sake of test coverage, I would like to see how it works in DMA case. > [ 3.587519] haswell-pcm-audio haswell-pcm-audio: error: DMA device > register failed > [ 3.587524] haswell-pcm-audio haswell-pcm-audio: sst_dma_new failed -22 > [ 3.598010] bdw-rt5677 bdw-rt5677: ASoC: failed to init link System PCM > > I don't have the datasheets for this component but I am wondering if > you could help us troubleshoot this bug it would be greatly > appreciated if possible. I am not sure if we are seeing a hardware > boot failure or a boot race. I was hoping you could shed some light on > this and if it is a boot race help us recover from it. I know Intel > relies on no defers typically but it would be nice to see if we can > fix this. Below is where I have traced the error source to in core.c. So, the correct fix is to provide a platform data, like it's done in drivers/dma/dw/pci.c::idma32_pdata, in the sst-firmware.c::dw_probe(), and call idma32_dma_probe() with idma32_dma_remove() respectively on removal stage. (It will require latest patches to be applied, which are material for v5.x) > dw_params = dma_readl(dw, DW_PARAMS); > dev_dbg(chip->dev, "DW_PARAMS: 0x%08x\n", dw_params); > > autocfg = dw_params >> DW_PARAMS_EN & 1; > if (!autocfg) { > err = -EINVAL; > goto err_pdata; > } > Let me know if you have any ideas on how to mitigate this, thanks. -- With Best Regards, Andy Shevchenko

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[alsa-devel] [PATCH 1/2] ASoC: rt1011: fix DC calibration offset not applying
by shumingf＠realtek.com 11 Jul '19

11 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> There are two issues to fix: - DC offset calibration data will be reset after stopping playback. - DC offset calibration data should be applied in the initial setting. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 5605b660f4bf..0a6ff13d76e1 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -39,7 +39,7 @@ static const struct reg_sequence init_list[] = { { RT1011_POWER_9, 0xa840 }, { RT1011_ADC_SET_5, 0x0a20 }, - { RT1011_DAC_SET_2, 0xa232 }, + { RT1011_DAC_SET_2, 0xa032 }, { RT1011_ADC_SET_1, 0x2925 }, { RT1011_SPK_PRO_DC_DET_1, 0xb00c }, @@ -1917,7 +1917,7 @@ static int rt1011_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT1011_SYSTEM_RESET_2, 0x0000); snd_soc_component_write(component, - RT1011_SYSTEM_RESET_3, 0x0000); + RT1011_SYSTEM_RESET_3, 0x0001); snd_soc_component_write(component, RT1011_SYSTEM_RESET_1, 0x003f); snd_soc_component_write(component, -- 2.22.0

2 1

[alsa-devel] [PATCH v2 0/6] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 11 Jul '19

11 Jul '19

VAG power control is improved to fit the manual. This fixes as minimum one bug: if customer muxes Headphone to Line-In right after boot w/o playing any sound, the VAG power remains off that leads to poor sound quality from line-in. I.e. after boot: - Connect sound source to Line-In jack; - Connect headphone to HP jack; - Run following commands: $ amixer set 'Headphone' 80% $ amixer set 'Headphone Mux' LINE_IN Also this series includes fixes of non-important bugs in sgtl5000 codec driver. Fix patch formatting in patchset v2. Oleksandr Suvorov (6): ASoC: sgtl5000: Fix definition of VAG Ramp Control ASoC: sgtl5000: add ADC mute control ASoC: sgtl5000: Fix of unmute outputs on probe ASoC: sgtl5000: Fix charge pump source assignment ASoC: Define a set of DAPM pre/post-up events ASoC: sgtl5000: Improve VAG power and mute control include/sound/soc-dapm.h | 2 + sound/soc/codecs/sgtl5000.c | 250 ++++++++++++++++++++++++++++++------ sound/soc/codecs/sgtl5000.h | 2 +- 3 files changed, 212 insertions(+), 42 deletions(-) -- 2.20.1

2 11

[alsa-devel] [PATCH 2/2] ASoC: rt1011: Add R0 temperature and TDM1 ADC2DAT Swap control
by shumingf＠realtek.com 11 Jul '19

11 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> - The user level application could set the R0 temperature after booting system. The degree Celsius of R0 temperature store in the non-volatile space when doing R0 calibration. - TDM1 ADC2DAT Swap controls use to control TDM slot2/3 data Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 9 ++++++++- sound/soc/codecs/rt1011.h | 1 + 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 0a6ff13d76e1..b25b4c1127ca 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1029,6 +1029,8 @@ static const char * const rt1011_tdm_adc_swap_select[] = { static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, rt1011_tdm_adc_swap_select); +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, + rt1011_tdm_adc_swap_select); static void rt1011_reset(struct regmap *regmap) { @@ -1333,7 +1335,8 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* TDM1 Data Out Selection */ SOC_ENUM("TDM1 DOUT Source", rt1011_tdm1_adc1_dat_enum), SOC_ENUM("TDM1 DOUT Location", rt1011_tdm1_adc1_loc_enum), - SOC_ENUM("TDM1 ADCDAT Swap Select", rt1011_tdm_adc1_1_enum), + SOC_ENUM("TDM1 ADC1DAT Swap Select", rt1011_tdm_adc1_1_enum), + SOC_ENUM("TDM1 ADC2DAT Swap Select", rt1011_tdm_adc2_1_enum), /* Data Out Mode */ SOC_ENUM("I2S ADC DOUT Mode", rt1011_adc_dout_mode_enum), @@ -1355,6 +1358,10 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_SINGLE_EXT("R0 Calibration", SND_SOC_NOPM, 0, 1, 0, rt1011_r0_cali_get, rt1011_r0_cali_put), RT1011_R0_LOAD("R0 Load Mode"), + + /* R0 temperature */ + SOC_SINGLE("R0 Temperature", RT1011_STP_INITIAL_RESISTANCE_TEMP, + 2, 255, 0), }; static int rt1011_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 98a38800c4df..5d431d7f1d16 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -227,6 +227,7 @@ #define RT1011_STP_CALIB_RS_TEMP 0x152a #define RT1011_INIT_RECIPROCAL_REG_24_16 0x1538 #define RT1011_INIT_RECIPROCAL_REG_15_0 0x1539 +#define RT1011_STP_INITIAL_RESISTANCE_TEMP 0x153c #define RT1011_STP_ALPHA_RECIPROCAL_MSB 0x153e #define RT1011_SPK_RESISTANCE_1 0x1544 #define RT1011_SPK_RESISTANCE_2 0x1546 -- 2.22.0

1 0

Re: [alsa-devel] [PATCH V3 2/2] ASoC: fsl_esai: recover the channel swap after xrun
by S.j. Wang 11 Jul '19

11 Jul '19

> > Hi Shengjiu, > > Mostly looks good to me, just some small comments. > > On Mon, Jul 08, 2019 at 02:38:52PM +0800, shengjiu.wang(a)nxp.com wrote: > > > +static void fsl_esai_hw_reset(unsigned long arg) { > > + struct fsl_esai *esai_priv = (struct fsl_esai *)arg; > > + u32 saisr, tfcr, rfcr; > > + bool tx = true, rx = false, enabled[2]; > > Could we swap the lines of u32 and bool? It'd look better. > > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, > > + ESAI_xCR_xPR_MASK, ESAI_xCR_xPR); > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, > > + ESAI_xCR_xPR_MASK, ESAI_xCR_xPR); > > Let's add a line of comments for these two: > /* Enforce ESAI personal resets for both TX and RX */ > > > + /* > > + * Restore registers by regcache_sync, and ignore > > + * return value > > + */ > > Could fit into single-line? > > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, > > + ESAI_xCR_xPR_MASK, 0); > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, > > + ESAI_xCR_xPR_MASK, 0); > > + > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, > > + ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO)); > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, > > + ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO)); > > Could remove the blank line and add a line of comments: > /* Remove ESAI personal resets by configuring PCRC and PRRC also */ > > Btw, I still feel this personal reset can be stuffed into one of the wrapper > functions. But let's keep this simple for now. > > > + regmap_read(esai_priv->regmap, REG_ESAI_SAISR, &saisr); > > Why do we read saisr here? All its bits would get cleared by the hardware > reset. If it's a must to clear again, we should add a line of comments to > emphasize it. This line can be removed. Best regards Wang Shengjiu

1 0

[alsa-devel] [PATCH V3 0/2] recover the channel swap after xrun
by shengjiu.wang＠nxp.com 11 Jul '19

11 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> recover the channel swap after xrun Shengjiu Wang (2): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun sound/soc/fsl/fsl_esai.c | 270 ++++++++++++++++++++++++++++----------- 1 file changed, 197 insertions(+), 73 deletions(-) Changes in V3 - update code sytle. Changes in v2 - add one patch for wrap operations to functions. - fix some coding style issue -- 2.21.0

2 3

[alsa-devel] [RFC PATCH] ASoC: max98357a: use mdelay for sdmode-delay
by Tzung-Bi Shih 10 Jul '19

10 Jul '19

max98357a_daiops_trigger() is possible to be called in atomic context if the .nonatomic flag is equal to 0 in the DAI links. When cancel_delayed_work_sync() in max98357a_daiops_trigger() is called in atomic context, kernel emits the following message: "BUG: sleeping function called from invalid context". According to the DT binding document, value less than or equal to 5ms of sdmod-delay should be sufficient to avoid the pop noise. Use mdelay (i.e. busy loop) for such low delay should be acceptable. Fixes: cec5b01f8f1c ("ASoC: max98357a: avoid speaker pop when playback startup") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Hi, Not sure whether this fix is acceptable or not. Because in the document "timers-howto.txt", it states "In general, use of mdelay is discouraged and code should be refactored to allow for the use of msleep." sound/soc/codecs/max98357a.c | 25 ++++--------------------- 1 file changed, 4 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 6f0e28f903bf..16313b973eaa 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -20,20 +20,10 @@ #include <sound/soc-dapm.h> struct max98357a_priv { - struct delayed_work enable_sdmode_work; struct gpio_desc *sdmode; unsigned int sdmode_delay; }; -static void max98357a_enable_sdmode_work(struct work_struct *work) -{ - struct max98357a_priv *max98357a = - container_of(work, struct max98357a_priv, - enable_sdmode_work.work); - - gpiod_set_value(max98357a->sdmode, 1); -} - static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -46,14 +36,12 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - queue_delayed_work(system_power_efficient_wq, - &max98357a->enable_sdmode_work, - msecs_to_jiffies(max98357a->sdmode_delay)); + mdelay(max98357a->sdmode_delay); + gpiod_set_value(max98357a->sdmode, 1); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - cancel_delayed_work_sync(&max98357a->enable_sdmode_work); gpiod_set_value(max98357a->sdmode, 0); break; } @@ -112,30 +100,25 @@ static int max98357a_platform_probe(struct platform_device *pdev) int ret; max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL); - if (!max98357a) return -ENOMEM; max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(max98357a->sdmode)) return PTR_ERR(max98357a->sdmode); ret = device_property_read_u32(&pdev->dev, "sdmode-delay", &max98357a->sdmode_delay); - if (ret) { max98357a->sdmode_delay = 0; dev_dbg(&pdev->dev, - "no optional property 'sdmode-delay' found, default: no delay\n"); + "no optional property 'sdmode-delay' found, " + "default: no delay\n"); } dev_set_drvdata(&pdev->dev, max98357a); - INIT_DELAYED_WORK(&max98357a->enable_sdmode_work, - max98357a_enable_sdmode_work); - return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, &max98357a_dai_driver, 1); -- 2.22.0.410.gd8fdbe21b5-goog

2 1

[alsa-devel] [PATCH] ASoC: rt1308: Remove executable attribute from source files
by Joe Perches 10 Jul '19

10 Jul '19

These are source files not executable. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/codecs/rt1308.c | 0 sound/soc/codecs/rt1308.h | 0 2 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h old mode 100755 new mode 100644

2 1

[alsa-devel] [PATCH] ASoC: Relocate my e-mail to .com domain zone
by Kirill Marinushkin 10 Jul '19

10 Jul '19

Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.com> --- MAINTAINERS | 2 +- sound/soc/codecs/pcm3060-i2c.c | 4 ++-- sound/soc/codecs/pcm3060-spi.c | 4 ++-- sound/soc/codecs/pcm3060.c | 4 ++-- sound/soc/codecs/pcm3060.h | 2 +- 5 files changed, 8 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 558acf24ea1e..9cdc10e80d78 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -15829,7 +15829,7 @@ S: Maintained F: drivers/net/ethernet/ti/netcp* TI PCM3060 ASoC CODEC DRIVER -M: Kirill Marinushkin <kmarinushkin(a)birdec.tech> +M: Kirill Marinushkin <kmarinushkin(a)birdec.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: Documentation/devicetree/bindings/sound/pcm3060.txt diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c index cdc8314882bc..abcdeb922201 100644 --- a/sound/soc/codecs/pcm3060-i2c.c +++ b/sound/soc/codecs/pcm3060-i2c.c @@ -2,7 +2,7 @@ // // PCM3060 I2C driver // -// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.tech> +// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.com> #include <linux/i2c.h> #include <linux/module.h> @@ -56,5 +56,5 @@ static struct i2c_driver pcm3060_i2c_driver = { module_i2c_driver(pcm3060_i2c_driver); MODULE_DESCRIPTION("PCM3060 I2C driver"); -MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin(a)birdec.tech>"); +MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin(a)birdec.com>"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c index f6f19fa80932..3b79734b832b 100644 --- a/sound/soc/codecs/pcm3060-spi.c +++ b/sound/soc/codecs/pcm3060-spi.c @@ -2,7 +2,7 @@ // // PCM3060 SPI driver // -// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.tech> +// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.com> #include <linux/module.h> #include <linux/spi/spi.h> @@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = { module_spi_driver(pcm3060_spi_driver); MODULE_DESCRIPTION("PCM3060 SPI driver"); -MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin(a)birdec.tech>"); +MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin(a)birdec.com>"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 32b26f1c2282..b2358069cf9b 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -2,7 +2,7 @@ // // PCM3060 codec driver // -// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.tech> +// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.com> #include <linux/module.h> #include <sound/pcm_params.h> @@ -342,5 +342,5 @@ int pcm3060_probe(struct device *dev) EXPORT_SYMBOL(pcm3060_probe); MODULE_DESCRIPTION("PCM3060 codec driver"); -MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin(a)birdec.tech>"); +MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin(a)birdec.com>"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index 75931c9a9d85..18d51e5dac2c 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -2,7 +2,7 @@ /* * PCM3060 codec driver * - * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.tech> + * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin(a)birdec.com> */ #ifndef _SND_SOC_PCM3060_H -- 2.13.6

2 1

[alsa-devel] [PATCH 0/4] Fix some use-after-free problems in sound/soc/generic
by Wen Yang 10 Jul '19

10 Jul '19

We developed a coccinelle SmPL to detect sound/sooc/generic code and found some use-after-free problems. This patch series fixes those problems. Wen Yang (4): ASoC: simple-card: fix an use-after-free in simple_dai_link_of_dpcm() ASoC: simple-card: fix an use-after-free in simple_for_each_link() ASoC: audio-graph-card: fix use-after-free in graph_dai_link_of_dpcm() ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id() sound/soc/generic/audio-graph-card.c | 30 ++++++++++++++++-------------- sound/soc/generic/simple-card.c | 26 +++++++++++++------------- 2 files changed, 29 insertions(+), 27 deletions(-) Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org -- 2.9.5

3 9

[alsa-devel] [PATCH for v5.3] ASoC: simple_card_utils.h: care NULL dai at asoc_simple_debug_dai()
by Kuninori Morimoto 10 Jul '19

10 Jul '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> props->xxx_dai might be NULL when DPCM. This patch cares it for debug. Fixes: commit 0580dde59438 ("ASoC: simple-card-utils: add asoc_simple_debug_info()") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 954563e..985a5f5 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -141,6 +141,10 @@ inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv, { struct device *dev = simple_priv_to_dev(priv); + /* dai might be NULL */ + if (!dai) + return; + if (dai->name) dev_dbg(dev, "%s dai name = %s\n", name, dai->name); -- 2.7.4

2 1

[alsa-devel] [PATCH for v5.3] ASoC: simple-card-utils: care no Platform for DPCM
by Kuninori Morimoto 10 Jul '19

10 Jul '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 34614739988ad ("ASoC: soc-core: support dai_link with platforms_num != 1") supports multi Platform, and commit 9f3eb91753451 ("ASoC: simple-card-utils: consider CPU-Platform possibility") removed no Platform from simple-card. Multi Platform is now checking both Platform name/of_node are NULL case. But in normal case, DPCM be doesn't have Platform. asoc_simple_canonicalize_platform() try to use CPU of_node to Platform (This is needed for DMAEngine platform case), but it still might be NULL at DPCM be. This patch try to use no Platform after that if Platform of_node is still NULL. It can't probe without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index ac8678f..556b1a7 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -349,6 +349,13 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link) /* Assumes platform == cpu */ if (!dai_link->platforms->of_node) dai_link->platforms->of_node = dai_link->cpus->of_node; + + /* + * DPCM BE can be no platform. + * Alloced memory will be waste, but not leak. + */ + if (!dai_link->platforms->of_node) + dai_link->num_platforms = 0; } EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_platform); -- 2.7.4

2 1

[alsa-devel] [PATCH V6] ALSA: usb-audio: Scarlett Gen 2 mixer interface
by Geoffrey D. Bennett 10 Jul '19

10 Jul '19

Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2 audio interfaces. Although the interfaces are USB compliant, additional input/output level controls and hardware routing/mixing functionality are available using proprietary USB requests. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- Hi all, This patch adds the following controls for the Scarlett 6i6, 18i8, and 18i20 Gen 2: - Master volume knob indicator (18i20 only) - Mute and dim switches (18i20 only) - Volume controls for the analogue HW outputs - HW/SW volume switches for the 10 analogue HW outputs (18i20 only) - Line Level/Instrument Level and Pad controls (6i6 and 18i8 only) - Output mux (where the sound for the HW outputs comes from; defaults to PCM outputs) - Capture mux (where the sound for PCM recording comes from; defaults to HW inputs) - Matrix mux (where the sound going into the mixer comes from; 18 inputs default off) - Mixer matrix (18 inputs * 10 outputs = 180 controls) - Level meters Changes since v1: - Add support for the Scarlett 18i8 Gen 2 - Save configuration parameters to NVRAM - Implemented feedback from Takashi's email 24/Apr/2019 - Moved private field from struct snd_usb_audio to struct usb_mixer_interface - Added timer and buffer fields to struct usb_mixer_interface - Other small code fixes/cleanups/improvements Changes since v2: - Add support for the Scarlett 6i6 Gen 2 - Add support for controlling the mute/dim switches (18i20 only) - Implemented feedback from Takashi's email 29/Apr/2019: One pointer field private_data and one function pointer private_free. Replaced timer with delayed work (thanks for that suggestion; it makes the code a *lot* simpler!). - The new functionality is disabled by default as there were reports of some Gen 2 devices not responding to the initialisation sequence and hanging. Added module parameter "scarlett_gen2_mixer_enable". If this parameter is not set, logs a message "Focusrite Scarlett Gen 2 Mixer Driver disabled; use options snd_usb_audio scarlett_gen2_mixer_enable=1 to enable and report any issues to g(a)b4.vu". Changes since v3: - No functional changes; fix checkpatch.pl error and typo in declaration. Changes since v4: - Implemented feedback from Takashi's email 9/Jul/2019: cancel delayed work on suspend, drop copyright texts, remove unnecessary blank lines, and run kmalloc outside the mutex. Changes since v5: - Implemented feedback from Takashi's emails 9-10/Jul/2019: use chip->setup value (device_setup option) instead of the scarlett_gen2_mixer_enable option and sync config to NVRAM immediately on suspend. - Fixed issue where the config save delayed work would be cancelled and not restarted if a scarlett2_*_ctl_put() function was called and the parameter value was not changed. - Renaming definitions and functions for consistency and move private free/suspend functions above their use so pre-declaration isn't necessary. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e1ce257ab705..f4b7384caa9d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,6 +11,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c703f8534b07..734846faee7a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3505,6 +3505,8 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) usb_kill_urb(mixer->urb); if (mixer->rc_urb) usb_kill_urb(mixer->rc_urb); + if (mixer->private_free) + mixer->private_free(mixer); mixer->disconnected = true; } @@ -3532,6 +3534,8 @@ static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) { snd_usb_mixer_inactivate(mixer); + if (mixer->private_suspend) + mixer->private_suspend(mixer); return 0; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 394cd9107507..37e1b234c802 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -28,6 +28,10 @@ struct usb_mixer_interface { struct media_mixer_ctl *media_mixer_ctl; bool disconnected; + + void *private_data; + void (*private_free)(struct usb_mixer_interface *mixer); + void (*private_suspend)(struct usb_mixer_interface *mixer); }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..70805c15b0b5 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -32,6 +32,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -2258,6 +2259,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */ + case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..a772e21bddaa --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,2076 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 6i6/18i8/18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu> + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + */ + +/* Mixer Interface for the Focusrite Scarlett 6i6/18i8/18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * Scarlett 18i8 support added in April 2019. + * + * Scarlett 6i6 support added in June 2019 (thanks to Martin Wittmann + * for providing usbmon output and testing). + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * - line/inst level and pad controls + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ Software gain per channel. + * | Master Gain |<-- 18i20 only: Switch per channel + * +------+------+ to select HW or SW gain control. + * | + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +/* device_setup value to enable */ +#define SCARLETT2_ENABLE 0x01 + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT2_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT2_MIXER_MIN_DB -80 +#define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2) +#define SCARLETT2_MIXER_MAX_DB 6 +#define SCARLETT2_MIXER_MAX_VALUE \ + ((SCARLETT2_MIXER_MAX_DB - SCARLETT2_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_ANALOGUE_MAX 10 + +/* Maximum number of level and pad switches */ +#define SCARLETT2_LEVEL_SWITCH_MAX 2 +#define SCARLETT2_PAD_SWITCH_MAX 4 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +/* Hardware buttons on the 18i20 */ +#define SCARLETT2_BUTTON_MAX 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + u8 line_out_hw_vol; /* line out hw volume is sw controlled */ + u8 button_count; /* number of buttons */ + u8 level_input_count; /* inputs with level selectable */ + u8 pad_input_count; /* inputs with pad selectable */ + const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_mixer_data { + struct usb_mixer_interface *mixer; + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + struct delayed_work work; + const struct scarlett2_device_info *info; + int num_mux_srcs; + u16 scarlett2_seq; + u8 vol_updated; + u8 master_vol; + u8 vol[SCARLETT2_ANALOGUE_MAX]; + u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; + u8 buttons[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + u8 mux[SCARLETT2_MUX_MAX]; + u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s6i6_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first two analogue inputs have an optional pad. */ + .pad_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 8 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 4, 4, 4, 4, 4 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 6, 6, 6, 6, 6 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i8_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first four analogue inputs have an optional pad. */ + .pad_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 4 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 6, 6, 6, 6 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 0, 0, 0, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i20_gen2_info = { + /* The analogue line outputs on the 18i20 can be switched + * between software and hardware volume control + */ + .line_out_hw_vol = 1, + + /* Mute and dim buttons */ + .button_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_CONFIG_SAVE 6 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* volume status is read together (matches scarlett2_config_items[]) */ +struct scarlett2_usb_volume_status { + /* mute & dim buttons */ + u8 buttons[SCARLETT2_BUTTON_MAX]; + + u8 pad1; + + /* software volume setting */ + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + + /* actual volume of output inc. dim (-18dB) */ + s16 hw_vol[SCARLETT2_ANALOGUE_MAX]; + + u8 pad2[SCARLETT2_ANALOGUE_MAX]; + + /* sw (0) or hw (1) controlled */ + u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + + u8 pad3[6]; + + /* front panel volume knob */ + s16 master_vol; +} __packed; + +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, + SCARLETT2_CONFIG_SW_HW_SWITCH = 2, + SCARLETT2_CONFIG_LEVEL_SWITCH = 3, + SCARLETT2_CONFIG_PAD_SWITCH = 4, + SCARLETT2_CONFIG_COUNT = 5 +}; + +/* Location, size, and activation command number for the configuration + * parameters + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +static const struct scarlett2_config + scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { + /* Mute/Dim Buttons */ + { + .offset = 0x31, + .size = 1, + .activate = 2 + }, + + /* Line Out Volume */ + { + .offset = 0x34, + .size = 2, + .activate = 1 + }, + + /* SW/HW Volume Switch */ + { + .offset = 0x66, + .size = 1, + .activate = 3 + }, + + /* Level Switch */ + { + .offset = 0x7c, + .size = 1, + .activate = 7 + }, + + /* Pad Switch */ + { + .offset = 0x84, + .size = 1, + .activate = 8 + } +}; + +/* proprietary request/response format */ +struct scarlett2_usb_packet { + u32 cmd; + u16 size; + u16 seq; + u32 error; + u32 pad; + u8 data[]; +}; + +#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) + +static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, + struct scarlett2_usb_packet *req, + u32 cmd, u16 req_size) +{ + /* sequence must go up by 1 for each request */ + u16 seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; +} + +/* Send a proprietary format request to the Scarlett interface */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, u32 cmd, + void *req_data, u16 req_size, void *resp_data, u16 resp_size) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; + u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; + struct scarlett2_usb_packet *req = 0, *resp = 0; + int err = 0; + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + err = -ENOMEM; + goto error; + } + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + err = -ENOMEM; + goto error; + } + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + scarlett2_fill_request_header(private, req, cmd, req_size); + + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + if (err != req_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB request result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + /* send a second message to get the response */ + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB response result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + if (resp->cmd != req->cmd || + resp->seq != req->seq || + resp_size != le16_to_cpu(resp->size) || + resp->error || + resp->pad) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd), + le16_to_cpu(req->seq), le16_to_cpu(resp->seq), + resp_size, le16_to_cpu(resp->size), + le16_to_cpu(resp->error), + le16_to_cpu(resp->pad)); + err = -EINVAL; + goto unlock; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + +unlock: + mutex_unlock(&private->usb_mutex); +error: + kfree(req); + kfree(resp); + return err; +} + +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct usb_mixer_interface *mixer) +{ + u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); + + scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(u32), + NULL, 0); +} + +/* Delayed work to save config */ +static void scarlett2_config_save_work(struct work_struct *work) +{ + struct scarlett2_mixer_data *private = + container_of(work, struct scarlett2_mixer_data, work.work); + + scarlett2_config_save(private->mixer); +} + +/* Send a USB message to set a configuration parameter (volume level, + * sw/hw volume switch, line/inst level switch, or pad switch) + */ +static int scarlett2_usb_set_config( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int value) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + struct { + u32 offset; + u32 bytes; + s32 value; + } __packed req; + u32 req2; + int err; + struct scarlett2_mixer_data *private = mixer->private_data; + + /* Cancel any pending NVRAM save */ + cancel_delayed_work_sync(&private->work); + + /* Send the configuration parameter data */ + req.offset = cpu_to_le32(config_item.offset + index * config_item.size); + req.bytes = cpu_to_le32(config_item.size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + config_item.size, + NULL, 0); + if (err < 0) + return err; + + /* Activate the change */ + req2 = cpu_to_le32(config_item.activate); + err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + /* Schedule the change to be written to NVRAM */ + schedule_delayed_work(&private->work, msecs_to_jiffies(2000)); + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + int offset, void *buf, int size) +{ + struct { + u32 offset; + u32 size; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA, + &req, sizeof(req), buf, size); +} + +/* Send a USB message to get configuration parameters; result placed in *buf */ +static int scarlett2_usb_get_config( + struct usb_mixer_interface *mixer, + int config_item_num, int count, void *buf) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + int size = config_item.size * count; + + return scarlett2_usb_get(mixer, config_item.offset, buf, size); +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + struct { + u16 mix_num; + u16 data[SCARLETT2_INPUT_MIX_MAX]; + } __packed req; + + int i, j; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(u16), + NULL, 0); +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct { + u16 pad; + u16 num; + u32 data[SCARLETT2_MUX_MAX]; + } __packed req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + u16 *levels) +{ + struct { + u16 pad; + u16 num_meters; + u32 magic; + } __packed req; + u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* Update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < private->info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol[index] = val; + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, val - SCARLETT2_VOLUME_BIAS); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett2_gain, -SCARLETT2_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, private->master_vol - SCARLETT2_VOLUME_BIAS); + if (err < 0) + goto unlock; + + /* Notify of RO/RW change */ + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Line Level/Instrument Level Switch Controls ***/ + +static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Line", "Inst" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->level_switch[elem->control]; + return 0; +} + +static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->level_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->level_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_level_enum_ctl_info, + .get = scarlett2_level_enum_ctl_get, + .put = scarlett2_level_enum_ctl_put, +}; + +/*** Pad Switch Controls ***/ + +static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->pad_switch[elem->control]; + return 0; +} + +static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->pad_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pad_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_pad_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_pad_ctl_get, + .put = scarlett2_pad_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->buttons[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->buttons[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get, + .put = scarlett2_button_ctl_put +}; + +/*** Create the analogue output controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), "Master HW Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_master_volume_ctl, + 0, 1, s, &private->master_vol_ctl); + if (err < 0) + return err; + } + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(s, sizeof(s), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(s, sizeof(s), + "Line %02d Playback Volume", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_line_out_volume_ctl, + i, 1, s, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_sw_hw_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + /* Add HW button controls */ + for (i = 0; i < private->info->button_count; i++) { + err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Create the analogue input controls ***/ + +static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add input level (line/inst) controls */ + for (i = 0; i < info->level_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + /* Add input pad controls */ + for (i = 0; i < info->pad_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) + goto unlock; + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(mixer, mix_num); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett2_mixer, + SCARLETT2_MIXER_MIN_DB * 100, + SCARLETT2_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT2_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett2_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(s, sizeof(s), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) + goto unlock; + + private->mux[index] = val; + err = scarlett2_usb_set_mux(mixer); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(s, sizeof(s) - 5, descr, channel + 1); + strcat(s, " Enum"); + + err = scarlett2_add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, + "Level Meter", NULL); +} + +/*** Cleanup/Suspend Callbacks ***/ + +static void scarlett2_private_free(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + cancel_delayed_work_sync(&private->work); + kfree(private); + mixer->private_data = NULL; +} + +static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + if (cancel_delayed_work_sync(&private->work)) + scarlett2_config_save(private->mixer); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_mixer_data *private = + kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + private->mixer = mixer; + mixer->private_data = private; + mixer->private_free = scarlett2_private_free; + mixer->private_suspend = scarlett2_private_suspend; + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, + NULL, 0, NULL, 0); +} + +/* Read line-in config and line-out volume settings on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + if (info->level_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count, + level_switches); + if (err < 0) + return err; + for (i = 0; i < info->level_input_count; i++) + private->level_switch[i] = level_switches[i]; + } + + if (info->pad_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, + pad_switches); + if (err < 0) + return err; + for (i = 0; i < info->pad_input_count; i++) + private->pad_switch[i] = pad_switches[i]; + } + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = + info->line_out_hw_vol + && volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + int i; + + private->vol_updated = 1; + + for (i = 0; i < private->info->button_count; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + usb_audio_err(mixer->chip, + "scarlett mixer interrupt length %d\n", len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + void *transfer_buffer; + + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb( + mixer->urb, + mixer->chip->dev, + usb_rcvintpipe(mixer->chip->dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT), + transfer_buffer, + SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, + mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8203): + info = &s6i6_gen2_info; + break; + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + if (!(mixer->chip->setup & SCARLETT2_ENABLE)) { + usb_audio_err(mixer->chip, + "Focusrite Scarlett Gen 2 Mixer Driver disabled; " + "use options snd_usb_audio device_setup=1 " + "to enable and report any issues to g(a)b4.vu"); + return 0; + } + + /* Initialise private data, routing, sequence number */ + err = scarlett2_init_private(mixer, info); + if (err < 0) + return err; + + /* Read volume levels and controls from the interface */ + err = scarlett2_read_configs(mixer); + if (err < 0) + return err; + + /* Create the analogue output controls */ + err = scarlett2_add_line_out_ctls(mixer); + if (err < 0) + return err; + + /* Create the analogue input controls */ + err = scarlett2_add_line_in_ctls(mixer); + if (err < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + err = scarlett2_add_mux_enums(mixer); + if (err < 0) + return err; + + /* Create the matrix mixer controls */ + err = scarlett2_add_mixer_ctls(mixer); + if (err < 0) + return err; + + /* Create the level meter controls */ + err = scarlett2_add_meter_ctl(mixer); + if (err < 0) + return err; + + /* Set up the interrupt polling if there are hardware buttons */ + if (info->button_count) { + err = scarlett2_mixer_status_create(mixer); + if (err < 0) + return err; + } + + return 0; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */

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[alsa-devel] [PATCH V4] ALSA: usb-audio: Scarlett Gen 2 mixer interface
by Geoffrey D. Bennett 10 Jul '19

10 Jul '19

Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2 audio interfaces. Although the interfaces are USB compliant, additional input/output level controls and hardware routing/mixing functionality are available using proprietary USB requests. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- Hi all, This patch adds the following controls for the Scarlett 6i6, 18i8, and 18i20 Gen 2: - Master volume knob indicator (18i20 only) - Mute and dim switches (18i20 only) - Volume controls for the analogue HW outputs - HW/SW volume switches for the 10 analogue HW outputs (18i20 only) - Line Level/Instrument Level and Pad controls (6i6 and 18i8 only) - Output mux (where the sound for the HW outputs comes from; defaults to PCM outputs) - Capture mux (where the sound for PCM recording comes from; defaults to HW inputs) - Matrix mux (where the sound going into the mixer comes from; 18 inputs default off) - Mixer matrix (18 inputs * 10 outputs = 180 controls) - Level meters Changes since v1: - Add support for the Scarlett 18i8 Gen 2 - Save configuration parameters to NVRAM - Implemented feedback from Takashi's email 24/Apr/2019 - Moved private field from struct snd_usb_audio to struct usb_mixer_interface - Added timer and buffer fields to struct usb_mixer_interface - Other small code fixes/cleanups/improvements Changes since v2: - Add support for the Scarlett 6i6 Gen 2 - Add support for controlling the mute/dim switches (18i20 only) - Implemented feedback from Takashi's email 29/Apr/2019: One pointer field private_data and one function pointer private_free. Replaced timer with delayed work (thanks for that suggestion; it makes the code a *lot* simpler!). - The new functionality is disabled by default as there were reports of some Gen 2 devices not responding to the initialisation sequence and hanging. Added module parameter "scarlett_gen2_mixer_enable". If this parameter is not set, logs a message "Focusrite Scarlett Gen 2 Mixer Driver disabled; use options snd_usb_audio scarlett_gen2_mixer_enable=1 to enable and report any issues to g(a)b4.vu". Changes since v3: - No functional changes; fix checkpatch.pl error and typo in declaration. Only outstanding question I believe: - Takashi, you wrote last time "Also, the delayed work needs to be canceled or flushed at PM suspend callback as well as disconnecting." I'm not sure why it should be cancelled; is it not okay to do on resume? But I tested suspending while the delayed work was waiting and it seemed like the delayed work was cancelled anyway. Probably better to do a flush on suspend though; should this be done through a hook like: diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2444737a14b2..1572a89267c6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3547,6 +3547,8 @@ static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) { snd_usb_mixer_inactivate(mixer); + if (mixer->private_suspend) + mixer->private_suspend(mixer); return 0; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index fa6e216a06f0..d94c688c65f7 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -28,6 +28,7 @@ struct usb_mixer_interface { void *private_data; void (*private_free)(struct usb_mixer_interface *private_data); + void (*private_suspend)(struct usb_mixer_interface *); }; #define MAX_CHANNELS 16 /* max logical channels */ ...or just keep the existing behaviour where it seems to get cancelled? Thanks, Geoffrey. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e1ce257ab705..f4b7384caa9d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,6 +11,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c703f8534b07..079ebb85805f 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3505,6 +3505,8 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) usb_kill_urb(mixer->urb); if (mixer->rc_urb) usb_kill_urb(mixer->rc_urb); + if (mixer->private_free) + mixer->private_free(mixer); mixer->disconnected = true; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 394cd9107507..41e8be671b73 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -28,6 +28,9 @@ struct usb_mixer_interface { struct media_mixer_ctl *media_mixer_ctl; bool disconnected; + + void *private_data; + void (*private_free)(struct usb_mixer_interface *mixer); }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..70805c15b0b5 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -32,6 +32,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -2258,6 +2259,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */ + case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..02bab6ae59e8 --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,2078 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 6i6/18i8/18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu> + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +/* Mixer Interface for the Focusrite Scarlett 6i6/18i8/18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * Scarlett 18i8 support added in April 2019. + * + * Scarlett 6i6 support added in June 2019 (thanks to Martin Wittmann + * for providing usbmon output and testing). + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * - line/inst level and pad controls + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ Software gain per channel. + * | Master Gain |<-- 18i20 only: Switch per channel + * +------+------+ to select HW or SW gain control. + * | + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +static int scarlett_gen2_mixer_enable; +module_param(scarlett_gen2_mixer_enable, int, 0444); +MODULE_PARM_DESC(scarlett_gen2_mixer_enable, + "Focusrite Scarlett Gen 2 Mixer Driver Enable"); + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT_MIXER_MIN_DB -80 +#define SCARLETT_MIXER_BIAS (-SCARLETT_MIXER_MIN_DB * 2) +#define SCARLETT_MIXER_MAX_DB 6 +#define SCARLETT_MIXER_MAX_VALUE \ + ((SCARLETT_MIXER_MAX_DB - SCARLETT_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_ANALOGUE_MAX 10 + +/* Maximum number of level and pad switches */ +#define SCARLETT2_LEVEL_SWITCH_MAX 2 +#define SCARLETT2_PAD_SWITCH_MAX 4 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +/* Hardware buttons on the 18i20 */ +#define SCARLETT2_BUTTON_MAX 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + u8 line_out_hw_vol; /* line out hw volume is sw controlled */ + u8 button_count; /* number of buttons */ + u8 level_input_count; /* inputs with level selectable */ + u8 pad_input_count; /* inputs with pad selectable */ + const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_mixer_data { + struct usb_mixer_interface *mixer; + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + struct delayed_work work; + const struct scarlett2_device_info *info; + int num_mux_srcs; + u16 scarlett2_seq; + u8 vol_updated; + u8 master_vol; + u8 vol[SCARLETT2_ANALOGUE_MAX]; + u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; + u8 buttons[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + u8 mux[SCARLETT2_MUX_MAX]; + u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +static void snd_scarlett_gen2_private_free(struct usb_mixer_interface *mixer); + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s6i6_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first two analogue inputs have an optional pad. */ + .pad_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 8 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 4, 4, 4, 4, 4 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 6, 6, 6, 6, 6 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i8_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first four analogue inputs have an optional pad. */ + .pad_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 4 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 6, 6, 6, 6 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 0, 0, 0, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i20_gen2_info = { + /* The analogue line outputs on the 18i20 can be switched + * between software and hardware volume control + */ + .line_out_hw_vol = 1, + + /* Mute and dim buttons */ + .button_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_CONFIG_SAVE 6 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* volume status is read together (matches scarlett2_config_items[]) */ +struct scarlett2_usb_volume_status { + /* mute & dim buttons */ + u8 buttons[SCARLETT2_BUTTON_MAX]; + + u8 pad1; + + /* software volume setting */ + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + + /* actual volume of output inc. dim (-18dB) */ + s16 hw_vol[SCARLETT2_ANALOGUE_MAX]; + + u8 pad2[SCARLETT2_ANALOGUE_MAX]; + + /* sw (0) or hw (1) controlled */ + u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + + u8 pad3[6]; + + /* front panel volume knob */ + s16 master_vol; +} __packed; + +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, + SCARLETT2_CONFIG_SW_HW_SWITCH = 2, + SCARLETT2_CONFIG_LEVEL_SWITCH = 3, + SCARLETT2_CONFIG_PAD_SWITCH = 4, + SCARLETT2_CONFIG_COUNT = 5 +}; + +/* Location, size, and activation command number for the configuration + * parameters + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +static const struct scarlett2_config + scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { + /* Mute/Dim Buttons */ + { + .offset = 0x31, + .size = 1, + .activate = 2 + }, + + /* Line Out Volume */ + { + .offset = 0x34, + .size = 2, + .activate = 1 + }, + + /* SW/HW Volume Switch */ + { + .offset = 0x66, + .size = 1, + .activate = 3 + }, + + /* Level Switch */ + { + .offset = 0x7c, + .size = 1, + .activate = 7 + }, + + /* Pad Switch */ + { + .offset = 0x84, + .size = 1, + .activate = 8 + } +}; + +/* proprietary request/response format */ +struct scarlett2_usb_packet { + u32 cmd; + u16 size; + u16 seq; + u32 error; + u32 pad; + u8 data[]; +}; + +#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) + +static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, + struct scarlett2_usb_packet *req, + u32 cmd, u16 req_size) +{ + /* sequence must go up by 1 for each request */ + u16 seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; +} + +/* Send a proprietary format request to the Scarlett interface */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, u32 cmd, + void *req_data, u16 req_size, void *resp_data, u16 resp_size) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; + u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; + + struct scarlett2_usb_packet *req = 0, *resp = 0; + + int err = 0; + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + err = -ENOMEM; + goto unlock; + } + + scarlett2_fill_request_header(private, req, cmd, req_size); + + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + if (err != req_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB request result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + /* send a second message to get the response */ + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + err = -ENOMEM; + goto unlock; + } + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB response result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + if (resp->cmd != req->cmd || + resp->seq != req->seq || + resp_size != le16_to_cpu(resp->size) || + resp->error || + resp->pad) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd), + le16_to_cpu(req->seq), le16_to_cpu(resp->seq), + resp_size, le16_to_cpu(resp->size), + le16_to_cpu(resp->error), + le16_to_cpu(resp->pad)); + err = -EINVAL; + goto unlock; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + +unlock: + kfree(req); + kfree(resp); + mutex_unlock(&private->usb_mutex); + return err; +} + +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct work_struct *work) +{ + struct scarlett2_mixer_data *private = + container_of(work, struct scarlett2_mixer_data, work.work); + struct usb_mixer_interface *mixer = private->mixer; + u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); + + scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(u32), + NULL, 0); +} + +/* Send a USB message to set a configuration parameter (volume level, + * sw/hw volume switch, line/inst level switch, or pad switch) + */ +static int scarlett2_usb_set_config( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int value) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + struct { + u32 offset; + u32 bytes; + s32 value; + } __packed req; + u32 req2; + int err; + struct scarlett2_mixer_data *private = mixer->private_data; + + /* Send the configuration parameter data */ + req.offset = cpu_to_le32(config_item.offset + index * config_item.size); + req.bytes = cpu_to_le32(config_item.size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + config_item.size, + NULL, 0); + if (err < 0) + return err; + + /* Activate the change */ + req2 = cpu_to_le32(config_item.activate); + err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + /* Schedule the change to be written to NVRAM */ + schedule_delayed_work(&private->work, msecs_to_jiffies(2000)); + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + int offset, void *buf, int size) +{ + struct { + u32 offset; + u32 size; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA, + &req, sizeof(req), buf, size); +} + +/* Send a USB message to get configuration parameters; result placed in *buf */ +static int scarlett2_usb_get_config( + struct usb_mixer_interface *mixer, + int config_item_num, int count, void *buf) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + int size = config_item.size * count; + + return scarlett2_usb_get(mixer, config_item.offset, buf, size); +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + struct { + u16 mix_num; + u16 data[SCARLETT2_INPUT_MIX_MAX]; + } __packed req; + + int i, j; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(u16), + NULL, 0); +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct { + u16 pad; + u16 num; + u32 data[SCARLETT2_MUX_MAX]; + } __packed req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + u16 *levels) +{ + struct { + u16 pad; + u16 num_meters; + u32 magic; + } __packed req; + u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* Update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < private->info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol[index] = val; + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, val - SCARLETT_VOLUME_BIAS); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_gain, -SCARLETT_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, private->master_vol - SCARLETT_VOLUME_BIAS); + if (err < 0) + goto unlock; + + /* Notify of RO/RW change */ + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Line Level/Instrument Level Switch Controls ***/ + +static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Line", "Inst" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->level_switch[elem->control]; + return 0; +} + +static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->level_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->level_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_level_enum_ctl_info, + .get = scarlett2_level_enum_ctl_get, + .put = scarlett2_level_enum_ctl_put, +}; + +/*** Pad Switch Controls ***/ + +static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->pad_switch[elem->control]; + return 0; +} + +static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->pad_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pad_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_pad_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_pad_ctl_get, + .put = scarlett2_pad_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->buttons[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->buttons[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get, + .put = scarlett2_button_ctl_put +}; + +/*** Create the analogue output controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), "Master HW Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_master_volume_ctl, + 0, 1, s, &private->master_vol_ctl); + if (err < 0) + return err; + } + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(s, sizeof(s), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(s, sizeof(s), + "Line %02d Playback Volume", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_line_out_volume_ctl, + i, 1, s, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_sw_hw_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + /* Add HW button controls */ + for (i = 0; i < private->info->button_count; i++) { + err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Create the analogue input controls ***/ + +static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add input level (line/inst) controls */ + for (i = 0; i < info->level_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + /* Add input pad controls */ + for (i = 0; i < info->pad_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) + goto unlock; + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(mixer, mix_num); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_mixer, + SCARLETT_MIXER_MIN_DB * 100, + SCARLETT_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(s, sizeof(s), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) + goto unlock; + + private->mux[index] = val; + err = scarlett2_usb_set_mux(mixer); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(s, sizeof(s) - 5, descr, channel + 1); + strcat(s, " Enum"); + + err = scarlett2_add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, + "Level Meter", NULL); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_mixer_data *private = + kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + INIT_DELAYED_WORK(&private->work, scarlett2_config_save); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + private->mixer = mixer; + mixer->private_data = private; + mixer->private_free = snd_scarlett_gen2_private_free; + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, + NULL, 0, NULL, 0); +} + +/* Read line-in config and line-out volume settings on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + if (info->level_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count, + level_switches); + if (err < 0) + return err; + for (i = 0; i < info->level_input_count; i++) + private->level_switch[i] = level_switches[i]; + } + + if (info->pad_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, + pad_switches); + if (err < 0) + return err; + for (i = 0; i < info->pad_input_count; i++) + private->pad_switch[i] = pad_switches[i]; + } + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = + info->line_out_hw_vol + && volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + int i; + + private->vol_updated = 1; + + for (i = 0; i < private->info->button_count; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + usb_audio_err(mixer->chip, + "scarlett mixer interrupt length %d\n", len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + void *transfer_buffer; + + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb( + mixer->urb, + mixer->chip->dev, + usb_rcvintpipe(mixer->chip->dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT), + transfer_buffer, + SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, + mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8203): + info = &s6i6_gen2_info; + break; + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + if (!scarlett_gen2_mixer_enable) { + usb_audio_err(mixer->chip, + "Focusrite Scarlett Gen 2 Mixer Driver disabled; use " + "options snd_usb_audio scarlett_gen2_mixer_enable=1 " + "to enable and report any issues to g(a)b4.vu"); + return 0; + } + + /* Initialise private data, routing, sequence number */ + err = scarlett2_init_private(mixer, info); + if (err < 0) + return err; + + /* Read volume levels and controls from the interface */ + err = scarlett2_read_configs(mixer); + if (err < 0) + return err; + + /* Create the analogue output controls */ + err = scarlett2_add_line_out_ctls(mixer); + if (err < 0) + return err; + + /* Create the analogue input controls */ + err = scarlett2_add_line_in_ctls(mixer); + if (err < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + err = scarlett2_add_mux_enums(mixer); + if (err < 0) + return err; + + /* Create the matrix mixer controls */ + err = scarlett2_add_mixer_ctls(mixer); + if (err < 0) + return err; + + /* Create the level meter controls */ + err = scarlett2_add_meter_ctl(mixer); + if (err < 0) + return err; + + /* Set up the interrupt polling if there are hardware buttons */ + if (info->button_count) { + err = scarlett2_mixer_status_create(mixer); + if (err < 0) + return err; + } + + return 0; +} + +static void snd_scarlett_gen2_private_free(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + cancel_delayed_work_sync(&private->work); + kfree(private); + mixer->private_data = NULL; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */

2 4

[alsa-devel] [PATCH V5] ALSA: usb-audio: Scarlett Gen 2 mixer interface
by Geoffrey D. Bennett 10 Jul '19

10 Jul '19

Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2 audio interfaces. Although the interfaces are USB compliant, additional input/output level controls and hardware routing/mixing functionality are available using proprietary USB requests. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- Hi all, This patch adds the following controls for the Scarlett 6i6, 18i8, and 18i20 Gen 2: - Master volume knob indicator (18i20 only) - Mute and dim switches (18i20 only) - Volume controls for the analogue HW outputs - HW/SW volume switches for the 10 analogue HW outputs (18i20 only) - Line Level/Instrument Level and Pad controls (6i6 and 18i8 only) - Output mux (where the sound for the HW outputs comes from; defaults to PCM outputs) - Capture mux (where the sound for PCM recording comes from; defaults to HW inputs) - Matrix mux (where the sound going into the mixer comes from; 18 inputs default off) - Mixer matrix (18 inputs * 10 outputs = 180 controls) - Level meters Changes since v1: - Add support for the Scarlett 18i8 Gen 2 - Save configuration parameters to NVRAM - Implemented feedback from Takashi's email 24/Apr/2019 - Moved private field from struct snd_usb_audio to struct usb_mixer_interface - Added timer and buffer fields to struct usb_mixer_interface - Other small code fixes/cleanups/improvements Changes since v2: - Add support for the Scarlett 6i6 Gen 2 - Add support for controlling the mute/dim switches (18i20 only) - Implemented feedback from Takashi's email 29/Apr/2019: One pointer field private_data and one function pointer private_free. Replaced timer with delayed work (thanks for that suggestion; it makes the code a *lot* simpler!). - The new functionality is disabled by default as there were reports of some Gen 2 devices not responding to the initialisation sequence and hanging. Added module parameter "scarlett_gen2_mixer_enable". If this parameter is not set, logs a message "Focusrite Scarlett Gen 2 Mixer Driver disabled; use options snd_usb_audio scarlett_gen2_mixer_enable=1 to enable and report any issues to g(a)b4.vu". Changes since v3: - No functional changes; fix checkpatch.pl error and typo in declaration. Changes since v4: - Implemented feedback from Takashi's email 9/Jul/2019: cancel delayed work on suspend, drop copyright texts, remove unnecessary blank lines, and run kmalloc outside the mutex. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index e1ce257ab705..f4b7384caa9d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,6 +11,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c703f8534b07..734846faee7a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3505,6 +3505,8 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) usb_kill_urb(mixer->urb); if (mixer->rc_urb) usb_kill_urb(mixer->rc_urb); + if (mixer->private_free) + mixer->private_free(mixer); mixer->disconnected = true; } @@ -3532,6 +3534,8 @@ static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) { snd_usb_mixer_inactivate(mixer); + if (mixer->private_suspend) + mixer->private_suspend(mixer); return 0; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 394cd9107507..37e1b234c802 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -28,6 +28,10 @@ struct usb_mixer_interface { struct media_mixer_ctl *media_mixer_ctl; bool disconnected; + + void *private_data; + void (*private_free)(struct usb_mixer_interface *mixer); + void (*private_suspend)(struct usb_mixer_interface *mixer); }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..70805c15b0b5 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -32,6 +32,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -2258,6 +2259,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */ + case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..1f055cb89e91 --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,2075 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 6i6/18i8/18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu> + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + */ + +/* Mixer Interface for the Focusrite Scarlett 6i6/18i8/18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * Scarlett 18i8 support added in April 2019. + * + * Scarlett 6i6 support added in June 2019 (thanks to Martin Wittmann + * for providing usbmon output and testing). + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * - line/inst level and pad controls + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ Software gain per channel. + * | Master Gain |<-- 18i20 only: Switch per channel + * +------+------+ to select HW or SW gain control. + * | + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +static int scarlett_gen2_mixer_enable; +module_param(scarlett_gen2_mixer_enable, int, 0444); +MODULE_PARM_DESC(scarlett_gen2_mixer_enable, + "Focusrite Scarlett Gen 2 Mixer Driver Enable"); + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT_MIXER_MIN_DB -80 +#define SCARLETT_MIXER_BIAS (-SCARLETT_MIXER_MIN_DB * 2) +#define SCARLETT_MIXER_MAX_DB 6 +#define SCARLETT_MIXER_MAX_VALUE \ + ((SCARLETT_MIXER_MAX_DB - SCARLETT_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_ANALOGUE_MAX 10 + +/* Maximum number of level and pad switches */ +#define SCARLETT2_LEVEL_SWITCH_MAX 2 +#define SCARLETT2_PAD_SWITCH_MAX 4 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +/* Hardware buttons on the 18i20 */ +#define SCARLETT2_BUTTON_MAX 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + u8 line_out_hw_vol; /* line out hw volume is sw controlled */ + u8 button_count; /* number of buttons */ + u8 level_input_count; /* inputs with level selectable */ + u8 pad_input_count; /* inputs with pad selectable */ + const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_mixer_data { + struct usb_mixer_interface *mixer; + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + struct delayed_work work; + const struct scarlett2_device_info *info; + int num_mux_srcs; + u16 scarlett2_seq; + u8 vol_updated; + u8 master_vol; + u8 vol[SCARLETT2_ANALOGUE_MAX]; + u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; + u8 buttons[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + u8 mux[SCARLETT2_MUX_MAX]; + u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +static void snd_scarlett_gen2_private_free(struct usb_mixer_interface *mixer); +static void snd_scarlett_gen2_private_suspend( + struct usb_mixer_interface *mixer); + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s6i6_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first two analogue inputs have an optional pad. */ + .pad_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 8 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 4, 4, 4, 4, 4 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 6, 6, 6, 6, 6 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i8_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first four analogue inputs have an optional pad. */ + .pad_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 4 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 6, 6, 6, 6 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 0, 0, 0, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i20_gen2_info = { + /* The analogue line outputs on the 18i20 can be switched + * between software and hardware volume control + */ + .line_out_hw_vol = 1, + + /* Mute and dim buttons */ + .button_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_CONFIG_SAVE 6 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* volume status is read together (matches scarlett2_config_items[]) */ +struct scarlett2_usb_volume_status { + /* mute & dim buttons */ + u8 buttons[SCARLETT2_BUTTON_MAX]; + + u8 pad1; + + /* software volume setting */ + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + + /* actual volume of output inc. dim (-18dB) */ + s16 hw_vol[SCARLETT2_ANALOGUE_MAX]; + + u8 pad2[SCARLETT2_ANALOGUE_MAX]; + + /* sw (0) or hw (1) controlled */ + u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + + u8 pad3[6]; + + /* front panel volume knob */ + s16 master_vol; +} __packed; + +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, + SCARLETT2_CONFIG_SW_HW_SWITCH = 2, + SCARLETT2_CONFIG_LEVEL_SWITCH = 3, + SCARLETT2_CONFIG_PAD_SWITCH = 4, + SCARLETT2_CONFIG_COUNT = 5 +}; + +/* Location, size, and activation command number for the configuration + * parameters + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +static const struct scarlett2_config + scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { + /* Mute/Dim Buttons */ + { + .offset = 0x31, + .size = 1, + .activate = 2 + }, + + /* Line Out Volume */ + { + .offset = 0x34, + .size = 2, + .activate = 1 + }, + + /* SW/HW Volume Switch */ + { + .offset = 0x66, + .size = 1, + .activate = 3 + }, + + /* Level Switch */ + { + .offset = 0x7c, + .size = 1, + .activate = 7 + }, + + /* Pad Switch */ + { + .offset = 0x84, + .size = 1, + .activate = 8 + } +}; + +/* proprietary request/response format */ +struct scarlett2_usb_packet { + u32 cmd; + u16 size; + u16 seq; + u32 error; + u32 pad; + u8 data[]; +}; + +#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) + +static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, + struct scarlett2_usb_packet *req, + u32 cmd, u16 req_size) +{ + /* sequence must go up by 1 for each request */ + u16 seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; +} + +/* Send a proprietary format request to the Scarlett interface */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, u32 cmd, + void *req_data, u16 req_size, void *resp_data, u16 resp_size) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; + u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; + struct scarlett2_usb_packet *req = 0, *resp = 0; + int err = 0; + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + err = -ENOMEM; + goto error; + } + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + err = -ENOMEM; + goto error; + } + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + scarlett2_fill_request_header(private, req, cmd, req_size); + + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + if (err != req_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB request result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + /* send a second message to get the response */ + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB response result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + if (resp->cmd != req->cmd || + resp->seq != req->seq || + resp_size != le16_to_cpu(resp->size) || + resp->error || + resp->pad) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd), + le16_to_cpu(req->seq), le16_to_cpu(resp->seq), + resp_size, le16_to_cpu(resp->size), + le16_to_cpu(resp->error), + le16_to_cpu(resp->pad)); + err = -EINVAL; + goto unlock; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + +unlock: + mutex_unlock(&private->usb_mutex); +error: + kfree(req); + kfree(resp); + return err; +} + +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct work_struct *work) +{ + struct scarlett2_mixer_data *private = + container_of(work, struct scarlett2_mixer_data, work.work); + struct usb_mixer_interface *mixer = private->mixer; + u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); + + scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(u32), + NULL, 0); +} + +/* Send a USB message to set a configuration parameter (volume level, + * sw/hw volume switch, line/inst level switch, or pad switch) + */ +static int scarlett2_usb_set_config( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int value) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + struct { + u32 offset; + u32 bytes; + s32 value; + } __packed req; + u32 req2; + int err; + struct scarlett2_mixer_data *private = mixer->private_data; + + /* Send the configuration parameter data */ + req.offset = cpu_to_le32(config_item.offset + index * config_item.size); + req.bytes = cpu_to_le32(config_item.size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + config_item.size, + NULL, 0); + if (err < 0) + return err; + + /* Activate the change */ + req2 = cpu_to_le32(config_item.activate); + err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + /* Schedule the change to be written to NVRAM */ + schedule_delayed_work(&private->work, msecs_to_jiffies(2000)); + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + int offset, void *buf, int size) +{ + struct { + u32 offset; + u32 size; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA, + &req, sizeof(req), buf, size); +} + +/* Send a USB message to get configuration parameters; result placed in *buf */ +static int scarlett2_usb_get_config( + struct usb_mixer_interface *mixer, + int config_item_num, int count, void *buf) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + int size = config_item.size * count; + + return scarlett2_usb_get(mixer, config_item.offset, buf, size); +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + struct { + u16 mix_num; + u16 data[SCARLETT2_INPUT_MIX_MAX]; + } __packed req; + + int i, j; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(u16), + NULL, 0); +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct { + u16 pad; + u16 num; + u32 data[SCARLETT2_MUX_MAX]; + } __packed req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + u16 *levels) +{ + struct { + u16 pad; + u16 num_meters; + u32 magic; + } __packed req; + u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* Update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < private->info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol[index] = val; + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, val - SCARLETT_VOLUME_BIAS); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_gain, -SCARLETT_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, private->master_vol - SCARLETT_VOLUME_BIAS); + if (err < 0) + goto unlock; + + /* Notify of RO/RW change */ + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Line Level/Instrument Level Switch Controls ***/ + +static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Line", "Inst" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->level_switch[elem->control]; + return 0; +} + +static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->level_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->level_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_level_enum_ctl_info, + .get = scarlett2_level_enum_ctl_get, + .put = scarlett2_level_enum_ctl_put, +}; + +/*** Pad Switch Controls ***/ + +static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->pad_switch[elem->control]; + return 0; +} + +static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->pad_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pad_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_pad_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_pad_ctl_get, + .put = scarlett2_pad_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->buttons[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->buttons[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get, + .put = scarlett2_button_ctl_put +}; + +/*** Create the analogue output controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), "Master HW Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_master_volume_ctl, + 0, 1, s, &private->master_vol_ctl); + if (err < 0) + return err; + } + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(s, sizeof(s), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(s, sizeof(s), + "Line %02d Playback Volume", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_line_out_volume_ctl, + i, 1, s, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_sw_hw_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + /* Add HW button controls */ + for (i = 0; i < private->info->button_count; i++) { + err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Create the analogue input controls ***/ + +static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add input level (line/inst) controls */ + for (i = 0; i < info->level_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + /* Add input pad controls */ + for (i = 0; i < info->pad_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) + goto unlock; + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(mixer, mix_num); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_mixer, + SCARLETT_MIXER_MIN_DB * 100, + SCARLETT_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(s, sizeof(s), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) + goto unlock; + + private->mux[index] = val; + err = scarlett2_usb_set_mux(mixer); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(s, sizeof(s) - 5, descr, channel + 1); + strcat(s, " Enum"); + + err = scarlett2_add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, + "Level Meter", NULL); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_mixer_data *private = + kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + INIT_DELAYED_WORK(&private->work, scarlett2_config_save); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + private->mixer = mixer; + mixer->private_data = private; + mixer->private_free = snd_scarlett_gen2_private_free; + mixer->private_suspend = snd_scarlett_gen2_private_suspend; + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, + NULL, 0, NULL, 0); +} + +/* Read line-in config and line-out volume settings on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + if (info->level_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count, + level_switches); + if (err < 0) + return err; + for (i = 0; i < info->level_input_count; i++) + private->level_switch[i] = level_switches[i]; + } + + if (info->pad_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, + pad_switches); + if (err < 0) + return err; + for (i = 0; i < info->pad_input_count; i++) + private->pad_switch[i] = pad_switches[i]; + } + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = + info->line_out_hw_vol + && volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + int i; + + private->vol_updated = 1; + + for (i = 0; i < private->info->button_count; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + usb_audio_err(mixer->chip, + "scarlett mixer interrupt length %d\n", len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + void *transfer_buffer; + + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb( + mixer->urb, + mixer->chip->dev, + usb_rcvintpipe(mixer->chip->dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT), + transfer_buffer, + SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, + mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8203): + info = &s6i6_gen2_info; + break; + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + if (!scarlett_gen2_mixer_enable) { + usb_audio_err(mixer->chip, + "Focusrite Scarlett Gen 2 Mixer Driver disabled; use " + "options snd_usb_audio scarlett_gen2_mixer_enable=1 " + "to enable and report any issues to g(a)b4.vu"); + return 0; + } + + /* Initialise private data, routing, sequence number */ + err = scarlett2_init_private(mixer, info); + if (err < 0) + return err; + + /* Read volume levels and controls from the interface */ + err = scarlett2_read_configs(mixer); + if (err < 0) + return err; + + /* Create the analogue output controls */ + err = scarlett2_add_line_out_ctls(mixer); + if (err < 0) + return err; + + /* Create the analogue input controls */ + err = scarlett2_add_line_in_ctls(mixer); + if (err < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + err = scarlett2_add_mux_enums(mixer); + if (err < 0) + return err; + + /* Create the matrix mixer controls */ + err = scarlett2_add_mixer_ctls(mixer); + if (err < 0) + return err; + + /* Create the level meter controls */ + err = scarlett2_add_meter_ctl(mixer); + if (err < 0) + return err; + + /* Set up the interrupt polling if there are hardware buttons */ + if (info->button_count) { + err = scarlett2_mixer_status_create(mixer); + if (err < 0) + return err; + } + + return 0; +} + +static void snd_scarlett_gen2_private_free(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + cancel_delayed_work_sync(&private->work); + kfree(private); + mixer->private_data = NULL; +} + +static void snd_scarlett_gen2_private_suspend(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + cancel_delayed_work_sync(&private->work); +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */

2 1

[alsa-devel] [PATCH] sound: ppc: snd_ps3: Remove Unneeded variable: "ret"
by Hariprasad Kelam 10 Jul '19

10 Jul '19

This patch fixes below issue reported by coccicheck sound/ppc/snd_ps3.c:631:5-8: Unneeded variable: "ret". Return "0" on line 668 We cannot change return type of snd_ps3_pcm_trigger as it is registered with snd_pcm_ops->trigger Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/ppc/snd_ps3.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 71b7fd3..c213eb7 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -628,7 +628,6 @@ static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); - int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -665,7 +664,7 @@ static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream, } - return ret; + return 0; }; /* -- 2.7.4

2 1

[alsa-devel] [PATCH] sound: pci: lx6464es: Remove unneeded variable err
by Hariprasad Kelam 10 Jul '19

10 Jul '19

This patch fixes below issue reported by coccicheck sound/pci/lx6464es/lx6464es.c:256:5-8: Unneeded variable: "err". Return "0" on line 258 We cannot change return value as its registered with snd_pcm_ops->close Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/pci/lx6464es/lx6464es.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 1771a6d..583ca73 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -253,9 +253,8 @@ static int lx_pcm_open(struct snd_pcm_substream *substream) static int lx_pcm_close(struct snd_pcm_substream *substream) { - int err = 0; dev_dbg(substream->pcm->card->dev, "->lx_pcm_close\n"); - return err; + return 0; } static snd_pcm_uframes_t lx_pcm_stream_pointer(struct snd_pcm_substream -- 2.7.4

2 2

[alsa-devel] [PATCH] ALSA: pcm: Enable MMAP status and control for ARMv7 and ARMv8
by twischer＠de.adit-jv.com 10 Jul '19

10 Jul '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Since ARMv7 hardware cache coherence is supported. "The SCU maintains coherency between the individual data caches in the Cortex-A5 MPCore processor using a variation of the MOESI protocol" [1]. Therefore this patch enables the MMAP access to the status and control structures. This avoids HWSYYNC ioctl calls and therefore lowers the CPU usage. [1] http://infocenter.arm.com/help/index.jsp?topic=/com.arm.doc.ddi0434c/ BABJECBF.html Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1d84529..b8019ef 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3225,7 +3225,8 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) * Only on coherent architectures, we can mmap the status and the control records * for effcient data transfer. On others, we have to use HWSYNC ioctl... */ -#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA) +#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA) || \ + (defined(CONFIG_ARM) && defined(CONFIG_CPU_V7)) || defined(CONFIG_ARM64) /* * mmap status record */ -- 2.7.4

3 4

[alsa-devel] [PATCH v2 0/4] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 10 Jul '19

10 Jul '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v1 to v2: - hdmi-codec.c: cleanup the sequence of hdmi_codec_jack_report and hdmi_codec_set_jack_detect. - dw-hdmi.c: change argument of hdmi_codec_plugged_cb so it takes a generic device which has hdmi_codec_priv in its drvdata. - dw-hdmi.c: add a helper function handle_plugged_change to check audio platform device and codec platform device before calling callback. - dw-hdmi-c: avoid setting callback function if audio platform device or codec platform device is missing. - rockchip_max98090: fix the checking of return code when setting sysclk on cpu_dai and codec_dai. cpu_dai error should be reported. For HDMI codec_dai, there is no need to set sysclk. Cheng-Yi Chiang (4): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../gpu/drm/bridge/synopsys/dw-hdmi-audio.h | 3 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 10 ++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 55 ++++++++- include/sound/hdmi-codec.h | 16 +++ sound/soc/codecs/hdmi-codec.c | 45 +++++++ sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 6 files changed, 221 insertions(+), 24 deletions(-) -- 2.22.0.410.gd8fdbe21b5-goog

1 4

[alsa-devel] [PATCH 0/7] Add HDMI audio support on RK3288 veyron board
by Cheng-Yi Chiang 10 Jul '19

10 Jul '19

This patch series is to support HDMI audio on RK3288 veyron board. To support jack reporting, there are two old patches: video: add HDMI state notifier support <https://lore.kernel.org/linux-media/20161213150813.37966-2-hverkuil@xs4all.…> ASoC: hdmi-codec: use HDMI state notifier to add jack support <https://patchwork.kernel.org/patch/9430355/> They are modified to pass checkpatch checking based on latest ASoC tree https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next branch. With these two patches at hand, hdmi-notifier support is then added to dw-hdmi driver so the plug/unplug event can be passed to codec driver. The rest patches are about machine driver rockchip_max98090. A HDMI DAI link is added for HDMI playback so there will be two devices on this sound card. One for max98090 and one for HDMI. The HDMI node is passed from DTS so machine driver can set the correct hdmi-notifier on codec driver. Cheng-Yi Chiang (5): drm/bridge/synopsys: dw-hdmi: Add HDMI notifier support ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip: rockchip-max98090: Add node for HDMI ASoC: rockchip_max98090: Add HDMI jack support ARM: dts: rockchip: Specify HDMI node to sound card node Hans Verkuil (1): video: add HDMI state notifier support Philipp Zabel (1): ASoC: hdmi-codec: use HDMI state notifier to add jack support .../bindings/sound/rockchip-max98090.txt | 2 + MAINTAINERS | 6 + .../boot/dts/rk3288-veyron-analog-audio.dtsi | 1 + drivers/gpu/drm/bridge/synopsys/Kconfig | 1 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 28 +++- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 145 ++++++++++++++++++ include/linux/hdmi-notifier.h | 112 ++++++++++++++ include/sound/hdmi-codec.h | 7 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 104 ++++++++++++- sound/soc/rockchip/rockchip_max98090.c | 123 ++++++++++++--- 13 files changed, 508 insertions(+), 26 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.22.0.rc1.257.g3120a18244-goog

5 29

[alsa-devel] [PATCH] sound: soc: codecs: wcd9335: add irqflag IRQF_ONESHOT flag
by Hariprasad Kelam 10 Jul '19

10 Jul '19

Add IRQF_ONESHOT to ensure "Interrupt is not reenabled after the hardirq handler finished". fixes below issue reported by coccicheck sound/soc/codecs/wcd9335.c:4068:8-33: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/soc/codecs/wcd9335.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 85737fe..7ab9bf6f 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4056,6 +4056,9 @@ static struct wcd9335_irq wcd9335_irqs[] = { static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) { int irq, ret, i; + unsigned long irqflags; + + irqflags = IRQF_TRIGGER_RISING | IRQF_ONESHOT; for (i = 0; i < ARRAY_SIZE(wcd9335_irqs); i++) { irq = regmap_irq_get_virq(wcd->irq_data, wcd9335_irqs[i].irq); @@ -4067,7 +4070,7 @@ static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) ret = devm_request_threaded_irq(wcd->dev, irq, NULL, wcd9335_irqs[i].handler, - IRQF_TRIGGER_RISING, + irqflags, wcd9335_irqs[i].name, wcd); if (ret) { dev_err(wcd->dev, "Failed to request %s\n", -- 2.7.4

2 2

[alsa-devel] Reminder: 4 open syzbot bugs in sound subsystem
by Eric Biggers 09 Jul '19

09 Jul '19

[This email was generated by a script. Let me know if you have any suggestions to make it better, or if you want it re-generated with the latest status.] Of the currently open syzbot reports against the upstream kernel, I've manually marked 4 of them as possibly being bugs in the sound subsystem. I've listed these reports below, sorted by an algorithm that tries to list first the reports most likely to be still valid, important, and actionable. If you believe a bug is no longer valid, please close the syzbot report by sending a '#syz fix', '#syz dup', or '#syz invalid' command in reply to the original thread, as explained at https://goo.gl/tpsmEJ#status If you believe I misattributed a bug to the sound subsystem, please let me know, and if possible forward the report to the correct people or mailing list. Here are the bugs: -------------------------------------------------------------------------------- Title: INFO: rcu detected stall in snd_seq_write Last occurred: 72 days ago Reported: 316 days ago Branches: Mainline and others Dashboard link: https://syzkaller.appspot.com/bug?id=33501520944e11adedf1c454eec4cb818bee16… Original thread: https://lkml.kernel.org/lkml/000000000000e5050205746dcbb0@google.com/T/#u This bug has a syzkaller reproducer only. The original thread for this bug received 1 reply, 315 days ago. If you fix this bug, please add the following tag to the commit: Reported-by: syzbot+97aae04ce27e39cbfca9(a)syzkaller.appspotmail.com If you send any email or patch for this bug, please consider replying to the original thread. For the git send-email command to use, or tips on how to reply if the thread isn't in your mailbox, see the "Reply instructions" at https://lkml.kernel.org/r/000000000000e5050205746dcbb0@google.com -------------------------------------------------------------------------------- Title: WARNING: suspicious RCU usage in line6_pcm_acquire Last occurred: 44 days ago Reported: 75 days ago Branches: Mainline (with usb-fuzzer patches) Dashboard link: https://syzkaller.appspot.com/bug?id=a0ea128a37bfe56208042c02d080873dcbdf69… Original thread: https://lkml.kernel.org/lkml/0000000000007cb1ee0587591549@google.com/T/#u Unfortunately, this bug does not have a reproducer. No one has replied to the original thread for this bug yet. This looks like a bug in a sound USB driver. If you fix this bug, please add the following tag to the commit: Reported-by: syzbot+06b7a5a8c4acc0445995(a)syzkaller.appspotmail.com If you send any email or patch for this bug, please consider replying to the original thread. For the git send-email command to use, or tips on how to reply if the thread isn't in your mailbox, see the "Reply instructions" at https://lkml.kernel.org/r/0000000000007cb1ee0587591549@google.com -------------------------------------------------------------------------------- Title: KASAN: use-after-free Read in wake_up_if_idle Last occurred: 117 days ago Reported: 253 days ago Branches: Mainline and others Dashboard link: https://syzkaller.appspot.com/bug?id=b1e300cd7b124fc83dd4199d4d1df26310111b… Original thread: https://lkml.kernel.org/lkml/00000000000066ab7105795f245e@google.com/T/#u Unfortunately, this bug does not have a reproducer. No one replied to the original thread for this bug. If you fix this bug, please add the following tag to the commit: Reported-by: syzbot+2c1253bc508adef78a7f(a)syzkaller.appspotmail.com If you send any email or patch for this bug, please consider replying to the original thread. For the git send-email command to use, or tips on how to reply if the thread isn't in your mailbox, see the "Reply instructions" at https://lkml.kernel.org/r/00000000000066ab7105795f245e@google.com -------------------------------------------------------------------------------- Title: INFO: task hung in snd_seq_write Last occurred: 110 days ago Reported: 315 days ago Branches: Mainline and others Dashboard link: https://syzkaller.appspot.com/bug?id=9366cef8455b032553567ce096a5b31d8307b7… Original thread: https://lkml.kernel.org/lkml/0000000000001f74de0574710d3e@google.com/T/#u Unfortunately, this bug does not have a reproducer. No one replied to the original thread for this bug. If you fix this bug, please add the following tag to the commit: Reported-by: syzbot+4c595632b98bb8ffcc66(a)syzkaller.appspotmail.com If you send any email or patch for this bug, please consider replying to the original thread. For the git send-email command to use, or tips on how to reply if the thread isn't in your mailbox, see the "Reply instructions" at https://lkml.kernel.org/r/0000000000001f74de0574710d3e@google.com

1 0

[alsa-devel] [PATCH] sound: soc: codecs: wcd9335: fix Unneeded variable: "ret"
by Hariprasad Kelam 09 Jul '19

09 Jul '19

this fixes below issues reported by coccicheck sound/soc/codecs/wcd9335.c:3611:5-8: Unneeded variable: "ret". Return "0" on line 3625 sound/soc/codecs/wcd9335.c:3941:5-8: Unneeded variable: "ret". Return "0" on line 3971 sound/soc/codecs/wcd9335.c:3745:5-8: Unneeded variable: "ret". Return "0" on line 3784 sound/soc/codecs/wcd9335.c:3896:5-8: Unneeded variable: "ret". Return "0" on line 3934 sound/soc/codecs/wcd9335.c:3026:5-8: Unneeded variable: "ret". Return "0" on line 3038 We cannot change return of these functions as they are callback functions. Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/soc/codecs/wcd9335.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 1bbbe42..e13af36 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -3018,7 +3018,6 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = snd_soc_component_get_drvdata(comp); struct wcd_slim_codec_dai_data *dai = &wcd->dai[w->shift]; - int ret = 0; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3030,7 +3029,7 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w, break; } - return ret; + return 0; } static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w, @@ -3603,7 +3602,6 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3617,7 +3615,7 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static void wcd9335_codec_hph_post_pa_config(struct wcd9335_codec *wcd, @@ -3737,7 +3735,6 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3776,7 +3773,7 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, @@ -3888,7 +3885,6 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3926,14 +3922,13 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); - int ret = 0; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3963,7 +3958,7 @@ static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) -- 2.7.4

1 0

[alsa-devel] [PATCH] MLK-22197 sound: asoc: add micfil dc remover amixer control
by Irina Patru 09 Jul '19

09 Jul '19

Add control for DC Remover Control register: 11b - DC Remover is bypassed 00b - DC Remover is cut-off at 21Hz 01b - DC Remover is cut-off at 83Hz 10b - DC Remover is cut-off at 152Hz This control modifies DC mode for all channels. For more information see DC Remover Block Diagram in micfil reference manual. Signed-off-by: Irina Patru <ioana-irina.patru(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 59 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_micfil.h | 10 +++++++ 2 files changed, 69 insertions(+) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 7bb8d4cb5f71..2eaf9fda4865 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -45,6 +45,7 @@ struct fsl_micfil { bool slave_mode; int channel_gain[8]; int clk_src_id; + int dc_remover; int vad_sound_gain; int vad_noise_gain; int vad_input_gain; @@ -140,6 +141,17 @@ static const char * const micfil_clk_src_texts[] = { "Auto", "AudioPLL1", "AudioPLL2", "ExtClk3", }; +/* DC Remover Control + * Filter Bypassed 1 1 + * Cut-off @21Hz 0 0 + * Cut-off @83Hz 0 1 + * Cut-off @152HZ 1 0 + */ +static const char * const micfil_dc_remover_texts[] = { + "Cut-off @21Hz", "Cut-off @83Hz", + "Cut-off @152Hz", "Bypass", +}; + static const struct soc_enum fsl_micfil_quality_enum = SOC_ENUM_SINGLE(REG_MICFIL_CTRL2, MICFIL_CTRL2_QSEL_SHIFT, @@ -168,6 +180,10 @@ static const struct soc_enum fsl_micfil_hwvad_rate_enum = static const struct soc_enum fsl_micfil_clk_src_enum = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(micfil_clk_src_texts), micfil_clk_src_texts); +static const struct soc_enum fsl_micfil_dc_remover_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(micfil_dc_remover_texts), + micfil_dc_remover_texts); + static int micfil_put_clk_src(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -194,6 +210,45 @@ static int micfil_get_clk_src(struct snd_kcontrol *kcontrol, return 0; } +static int micfil_put_dc_remover_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + struct fsl_micfil *micfil = snd_soc_component_get_drvdata(comp); + unsigned int *item = ucontrol->value.enumerated.item; + int val = snd_soc_enum_item_to_val(e, item[0]); + int i = 0, ret = 0; + u32 reg_val = 0; + + if (val < 0 || val > 3) + return -1; + micfil->dc_remover = val; + + /* Calculate total value for all channels */ + for (i = 0; i < 8; i++) + reg_val |= MICFIL_DC_MODE(val, i); + + /* Update DC Remover mode for all channels */ + ret = snd_soc_component_update_bits(comp, + REG_MICFIL_DC_CTRL, + MICFIL_DC_CTRL_MASK, + reg_val); + return ret; +} + +static int micfil_get_dc_remover_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_micfil *micfil = snd_soc_component_get_drvdata(comp); + + ucontrol->value.enumerated.item[0] = micfil->dc_remover; + + return 0; +} + + static int hwvad_put_init_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -676,6 +731,10 @@ static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_ENUM_EXT("Clock Source", fsl_micfil_clk_src_enum, micfil_get_clk_src, micfil_put_clk_src), + SOC_ENUM_EXT("MICFIL DC Remover Control", + fsl_micfil_dc_remover_enum, + micfil_get_dc_remover_state, + micfil_put_dc_remover_state), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "HWVAD Input Gain", diff --git a/sound/soc/fsl/fsl_micfil.h b/sound/soc/fsl/fsl_micfil.h index 792187b717f0..e47dba9b90b8 100644 --- a/sound/soc/fsl/fsl_micfil.h +++ b/sound/soc/fsl/fsl_micfil.h @@ -278,6 +278,16 @@ #define MICFIL_HWVAD_HPF_102HZ 3 #define MICFIL_HWVAD_FRAMET_DEFAULT 10 +/* MICFIL DC Remover Control Register -- REG_MICFIL_DC_CTRL */ +#define MICFIL_DC_CTRL_SHIFT 0 +#define MICFIL_DC_CTRL_MASK 0xFFFF +#define MICFIL_DC_CTRL_WIDTH 2 +#define MICFIL_DC_CHX_SHIFT(v) (2 * (v)) +#define MICFIL_DC_CHX_MASK(v) ((BIT(MICFIL_DC_CTRL_WIDTH) - 1) \ + << MICFIL_DC_CHX_SHIFT(v)) +#define MICFIL_DC_MODE(v1, v2) (((v1) << MICFIL_DC_CHX_SHIFT(v2)) \ + & MICFIL_DC_CHX_MASK(v2)) + /* MICFIL Output Control Register */ #define MICFIL_OUTGAIN_CHX_SHIFT(v) (4 * (v)) -- 2.21.0

4 3

[alsa-devel] [PATCH 1/4] ALSA: compress: Fix regression on compressed capture streams
by Charles Keepax 09 Jul '19

09 Jul '19

A previous fix to the stop handling on compressed capture streams causes some knock on issues. The previous fix updated snd_compr_drain_notify to set the state back to PREPARED for capture streams. This causes some issues however as the handling for snd_compr_poll differs between the two states and some user-space applications were relying on the poll failing after the stream had been stopped. To correct this regression whilst still fixing the original problem the patch was addressing, update the capture handling to skip the PREPARED state rather than skipping the SETUP state as it has done until now. Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/compress_driver.h | 5 +---- sound/core/compress_offload.c | 16 +++++++++++----- 2 files changed, 12 insertions(+), 9 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index c5188ff724d12..bc88d6f964da9 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -173,10 +173,7 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) if (snd_BUG_ON(!stream)) return; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); } diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a1a6fd75cfe50..f031495311ee4 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -587,10 +587,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -706,8 +703,17 @@ static int snd_compr_start(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + if (stream->direction != SND_COMPRESS_CAPTURE) + return -EPERM; + break; + case SNDRV_PCM_STATE_PREPARED: + break; + default: return -EPERM; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); if (!retval) stream->runtime->state = SNDRV_PCM_STATE_RUNNING; -- 2.11.0

2 5

[alsa-devel] [PATCH 0/4] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 09 Jul '19

09 Jul '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Cheng-Yi Chiang (4): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../gpu/drm/bridge/synopsys/dw-hdmi-audio.h | 3 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 10 ++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 32 ++++- include/sound/hdmi-codec.h | 16 +++ sound/soc/codecs/hdmi-codec.c | 52 ++++++++ sound/soc/rockchip/rockchip_max98090.c | 112 ++++++++++++++---- 6 files changed, 201 insertions(+), 24 deletions(-) -- 2.22.0.410.gd8fdbe21b5-goog

6 19

Re: [alsa-devel] [PATCH V2 2/2] ASoC: fsl_esai: recover the channel swap after xrun
by S.j. Wang 09 Jul '19

09 Jul '19

> > > > + > > > > + /* restore registers by regcache_sync */ > > > > + fsl_esai_register_restore(esai_priv); > > > > + > > > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, > > > > + ESAI_xCR_xPR_MASK, 0); > > > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, > > > > + ESAI_xCR_xPR_MASK, 0); > > > > > > And just for curious, can (or shall) we stuff this personal reset to > > > the reset() function? I found this one is a part of the reset > > > routine being mentioned in the RM -- it was done after ESAI reset is > done via ECR register. > > > > > > > There is a problem to do this, TPR/RPR need to be clear after > > configure the control register. (TCCR, TCR). So it seems not only one > > place (reset function) need to be changed. > > Do you know (or remember) why we suddenly involve this TPR/PRP? > The driver has no problem so far, even if we don't have them. > > The "personal reset" sounds like a feature that we would use to reset TX or > RX individually, while this hw_reset() does a full reset for both TX and RX. > So I wonder whether they're necessary. The hw_reset flow is suggested by design team, so involve TRP/RPP is from them, I don't know the detail. Best regards Wang shengjiu

1 0

Re: [alsa-devel] [PATCH V2 2/2] ASoC: fsl_esai: recover the channel swap after xrun
by S.j. Wang 09 Jul '19

09 Jul '19

> > > + > > + /* restore registers by regcache_sync */ > > + fsl_esai_register_restore(esai_priv); > > + > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, > > + ESAI_xCR_xPR_MASK, 0); > > + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, > > + ESAI_xCR_xPR_MASK, 0); > > And just for curious, can (or shall) we stuff this personal reset to the reset() > function? I found this one is a part of the reset routine being mentioned in > the RM -- it was done after ESAI reset is done via ECR register. > There is a problem to do this, TPR/RPR need to be clear after configure the control register. (TCCR, TCR). So it seems not only one place (reset function) need to be changed. Best regards Wang shengjiu

2 1

[alsa-devel] [PATCH] ALSA: hda: Simplify snd_hdac_refresh_widgets()
by Takashi Iwai 08 Jul '19

08 Jul '19

Along with the recent fix for the races of snd_hdac_refresh_widgets() it turned out that the instantiation of widgets sysfs at snd_hdac_sysfs_reinit() could cause a race. The race itself was already covered later by extending the mutex protection range, the commit 98482377dc72 ("ALSA: hda: Fix widget_mutex incomplete protection"), but this also indicated that the call of *_reinit() is basically superfluous, as the widgets shall be created sooner or later from snd_hdac_device_register(). This patch removes the redundant call of snd_hdac_sysfs_reinit() at first. By this removal, the sysfs argument itself in snd_hdac_refresh_widgets() becomes superfluous, too, because the only case sysfs=false is always with codec->widgets=NULL. So, we drop this redundant argument as well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 2 +- sound/hda/hdac_device.c | 13 +++++-------- sound/hda/hdac_sysfs.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 2 +- 5 files changed, 9 insertions(+), 12 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index e8346784cf3f..f475293d0668 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -120,7 +120,7 @@ void snd_hdac_device_unregister(struct hdac_device *codec); int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size); -int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs); +int snd_hdac_refresh_widgets(struct hdac_device *codec); unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid, unsigned int verb, unsigned int parm); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 11050bfd8068..a265c1d68876 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -89,7 +89,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, fg = codec->afg ? codec->afg : codec->mfg; - err = snd_hdac_refresh_widgets(codec, false); + err = snd_hdac_refresh_widgets(codec); if (err < 0) goto error; @@ -394,9 +394,8 @@ static void setup_fg_nodes(struct hdac_device *codec) /** * snd_hdac_refresh_widgets - Reset the widget start/end nodes * @codec: the codec object - * @sysfs: re-initialize sysfs tree, too */ -int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs) +int snd_hdac_refresh_widgets(struct hdac_device *codec) { hda_nid_t start_nid; int nums, err = 0; @@ -414,11 +413,9 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs) goto unlock; } - if (sysfs) { - err = hda_widget_sysfs_reinit(codec, start_nid, nums); - if (err < 0) - goto unlock; - } + err = hda_widget_sysfs_reinit(codec, start_nid, nums); + if (err < 0) + goto unlock; codec->num_nodes = nums; codec->start_nid = start_nid; diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 909d5ef1179c..e56e83325903 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -428,7 +428,7 @@ int hda_widget_sysfs_reinit(struct hdac_device *codec, int i; if (!codec->widgets) - return hda_widget_sysfs_init(codec); + return 0; tree = kmemdup(codec->widgets, sizeof(*tree), GFP_KERNEL); if (!tree) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fcdf2cd3783b..d1a0e0de80ac 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1016,7 +1016,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) hda_nid_t fg; int err; - err = snd_hdac_refresh_widgets(&codec->core, true); + err = snd_hdac_refresh_widgets(&codec->core); if (err < 0) return err; diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 660e0587f399..6302ad5b7128 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2043,7 +2043,7 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) "Failed in parse and map nid with err: %d\n", ret); return ret; } - snd_hdac_refresh_widgets(hdev, true); + snd_hdac_refresh_widgets(hdev); /* ASoC specific initialization */ ret = devm_snd_soc_register_component(&hdev->dev, &hdmi_hda_codec, -- 2.16.4

2 2

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 08 Jul '19

08 Jul '19

When DSP boots with timeout error, reinitialize, transfer and boot firmware to recover audio. Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 440bca7afbf1..a2844bdca1b8 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -455,13 +455,18 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + + ret = bxt_sst_init_fw(skl->dev, skl); + dev_warn(ctx->dev, "Reload fw status: %d\n", ret); + if (ret < 0) { + dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); + ret = -EIO; + goto err; + } } } -- 2.17.1

3 2

[alsa-devel] [PATCH 0/7] ALSA: firewire-lib: some fixes for tracepoints events
by Takashi Sakamoto 08 Jul '19

08 Jul '19

Hi, This kernel module, firewire-lib, adds 'amdtp_packet' tracepoints events to Linux system. After heavy code refactoring for v5.3, some event parameters include invalid values. This patchset is to fix them, and the last part of my refactoring work in this development period. Takashi Sakamoto (7): ALSA: firewire-lib: fix invalid length of tx packet payload for tracepoint events ALSA: firewire-lib/ficeface: fix initial value of data block counter for IR context with CIP_NO_HEADER ALSA: firewire-lib: fix initial value of data block count for IR context without CIP_DBC_IS_END_EVENT ALSA: firewire-lib: fix different data block counter between probed event and transferred isochronous packet ALSA: firewire-lib: code refactoring for error path of parser for CIP header ALSA: firewire-lib: code refactoring for post operation to data block counter ALSA: firewire-lib: code refactoring for local variables sound/firewire/amdtp-stream.c | 59 ++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 28 deletions(-) -- 2.20.1

2 8

[alsa-devel] [GIT PULL] ASoC updates for v5.3
by Mark Brown 08 Jul '19

08 Jul '19

The following changes since commit 6fbc7275c7a9ba97877050335f290341a1fd8dbf: Linux 5.2-rc7 (2019-06-30 11:25:36 +0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.3 for you to fetch changes up to a98429acadefc2b36611220f51659ecb3c1f35d2: Merge remote-tracking branch 'asoc/topic/meson' into asoc-next (2019-07-06 12:25:28 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.3 This is a very big update, mainly thanks to Morimoto-san's refactoring work and some fairly large new drivers. - Lots more work on moving towards a component based framework from Morimoto-san. - Support for force disconnecting muxes from Jerome Brunet. - New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant CX2072X, Realtek RT1011 and RT1308. ---------------------------------------------------------------- Amadeusz Sławiński (12): ASoC: core: Tell codec that jack is being removed ASoC: Intel: Skylake: Initialize lists before access so they are safe to use ALSA: hdac: Fix codec name after machine driver is unloaded and reloaded ASoC: compress: Fix memory leak from snd_soc_new_compress ASoC: Intel: Skylake: Don't return failure on machine driver reload ASoC: Intel: Skylake: Add function to cleanup debugfs interface ASoC: Intel: Skylake: Properly cleanup on component removal ASoC: Intel: Skylake: Fix NULL ptr dereference when unloading clk dev ASoC: Intel: hdac_hdmi: Set ops to NULL on remove ASoC: topology: Consolidate how dtexts and dvalues are freed ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow ASoC: Intel: Skylake: Reset pipeline before its deletion Amir Goldstein (1): ASoC: rename functions that pollute the simple_xxx namespace Andy Shevchenko (1): ASoC: Intel: Skylake: Switch to modern UUID API Arnd Bergmann (3): ASoC: rockchip: pdm: select CONFIG_RATIONAL ASoC: cx2072x: mark PM function as __maybe_unused ASoC: SOF: disallow building without CONFIG_PCI again Bard Liao (1): ASoC: SOF: dont wake dsp up in kcontrol IO Bard liao (1): ASoC: SOF: send time stamp to FW for alignment Ben Zhang (2): ASoC: rt5677: clear interrupts by polarity flip ASoC: rt5677: handle concurrent interrupts Bjorn Andersson (2): ASoC: qcom: common: Mark links as nonatomic ASoC: qcom: common: Fix NULL pointer in of parser Cezary Rojewski (3): ASoC: Intel: Skylake: Fix incorrect capture position reporting ASoC: Intel: Skylake: Read HIPCT extension before clearing DONE bit ASoC: Intel: Common: Fix NULL dereference in tx_wait_done Charles Keepax (1): ASoC: madera: Update SPDX headers Clément Péron (4): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Move quirks to the top ASoC: sun4i-spdif: Add TX fifo bit flush quirks ASoC: sun4i-spdif: Add support for H6 SoC Codrin Ciubotariu (3): ASoC: codecs: ad193x: Fix memory corruption on BE 64b systems ASoC: codecs: ad193x: Group register initialization at probe ASoC: codecs: ad193x: Reset used registers at probe Colin Ian King (5): ASoC: cx2072x: remove redundant assignment to pulse_len ASoC: cx2072x: fix integer overflow on unsigned int multiply ASoC: cx2072x: fix spelling mistake "configued" -> "configured" ASoC: topology: fix memory leaks on sm, se and sbe ASoC: rt5665: remove redundant assignment to variable idx Curtis Malainey (4): ASoC: rt5677-spi: Rename driver to differentiate from main codec ASoC: rt5677: Add component driver name ASoC: rt5677: Add missing voice activation register definitions ASoC: rt5677: depop stereo dac Daniel Baluta (1): ASoC: SOF: Do not send cmd via SHIM register David Lin (1): ASoC: nau8822: support master mode Derek Fang (1): ASoC: rt1308: Add RT1308 amplifier driver Dragos Tarcatu (1): ASOC: SOF: ipc: prevent logging trace messages Dylan Reid (1): ASoC: rt5677: use more of the volume range from DACs Enric Balletbo i Serra (1): ASoC: rk3399_gru_sound: Support 32, 44.1 and 88.2 kHz sample rates Evan Green (2): ASoC: SOF: Add Comet Lake PCI IDs ASoC: Intel: Skylake: Add Cometlake PCI IDs Fletcher Woodruff (2): ASoC: rt5677: fall back to DT prop names on error ASoC: rt5677: move jack-detect init to i2c probe Geert Uytterhoeven (1): ASoC: Add missing newline at end of file Gen Zhang (1): wcd9335: fix a incorrect use of kstrndup() Greg Kroah-Hartman (2): ASoC: wm_adsp: no need to check return value of debugfs_create functions ASoC: fsl: no need to check return value of debugfs_create functions Gustavo A. R. Silva (5): ASoC: dapm: Use struct_size() in krealloc() ASoC: Intel: Skylake: Use struct_size() helper ASoC: qdsp6: q6core: Use struct_size() in kmemdup() ASoC: SOF: topology: Use struct_size() helper ASoC: SOF: Use struct_size() in kmemdup() Gustaw Lewandowski (1): ASoC: Intel: Fix race condition in IPC rx list Janusz Krzysztofik (1): ASoC: ti: Fix SDMA users not providing channel names Jaska Uimonen (2): ASoC: SOF: uapi: remove unused sof header files ASoC: SOF: topology: add support for mux/demux component Jerome Brunet (21): ASoC: hdmi-codec: remove function name debug traces ASoC: hdmi-codec: remove reference to the current substream ASoC: hdmi-codec: remove reference to the dai drivers in the private data ASoC: hdmi-codec: remove ops dependency on the dai id ASoC: hdmi-codec: re-introduce mutex locking ASoC: meson: axg-card: set link name based on link node name ASoC: dapm: allow muxes to force a disconnect ASoC: meson: add tohdmitx DT bindings ASoC: meson: axg-card: add basic codec-to-codec link support ASoC: meson: add g12a tohdmitx control ASoC: meson: axg-card: fix null pointer dereference in clean up ASoC: meson: axg-tdmin: right_j is not supported ASoC: meson: axg-tdmout: right_j is not supported ASoC: meson: axg-tdm: fix sample clock inversion ASoC: meson: axg-tdm: consistently use SND_SOC_DAIFMT defines ASoC: meson: add tohdmitx DT bindings ASoC: soc-core: defer card registration if codec component is missing ASoC: soc-core: support dai_link with platforms_num != 1 ASoC: meson: axg-card: remove useless check on codec ASoC: meson: axg-tdm-formatter: add reset to the bindings documentation ASoC: meson: axg-tdm-formatter: add reset Jiaxin Yu (3): ASoC: Mediatek: MT8183: fix compile error ASoC: Mediatek: MT8183: fix compile error ASoC: Mediatek: MT8183: enable dev runtime suspend and resume John Hsu (1): ASoC: nau8825: fix fake interruption when booting Kai Vehmanen (8): ASoC: SOF: trace: remove code duplication in sof_wait_trace_avail() ASoC: SOF: force end-of-file for debugfs trace at suspend ASoC: codec: hdac_hdmi: fix pin connections at cvt enable ASoC: SOF: intel: extend IPC dump information ASoC: SOF: ignore unrecoverable CTX_SAVE IPC errors at suspend ASoC: hdac_hdmi: report codec link up/down status to bus ASoC: SOF: add runtime idle callback ASoC: SOF: Intel: implement runtime idle for CNL/APL Kamil Lulko (1): ASoC: Intel: Skylake: Strip T and L from TLV IPCs Keyon Jie (3): ASoC: Intel: bxt-da7219-max98357a: add BE dailink for dmic16k ASoC: SOF: Intel: hda-stream: fix a deadlock with bus->reg_lock ASoC: Intel: sof-rt5682: correct naming for dmic16k Krzysztof Kozlowski (1): Revert "ASoC: core: use component driver name as component name" Kuninori Morimoto (268): ASoC: soc-core: use i on snd_soc_resume() ASoC: soc-core: don't use codec_dais on soc_bind_dai_link() ASoC: soc-core: tidyup soc_bind_dai_link() comment balance ASoC: soc-core: add soc_component_to_node() ASoC: soc-core: use soc_find_component() at snd_soc_get_dai_id() ASoC: soc.h: fe_compr can be bit field ASoC: soc-pcm: do cpu_dai related operation at same place ASoC: soc-pcm: add missing cpu_dai->rate = 0 ASoC: soc-pcm: add soc_pcm_components_open() ASoC: rsnd: move pcm_new from snd_soc_component_driver to snd_soc_dai_driver ASoC: soc-core: WARN() is not related to component->driver->probe ASoC: soc-pcm: fixup try_module_get()/module_put() timing ASoC: soc-core: use snd_soc_dai_link_component for CPU ASoC: simple-card: support snd_soc_dai_link_component style for cpu ASoC: soc.h: add sound dai_link connection macro ASoC: mediatek: mt6797-mt6351: use modern dai_link style ASoC: mediatek: mt2701-wm8960: use modern dai_link style ASoC: mediatek: mt2701-cs42448: use modern dai_link style ASoC: mediatek: mt8173-max98090: use modern dai_link style ASoC: mediatek: mt8173-rt5650: use modern dai_link style ASoC: mediatek: mt8173-rt5650-rt5676: use modern dai_link style ASoC: mediatek: mt8173-rt5650-rt5514: use modern dai_link style ASoC: mediatek: mt8183-da7219-max98357: use modern dai_link style ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use modern dai_link style ASoC: samsung: bells: use modern dai_link style ASoC: samsung: smdk_wm8580: use modern dai_link style ASoC: samsung: smdk_wm8994: use modern dai_link style ASoC: samsung: smdk_wm8994pcm: use modern dai_link style ASoC: samsung: smdk_spdif: use modern dai_link style ASoC: samsung: h1940_uda1380: use modern dai_link style ASoC: samsung: arndale_rt5631: use modern dai_link style ASoC: samsung: tobermory: use modern dai_link style ASoC: samsung: tm2_wm5110: use modern dai_link style ASoC: samsung: speyside: use modern dai_link style ASoC: samsung: snow: use modern dai_link style ASoC: samsung: smartq_wm8987: use modern dai_link style ASoC: samsung: s3c24xx_uda134x: use modern dai_link style ASoC: samsung: s3c24xx_simtec_tlv320aic23: use modern dai_link style ASoC: samsung: s3c24xx_simtec_hermes: use modern dai_link style ASoC: samsung: rx1950_uda1380: use modern dai_link style ASoC: samsung: odroid: use modern dai_link style ASoC: samsung: neo1973_wm8753: use modern dai_link style ASoC: samsung: lowland: use modern dai_link style ASoC: samsung: littlemill: use modern dai_link style ASoC: samsung: jive_wm8750: use modern dai_link style ASoC: pxa: brownstone: use modern dai_link style ASoC: pxa: corgi: use modern dai_link style ASoC: pxa: e740_wm9705: use modern dai_link style ASoC: pxa: e750_wm9705: use modern dai_link style ASoC: pxa: e800_wm9712: use modern dai_link style ASoC: pxa: em-x270: use modern dai_link style ASoC: pxa: hx4700: use modern dai_link style ASoC: pxa: imote2: use modern dai_link style ASoC: pxa: magician: use modern dai_link style ASoC: pxa: mioa701_wm9713: use modern dai_link style ASoC: pxa: palm27x: use modern dai_link style ASoC: pxa: zylonite: use modern dai_link style ASoC: pxa: z2: use modern dai_link style ASoC: pxa: ttc-dkb: use modern dai_link style ASoC: pxa: tosa: use modern dai_link style ASoC: pxa: spitz: use modern dai_link style ASoC: pxa: poodle: use modern dai_link style ASoC: kirkwood: armada-370-db: use modern dai_link style ASoC: ti: n810: use modern dai_link style ASoC: ti: osk5912: use modern dai_link style ASoC: ti: ams-delta: use modern dai_link style ASoC: ti: omap3pandora: use modern dai_link style ASoC: ti: davinci-evm: use modern dai_link style ASoC: ti: omap-hdmi: use modern dai_link style ASoC: ti: omap-abe-twl6040: use modern dai_link style ASoC: ti: omap-twl4030: use modern dai_link style ASoC: ti: rx51: use modern dai_link style ASoC: sh: migor: use modern dai_link style ASoC: sh: sh7760-ac97: use modern dai_link style ASoC: amd: acp-da7219-max98357a: use modern dai_link style ASoC: amd: acp-rt5645: use modern dai_link style ASoC: atmel: mikroe-proto: use modern dai_link style ASoC: atmel: sam9x5_wm8731: use modern dai_link style ASoC: atmel: sam9g20_wm8731: use modern dai_link style ASoC: atmel: tse850-pcm5142: use modern dai_link style ASoC: atmel: atmel-wm8904: use modern dai_link style ASoC: atmel: atmel-pdmic: use modern dai_link style ASoC: atmel: atmel-classd: use modern dai_link style ASoC: au1x: db1000: use modern dai_link style ASoC: au1x: db1200: use modern dai_link style ASoC: cirrus: edb93xx: use modern dai_link style ASoC: cirrus: simone: use modern dai_link style ASoC: cirrus: snappercl15: use modern dai_link style ASoC: fsl: efika-audio-fabric: use modern dai_link style ASoC: fsl: imx-spdif: use modern dai_link style ASoC: fsl: mx27vis-aic32x4: use modern dai_link style ASoC: fsl: phycore-ac97: use modern dai_link style ASoC: fsl: wm1133-ev1: use modern dai_link style ASoC: fsl: fsl-asoc-card: use modern dai_link style ASoC: fsl: mpc8610_hpcd/p1022_ds/p1022_rdk: use modern dai_link style ASoC: fsl: imx-es8328: use modern dai_link style ASoC: fsl: eukrea-tlv320: use modern dai_link style ASoC: fsl: imx-sgtl5000: use modern dai_link style ASoC: fsl: imx-mc13783: use modern dai_link style ASoC: fsl: pcm030-audio-fabric: use modern dai_link style ASoC: fsl: imx-audmix: use modern dai_link style ASoC: jz4740: qi_lb60: use modern dai_link style ASoC: meson: axg-card: use modern dai_link style ASoC: mxs: mxs-sgtl5000: use modern dai_link style ASoC: nuc900: nuc900-audio: use modern dai_link style ASoC: qcom: storm: use modern dai_link style ASoC: qcom: common: use modern dai_link style ASoC: qcom: apq8016_sbc: use modern dai_link style ASoC: rockchip: rk3399_gru_sound: use modern dai_link style ASoC: rockchip: rk3288_hdmi_analog: use modern dai_link style ASoC: rockchip: rockchip_rt5645: use modern dai_link style ASoC: rockchip: rockchip_max98090: use modern dai_link style ASoC: sirf: sirf-audio: use modern dai_link style ASoC: ux500: mop500: use modern dai_link style ASoC: sunxi: sun4i-codec: use modern dai_link style ASoC: txx9: txx9aclc-generic: use modern dai_link style ASoC: tegra: tegra_rt5677: use modern dai_link style ASoC: tegra: tegra_wm8753: use modern dai_link style ASoC: tegra: trimslice: use modern dai_link style ASoC: tegra: tegra_alc5632: use modern dai_link style ASoC: tegra: tegra_sgtl5000: use modern dai_link style ASoC: tegra: tegra_rt5640: use modern dai_link style ASoC: tegra: tegra_wm8903: use modern dai_link style ASoC: tegra: tegra_wm9712: use modern dai_link style ASoC: tegra: tegra_max98090: use modern dai_link style ASoC: soc-topology: use modern dai_link style ASoC: vc4: vc4_htmi: use modern dai_link style ASoC: sof: use modern dai_link style ASoC: Intel: bdw-rt5677: use modern dai_link style ASoC: Intel: haswell: use modern dai_link style ASoC: Intel: broadwell: use modern dai_link style ASoC: Intel: bxt_da7219_max98357a: use modern dai_link style ASoC: Intel: bxt_rt298: use modern dai_link style ASoC: Intel: byt-rt5640: use modern dai_link style ASoC: Intel: byt-max98090: use modern dai_link style ASoC: Intel: bytcht_es8316: use modern dai_link style ASoC: Intel: bytcht_nocodec: use modern dai_link style ASoC: Intel: bytcht_da7213: use modern dai_link style ASoC: Intel: bytcht_cx2072x: use modern dai_link style ASoC: Intel: bytcr_rt5651: use modern dai_link style ASoC: Intel: bytcr_rt5640: use modern dai_link style ASoC: Intel: cht_bsw_rt5645: use modern dai_link style ASoC: Intel: cht_bsw_max98090_ti: use modern dai_link style ASoC: Intel: cht_bsw_rt5672: use modern dai_link style ASoC: Intel: cht_bsw_nau8824: use modern dai_link style ASoC: Intel: kbl_rt5660: use modern dai_link style ASoC: Intel: kbl_rt5663_max98927: use modern dai_link style ASoC: Intel: kbl_rt5663_rt5514_max98927: use modern dai_link style ASoC: Intel: kbl_da7219_max98927: use modern dai_link style ASoC: Intel: kbl_da7219_max98357a: use modern dai_link style ASoC: Intel: skl_nau88l25_ssm4567: use modern dai_link style ASoC: Intel: skl_rt286: use modern dai_link style ASoC: Intel: skl_nau88l25_max98357a: use modern dai_link style ASoC: Intel: skl-pcm: use modern dai_link style ASoC: Intel: skl_hda_dsp_common: use modern dai_link style ASoC: Intel: glk_rt5682_max98357a: use modern dai_link style ASoC: Intel: sof_rt5682: use modern dai_link style ASoC: soc-core: remove legacy style dai_link ASoC: soc.h: fixup for_each_card_links() macro ASoC: soc-core: allow no Platform on dai_link ASoC: vc4: vc4_hdmi: don't select unnecessary Platform ASoC: atmel: atmel-classd: don't select unnecessary Platform ASoC: atmel: atmel-pdmic: don't select unnecessary Platform ASoC: atmel: atmel_wm8904: don't select unnecessary Platform ASoC: atmel: mikroe-proto: don't select unnecessary Platform ASoC: atmel: sam9g20_wm8731: don't select unnecessary Platform ASoC: atmel: sam9x5_wm8731: don't select unnecessary Platform ASoC: atmel: tse850-pcm5142: don't select unnecessary Platform ASoC: fsl: eukrea-tlv320: don't select unnecessary Platform ASoC: fsl: fsl-asoc-card: don't select unnecessary Platform ASoC: fsl: imx-es8328: don't select unnecessary Platform ASoC: fsl: imx-sgtl5000: don't select unnecessary Platform ASoC: fsl: imx-spdif: don't select unnecessary Platform ASoC: fsl: imx-audmix: don't select unnecessary Platform ASoC: kirkwood: armada-370-db: don't select unnecessary Platform ASoC: mxs: mxs-sgtl5000: don't select unnecessary Platform ASoC: qcom: apq8016_sbc: don't select unnecessary Platform ASoC: qcom: storm: don't select unnecessary Platform ASoC: rockchip: rk3288_hdmi_analog: don't select unnecessary Platform ASoC: rockchip: rockchip_max98090: don't select unnecessary Platform ASoC: rockchip: rockchip_rt5645: don't select unnecessary Platform ASoC: samsung: arndale_rt5631: don't select unnecessary Platform ASoC: samsung: smdk_wm8994: don't select unnecessary Platform ASoC: samsung: snow: don't select unnecessary Platform ASoC: samsung: tm2_wm5110: don't select unnecessary Platform ASoC: sirf: sirf-audio: don't select unnecessary Platform ASoC: sunxi: sun4i-codec: don't select unnecessary Platform ASoC: tegra: tegra_alc5632: don't select unnecessary Platform ASoC: tegra: tegra_max98090: don't select unnecessary Platform ASoC: tegra: tegra_rt5640: don't select unnecessary Platform ASoC: tegra: tegra_rt5677: don't select unnecessary Platform ASoC: tegra: tegra_sgtl5000: don't select unnecessary Platform ASoC: tegra: tegra_wm8753: don't select unnecessary Platform ASoC: tegra: tegra_wm8903: don't select unnecessary Platform ASoC: tegra: tegra_wm9712: don't select unnecessary Platform ASoC: tegra: trimslice: don't select unnecessary Platform ASoC: ti: davinci-evm: don't select unnecessary Platform ASoC: ti: omap-abe-twl6040: don't select unnecessary Platform ASoC: ti: omap-hdmi: don't select unnecessary Platform ASoC: ti: omap-twl4030: don't select unnecessary Platform ASoC: ti: rx51: don't select unnecessary Platform ASoC: ux500: mop500: don't select unnecessary Platform ASoC: simple-card-utils: don't select unnecessary Platform ASoC: qcom: don't select unnecessary Platform ASoC: rockchip: rk3399_gru_sound: don't select unnecessary Platform ASoC: soc-utils: remove dummy Platform ASoC: rt5514-spi: don't use snd_soc_lookup_component() ASoC: soc-core: call snd_soc_unbind_card() under mutex_lock; ASoC: soc-core: move soc_find_component() ASoC: soc-core: soc_find_component() uses snd_soc_is_matching_component() ASoC: soc-core: soc_find_component() uses snd_soc_dai_link_component ASoC: soc-core: use soc_find_component() at snd_soc_find_dai() ASoC: rsnd: ssiu: tidyup SSI_MODE1/2 settings ASoC: rsnd: add missing pin sharing with SSI9 ASoC: soc-core: don't use soc_find_component() at snd_soc_find_dai() ASoC: atmel: atmel-pcm-dma.c: use devm_snd_dmaengine_pcm_register() ASoC: vc4: vc4_htmi: consider CPU-Platform possibility ASoC: atmel: atmel-classd: consider CPU-Platform possibility ASoC: atmel: atmel-pdmic: consider CPU-Platform possibility ASoC: atmel: atmel_wm8904: consider CPU-Platform possibility unnecessary Platform" ASoC: amtel: mikroe-proto: consider CPU-Platform possibility ASoC: atmel: sam9g20_wm8731: consider CPU-Platform possibility ASoC: atmel: sam9x5_wm8731: consider CPU-Platform possibility ASoC: atmel: tse850-pcm5142: consider CPU-Platform possibility ASoC: fsl: eukrea-tlv320: consider CPU-Platform possibility ASoC: fsl: fsl-asoc-card: consider CPU-Platform possibility ASoC: fsl: imx-es8328: consider CPU-Platform possibility ASoC: fsl: imx-sgtl5000: consider CPU-Platform possibility ASoC: fsl: imx-spdif: consider CPU-Platform possibility ASoC: fsl: imx-audmix: consider CPU-Platform possibility ASoC: kirkwood: armada-370-db: consider CPU-Platform possibility ASoC: mxs: mxs-sgtl5000: consider CPU-Platform possibility ASoC: qcom: apq8016_sbc: consider CPU-Platform possibility ASoC: qcom: storm: consider CPU-Platform possibility ASoC: rockchip: rk3288_hdmi_analog: consider CPU-Platform possibility ASoC: rockchip: rockchip_max98090: consider CPU-Platform possibility ASoC: rockchip: rockchip_rt5645: consider CPU-Platform possibility ASoC: samsung: arndale_rt5631: consider CPU-Platform possibility ASoC: samsung: smdk_wm8994: consider CPU-Platform possibility ASoC: samsung: snow: consider CPU-Platform possibility ASoC: samsung: tm2_wm5110: consider CPU-Platform possibility ASoC: sirf: sirf-audio: consider CPU-Platform possibility ASoC: sunxi: sun4i-codec: consider CPU-Platform possibility ASoC: tegra: tegra_alc5632: consider CPU-Platform possibility ASoC: tegra: tegra_max98090: consider CPU-Platform possibility ASoC: tegra: tegra_rt5640: consider CPU-Platform possibility ASoC: tegra: tegra_rt5677: consider CPU-Platform possibility ASoC: tegra: tegra_sgtl5000: consider CPU-Platform possibility ASoC: tegra: tegra_wm8753: consider CPU-Platform possibility ASoC: tegra: tegra_wm8903: consider CPU-Platform possibility ASoC: tegra: tegra_wm9712: consider CPU-Platform possibility ASoC: tegra: trimslice: consider CPU-Platform possibility ASoC: ti: davinci-evm: consider CPU-Platform possibility ASoC: ti: omap-abe-twl6040: consider CPU-Platform possibility ASoC: ti: omap-hdmi: consider CPU-Platform possibility ASoC: ti: omap-twl4030: consider CPU-Platform possibility ASoC: ti: rx51: consider CPU-Platform possibility ASoC: ux500: mop500: consider CPU-Platform possibility ASoC: simple-card-utils: consider CPU-Platform possibility ASoC: qcom: consider CPU-Platform possibility ASoC: rockchip: rk3399_gru_sound: consider CPU-Platform possibility ASoC: soc-utils: respawn dummy Platform ASoC: au1x: psc-i2s.c: use devm_snd_soc_register_component() ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() ASoC: rt5682: use devm_snd_soc_register_component() ASoC: ak4118: use devm_snd_soc_register_component() ASoC: cros_ec_codec: use devm_snd_soc_register_component() ASoC: pxa: pxa2xx-ac97.c: use devm_snd_soc_register_component() Luca Weiss (1): ASoC: sunxi: sun50i-codec-analog: Add earpiece Mac Chiang (2): ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function ASoC: max98357a: avoid speaker pop when playback startup Mark Brown (11): Merge tag 'v5.2-rc1' into asoc-5.3 Merge branch 'for-5.2' of https://git.kernel.org/.../broonie/sound into asoc-5.3 Merge branch 'asoc-5.2' into asoc-5.3 Merge branch 'asoc-5.2' into asoc-5.3 Merge branch 'asoc-5.2' into asoc-5.3 ASoC: core: Adapt for debugfs API change ASoC: dapm: Adapt for debugfs API change Merge tag 'v5.2-rc6' into asoc-5.3 Merge branch 'asoc-5.2' into asoc-linus Merge branch 'asoc-5.3' into asoc-next Merge remote-tracking branch 'asoc/topic/meson' into asoc-next Mauro Carvalho Chehab (1): ASoC: dt-bindings: fix some broken links from txt->yaml conversion Maxime Jourdan (1): ASoC: max98357a: Show KConfig entry Maxime Ripard (5): dt-bindings: sound: Convert Allwinner SPDIF binding to YAML dt-bindings: sound: sun4i-spdif: Document that the RX channel can be missing dt-bindings: sound: Convert Allwinner I2S binding to YAML dt-bindings: sound: sun4i-i2s: Document that the RX channel can be missing ASoC: sun4i-i2s: Change SR and WSS computation Nathan Chancellor (1): ASoC: rt1011: Mark format integer literals as unsigned Nathan Huckleberry (1): ASoC: tas571x: Fix -Wunused-const-variable Nilkanth Ahirrao (1): ASoC: rsnd: fixup mod ID calculation in rsnd_ctu_probe_ Olivier Moysan (4): ASoC: stm32: spdifrx: manage identification registers ASoC: stm32: i2s: manage identification registers ASoC: stm32: sai: manage identification registers ASoC: stm32: dfsdm: add 16 bits audio record support Pan Xiuli (1): ASoC: SOF: soundwire: add initial soundwire support Paul Gortmaker (1): ASoC: soc-acpi: fix implicit header use of module.h/export.h Paweł Harłoziński (1): ASoC: Intel: Skylake: Use recommended SDxFMT programming sequence Peter Ujfalusi (7): ASoC: pcm3168a: Implement set_tdm_slot callback bindings: sound: davinci-mcasp: Add support for optional auxclk-fs-ratio ASoC: ti: davinci-mcasp: Support for auxclk-fs-ratio ASoC: ti: davinci-mcasp: Set unused serializers as INACTIVE ASoC: ti: davinci-mcasp: Fix slot mask settings when using multiple AXRs ASoC: pcm3168a: Rename min_frame_size to slot_width ASoC: pcm3168a: Add support for multi DIN/DOUT with TDM slots parameter Pierre-Louis Bossart (15): ASoC: Intel: boards: remove unnecessary inclusion of skl.h ASoC: Intel: sof-rt5682: fix undefined references with Baytrail-only support ASoC: Intel: use common helpers to detect CPUs ASoC: Intel: boards: remove dependency on asm/platform_sst_audio.h ASoC: Intel: boards: Add COMPILE_TEST for new machine drivers ASoC: SOF: add COMPILE_TEST for PCI options ASoC: SOF: Intel: cnl-ipc: read all IPC registers first ASoC: SOF: Intel: cnl-ipc: move code around for clarity ASoC: SOF: Intel: cnl-ipc: re-enable IPC IRQ at end of handler ASoC: SOF: Intel: hda-ipc: read all IPC registers first ASoC: SOF: Intel: hda-ipc: re-enable IPC IRQ at end of handler ASoC: SOF: Intel: ipc: don't check for HIPCCTL register value ASoC: soc-topology: fix modern dai link style ASoC: SOF: uapi: align comments with firmware files ASoC: SOF: ipc: clarify operator precedence Randy Dunlap (1): ASoc: fix sound/soc/intel/skylake/slk-ssp-clk.c build error on IA64 Ranjani Sridharan (13): ASoC: SOF: pcm: remove runtime PM calls during pcm open/close ASoC: SOF: add mode parameter for snd_sof_debugfs_buf_item ASoC: SOF: ipc: Introduce SOF_IPC_GLB_TEST_MSG IPC command ASoC: SOF: debug: add new debugfs entries for IPC flood test ASoC: SOF: Intel: hda: save handle to sdev in sof_intel_hda_stream ASoC: SOF: Intel: hda: add new macro hstream_to_sof_hda_stream() ASoC: SOF: topology: add cpu_dai_name for DAIs ASoC: SOF: Intel: hda: assign link DMA channel at run-time ASoC: SOF: Intel: hda: reserve host DMA channel for hostless streams ASoC: SOF: Intel: hda: release link DMA for paused streams during suspend ASoC: SOF: Intel: hda: couple host and link DMA during FE hw_free ASoC: SOF: Intel: hda: modify stream interrupt handler ASoC: SOF: Intel: hda: clear stream status and wakests properly Ravulapati Vishnu vardhan rao (1): ASoC: amd: Reporting accurate hw_ptr for acp3x dma Richard Fitzgerald (5): ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs ASoC: madera: Add common support for Cirrus Logic Madera codecs ASoC: cs47l35: Add codec driver for Cirrus Logic CS47L35 ASoC: cs47l85: Add codec driver for Cirrus Logic CS47L85 ASoC: cs47l90: Add codec driver for Cirrus Logic CS47L90 S.j. Wang (5): ASoC: fsl_esai: Add pm runtime function ASoC: fsl_asrc: replace the process_option table with function ASoC: cs42xx8: Add reset gpio handling ASoC: cs42xx8: add reset-gpios in binding document ASoC: fsl_asrc: Unify the supported input and output rate Saravanan Sekar (1): ASoC: tlv320aic3x: Add support for high power analog output Sathya Prakash M R (1): ASoC: Intel: sof-rt5682: add newline for clarity Seppo Ingalsuo (1): ASoC: SOF: Add DMIC token for unmute gain ramp time Shengjiu Wang (2): ASoC: fsl_sai: Move clock operation to PM runtime ASoC: cs42xx8: Fix build error with CONFIG_GPIOLIB is not set Shuming Fan (2): ASoC: rt1011: Add RT1011 amplifier driver ASoC: rt1011: fix warning reported by kbuild test robot and minor issue Shunli Wang (3): ASoC: Mediatek: add memory interface data align ASoC: Mediatek: MT8183: set data align ASoC: Mediatek: MT8183: enable IIR filter Simon Ho (1): ASoC: Add support for Conexant CX2072X CODEC Srinivas Kandagatla (4): ASoC: msm8916-wcd-digital: Add sidetone support ASoC: core: Return -ENOTSUPP from set_channel_map() if no operation provided ASoC: qdsp6: q6afe-dai: Add missing Slimbus0 audio route ASoC: wcd9335: remove multiple defines. Sylwester Nawrocki (1): ASoC: Remove erroneous soc_cleanup_card_resources() call Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms Tzung-Bi Shih (5): ASoC: max98357a: request GPIO when device get probed ASoC: mediatek: mt8183: remove unused DAPM pins ASoC: mediatek: mt8183: move card registration to the end of probe ASoC: core: use component driver name as component name ASoC: Intel: sof_rt5682: use GFP_KERNEL instead of GFP_ATOMIC Wei Yongjun (1): ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write() Wen Yang (1): ASoC: audio-graph-card: fix use-after-free in graph_for_each_link Xun Zhang (1): ASoC: Intel: sof-rt5682: add MCLK support for BYT platform Yong Zhi (1): ASoC: Intel: glk_rt5682_max98357a: Remap button control-function YueHaibing (5): ASoC: cx2072x: remove set but not used variable 'is_right_j ' ASoC: mediatek: Make some symbols static ASoC: SOF: Intel: hda: remove duplicated include from hda.c ASoC: madera: Remove duplicated include from cs47l35.c ASoC: rt1308: Fix platform_no_drv_owner.cocci warnings Zhu Yingjiang (10): ASoC: SOF: Intel: ICL: add Icelake SSP count ASoC: SOF: Intel: ICL add Icelake chip info struct ASoC: SOF: Intel: hda: use the defined ppcap functions ASoC: SOF: topology: add min/max step for volume_table ASoC: SOF: topology: pass volume min/max linear value to FW ASoC: SOF: Intel: hda: use the SOF defined ppcap functions ASoC: SOF: Intel: hda: add function for hda stop chip ASoC: SOF: Intel: hda: use the defined stop chip in suspend ASoC: SOF: Intel: hda: make sure DMA is start/stop by read the RUN bit ASoC: SOF: Intel: hda: make sure RUN bit setting to 0 during clear stream status .../devicetree/bindings/media/st,stm32-dcmi.txt | 2 +- .../bindings/sound/allwinner,sun4i-a10-i2s.yaml | 132 + .../bindings/sound/allwinner,sun4i-a10-spdif.yaml | 120 + .../bindings/sound/amlogic,axg-tdm-formatters.txt | 6 +- .../bindings/sound/amlogic,g12a-tohdmitx.txt | 55 + .../devicetree/bindings/sound/cs42xx8.txt | 6 + .../bindings/sound/davinci-mcasp-audio.txt | 3 + Documentation/devicetree/bindings/sound/madera.txt | 67 + .../devicetree/bindings/sound/max98357a.txt | 4 + Documentation/devicetree/bindings/sound/rt1011.txt | 32 + Documentation/devicetree/bindings/sound/rt1308.txt | 17 + .../devicetree/bindings/sound/st,stm32-i2s.txt | 2 +- .../devicetree/bindings/sound/st,stm32-sai.txt | 2 +- .../devicetree/bindings/sound/sun4i-i2s.txt | 45 - .../bindings/sound/sunxi,sun4i-spdif.txt | 42 - MAINTAINERS | 7 +- drivers/gpu/drm/vc4/vc4_hdmi.c | 19 +- include/dt-bindings/sound/madera.h | 25 + include/dt-bindings/sound/meson-g12a-tohdmitx.h | 13 + include/linux/mfd/madera/pdata.h | 4 + include/sound/hda_codec.h | 3 + include/sound/madera-pdata.h | 59 + include/sound/simple_card_utils.h | 20 +- include/sound/soc.h | 165 +- include/sound/sof/dai-intel.h | 3 +- include/sound/sof/header.h | 7 +- include/sound/sof/topology.h | 7 +- include/sound/sof/trace.h | 14 +- include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/eq.h | 172 - include/uapi/sound/sof/manifest.h | 188 - include/uapi/sound/sof/tokens.h | 1 + include/uapi/sound/sof/tone.h | 21 - include/uapi/sound/sof/trace.h | 66 - sound/hda/ext/hdac_ext_bus.c | 8 +- sound/soc/amd/acp-da7219-max98357a.c | 42 +- sound/soc/amd/acp-rt5645.c | 21 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 43 +- sound/soc/atmel/atmel-classd.c | 21 +- sound/soc/atmel/atmel-pcm-dma.c | 9 +- sound/soc/atmel/atmel-pcm-pdc.c | 5 - sound/soc/atmel/atmel-pcm.h | 8 - sound/soc/atmel/atmel-pdmic.c | 21 +- sound/soc/atmel/atmel_ssc_dai.c | 11 - sound/soc/atmel/atmel_wm8904.c | 17 +- sound/soc/atmel/mikroe-proto.c | 21 +- sound/soc/atmel/sam9g20_wm8731.c | 22 +- sound/soc/atmel/sam9x5_wm8731.c | 20 +- sound/soc/atmel/tse850-pcm5142.c | 13 +- sound/soc/au1x/db1000.c | 10 +- sound/soc/au1x/db1200.c | 50 +- sound/soc/au1x/psc-i2s.c | 6 +- sound/soc/cirrus/edb93xx.c | 10 +- sound/soc/cirrus/ep93xx-i2s.c | 7 +- sound/soc/cirrus/simone.c | 10 +- sound/soc/cirrus/snappercl15.c | 11 +- sound/soc/codecs/Kconfig | 45 +- sound/soc/codecs/Makefile | 14 + sound/soc/codecs/ad193x.c | 68 +- sound/soc/codecs/ak4118.c | 11 +- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/cs42xx8.c | 13 + sound/soc/codecs/cs47l35.c | 1777 +++++++++ sound/soc/codecs/cs47l85.c | 2730 +++++++++++++ sound/soc/codecs/cs47l90.c | 2653 +++++++++++++ sound/soc/codecs/cx2072x.c | 1725 ++++++++ sound/soc/codecs/cx2072x.h | 314 ++ sound/soc/codecs/hdac_hdmi.c | 39 + sound/soc/codecs/hdmi-codec.c | 188 +- sound/soc/codecs/madera.c | 4177 ++++++++++++++++++++ sound/soc/codecs/madera.h | 442 +++ sound/soc/codecs/max98357a.c | 74 +- sound/soc/codecs/msm8916-wcd-digital.c | 282 ++ sound/soc/codecs/nau8822.c | 18 + sound/soc/codecs/nau8822.h | 7 + sound/soc/codecs/nau8825.c | 4 + sound/soc/codecs/nau8825.h | 2 + sound/soc/codecs/pcm3168a.c | 91 +- sound/soc/codecs/rt1011.c | 2244 +++++++++++ sound/soc/codecs/rt1011.h | 672 ++++ sound/soc/codecs/rt1308.c | 898 +++++ sound/soc/codecs/rt1308.h | 291 ++ sound/soc/codecs/rt5514-spi.c | 4 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5677-spi.c | 4 +- sound/soc/codecs/rt5677.c | 336 +- sound/soc/codecs/rt5677.h | 46 +- sound/soc/codecs/rt5682.c | 13 +- sound/soc/codecs/tas571x.c | 4 +- sound/soc/codecs/tlv320aic3x.c | 14 +- sound/soc/codecs/wcd9335.c | 7 +- sound/soc/codecs/wm_adsp.c | 37 +- sound/soc/fsl/efika-audio-fabric.c | 22 +- sound/soc/fsl/eukrea-tlv320.c | 19 +- sound/soc/fsl/fsl-asoc-card.c | 47 +- sound/soc/fsl/fsl_asrc.c | 103 +- sound/soc/fsl/fsl_esai.c | 141 +- sound/soc/fsl/fsl_sai.c | 54 +- sound/soc/fsl/fsl_ssi.c | 4 +- sound/soc/fsl/fsl_ssi.h | 8 +- sound/soc/fsl/fsl_ssi_dbg.c | 18 +- sound/soc/fsl/fsl_utils.c | 2 +- sound/soc/fsl/imx-audmix.c | 45 +- sound/soc/fsl/imx-audmux.c | 10 +- sound/soc/fsl/imx-es8328.c | 23 +- sound/soc/fsl/imx-mc13783.c | 10 +- sound/soc/fsl/imx-sgtl5000.c | 23 +- sound/soc/fsl/imx-spdif.c | 20 +- sound/soc/fsl/mpc8610_hpcd.c | 33 +- sound/soc/fsl/mx27vis-aic32x4.c | 11 +- sound/soc/fsl/p1022_ds.c | 36 +- sound/soc/fsl/p1022_rdk.c | 35 +- sound/soc/fsl/pcm030-audio-fabric.c | 20 +- sound/soc/fsl/phycore-ac97.c | 10 +- sound/soc/fsl/wm1133-ev1.c | 10 +- sound/soc/generic/audio-graph-card.c | 36 +- sound/soc/generic/simple-card-utils.c | 21 +- sound/soc/generic/simple-card.c | 47 +- sound/soc/intel/Kconfig | 25 + sound/soc/intel/atom/sst/sst_acpi.c | 65 +- sound/soc/intel/boards/Kconfig | 18 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5677.c | 22 +- sound/soc/intel/boards/broadwell.c | 46 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 163 +- sound/soc/intel/boards/bxt_rt298.c | 135 +- sound/soc/intel/boards/byt-max98090.c | 10 +- sound/soc/intel/boards/byt-rt5640.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 270 ++ sound/soc/intel/boards/bytcht_da7213.c | 38 +- sound/soc/intel/boards/bytcht_es8316.c | 51 +- sound/soc/intel/boards/bytcht_nocodec.c | 30 +- sound/soc/intel/boards/bytcr_rt5640.c | 60 +- sound/soc/intel/boards/bytcr_rt5651.c | 58 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 41 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 59 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 39 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 145 +- sound/soc/intel/boards/haswell.c | 46 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 115 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 286 +- sound/soc/intel/boards/kbl_rt5660.c | 83 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 184 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 126 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 67 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 120 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 132 +- sound/soc/intel/boards/skl_rt286.c | 118 +- sound/soc/intel/boards/sof_rt5682.c | 165 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + sound/soc/intel/common/soc-intel-quirks.h | 115 + sound/soc/intel/common/sst-ipc.c | 2 +- sound/soc/intel/skylake/cnl-sst.c | 2 +- sound/soc/intel/skylake/skl-debug.c | 9 + sound/soc/intel/skylake/skl-messages.c | 39 +- sound/soc/intel/skylake/skl-pcm.c | 44 +- sound/soc/intel/skylake/skl-ssp-clk.c | 16 +- sound/soc/intel/skylake/skl-sst-dsp.h | 6 +- sound/soc/intel/skylake/skl-sst-ipc.c | 4 +- sound/soc/intel/skylake/skl-sst-utils.c | 23 +- sound/soc/intel/skylake/skl-sst.c | 4 +- sound/soc/intel/skylake/skl-topology.c | 72 +- sound/soc/intel/skylake/skl-topology.h | 8 +- sound/soc/intel/skylake/skl.c | 38 +- sound/soc/intel/skylake/skl.h | 6 + sound/soc/jz4740/qi_lb60.c | 10 +- sound/soc/kirkwood/armada-370-db.c | 42 +- sound/soc/mediatek/common/Makefile | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 6 + sound/soc/mediatek/common/mtk-base-afe.h | 2 + sound/soc/mediatek/common/mtk-btcvsd.c | 22 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 95 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 34 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 115 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 34 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 51 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 85 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 74 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 23 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 197 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 14 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 193 +- sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 87 +- sound/soc/meson/axg-tdm-formatter.c | 29 + sound/soc/meson/axg-tdm-interface.c | 4 +- sound/soc/meson/axg-tdm.h | 2 +- sound/soc/meson/axg-tdmin.c | 1 - sound/soc/meson/axg-tdmout.c | 1 - sound/soc/meson/g12a-tohdmitx.c | 413 ++ sound/soc/mxs/mxs-sgtl5000.c | 27 +- sound/soc/nuc900/nuc900-audio.c | 10 +- sound/soc/pxa/brownstone.c | 10 +- sound/soc/pxa/corgi.c | 10 +- sound/soc/pxa/e740_wm9705.c | 20 +- sound/soc/pxa/e750_wm9705.c | 20 +- sound/soc/pxa/e800_wm9712.c | 21 +- sound/soc/pxa/em-x270.c | 20 +- sound/soc/pxa/hx4700.c | 10 +- sound/soc/pxa/imote2.c | 11 +- sound/soc/pxa/magician.c | 22 +- sound/soc/pxa/mioa701_wm9713.c | 20 +- sound/soc/pxa/palm27x.c | 20 +- sound/soc/pxa/poodle.c | 10 +- sound/soc/pxa/pxa2xx-ac97.c | 3 +- sound/soc/pxa/spitz.c | 10 +- sound/soc/pxa/tosa.c | 20 +- sound/soc/pxa/ttc-dkb.c | 10 +- sound/soc/pxa/z2.c | 10 +- sound/soc/pxa/zylonite.c | 30 +- sound/soc/qcom/apq8016_sbc.c | 19 +- sound/soc/qcom/common.c | 34 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 1 + sound/soc/qcom/qdsp6/q6core.c | 16 +- sound/soc/qcom/storm.c | 17 +- sound/soc/rockchip/Kconfig | 1 + sound/soc/rockchip/rk3288_hdmi_analog.c | 21 +- sound/soc/rockchip/rk3399_gru_sound.c | 65 +- sound/soc/rockchip/rockchip_max98090.c | 17 +- sound/soc/rockchip/rockchip_rt5645.c | 33 +- sound/soc/samsung/arndale_rt5631.c | 25 +- sound/soc/samsung/bells.c | 87 +- sound/soc/samsung/h1940_uda1380.c | 10 +- sound/soc/samsung/jive_wm8750.c | 10 +- sound/soc/samsung/littlemill.c | 19 +- sound/soc/samsung/lowland.c | 26 +- sound/soc/samsung/neo1973_wm8753.c | 18 +- sound/soc/samsung/odroid.c | 29 +- sound/soc/samsung/rx1950_uda1380.c | 11 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 11 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 11 +- sound/soc/samsung/s3c24xx_uda134x.c | 10 +- sound/soc/samsung/smartq_wm8987.c | 10 +- sound/soc/samsung/smdk_spdif.c | 10 +- sound/soc/samsung/smdk_wm8580.c | 20 +- sound/soc/samsung/smdk_wm8994.c | 30 +- sound/soc/samsung/smdk_wm8994pcm.c | 10 +- sound/soc/samsung/snow.c | 40 +- sound/soc/samsung/speyside.c | 26 +- sound/soc/samsung/tm2_wm5110.c | 42 +- sound/soc/samsung/tobermory.c | 10 +- sound/soc/sh/migor.c | 10 +- sound/soc/sh/rcar/adg.c | 1 + sound/soc/sh/rcar/core.c | 120 +- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/ssi.c | 1 + sound/soc/sh/rcar/ssiu.c | 92 +- sound/soc/sh/sh7760-ac97.c | 11 +- sound/soc/sirf/sirf-audio.c | 13 +- sound/soc/soc-acpi.c | 2 + sound/soc/soc-compress.c | 17 +- sound/soc/soc-core.c | 382 +- sound/soc/soc-dapm.c | 23 +- sound/soc/soc-pcm.c | 88 +- sound/soc/soc-topology.c | 146 +- sound/soc/sof/Kconfig | 8 + sound/soc/sof/control.c | 270 +- sound/soc/sof/debug.c | 247 +- sound/soc/sof/intel/Kconfig | 32 + sound/soc/sof/intel/apl.c | 2 + sound/soc/sof/intel/byt.c | 5 +- sound/soc/sof/intel/cnl.c | 79 +- sound/soc/sof/intel/hda-ctrl.c | 77 +- sound/soc/sof/intel/hda-dai.c | 293 +- sound/soc/sof/intel/hda-dsp.c | 63 +- sound/soc/sof/intel/hda-ipc.c | 43 +- sound/soc/sof/intel/hda-stream.c | 157 +- sound/soc/sof/intel/hda.c | 30 +- sound/soc/sof/intel/hda.h | 22 +- sound/soc/sof/ipc.c | 18 +- sound/soc/sof/loader.c | 10 +- sound/soc/sof/nocodec.c | 21 +- sound/soc/sof/ops.h | 24 +- sound/soc/sof/pcm.c | 36 +- sound/soc/sof/pm.c | 43 +- sound/soc/sof/sof-acpi-dev.c | 59 +- sound/soc/sof/sof-pci-dev.c | 32 +- sound/soc/sof/sof-priv.h | 23 +- sound/soc/sof/topology.c | 156 +- sound/soc/sof/trace.c | 66 +- sound/soc/stm/stm32_adfsdm.c | 49 +- sound/soc/stm/stm32_i2s.c | 60 +- sound/soc/stm/stm32_sai.c | 44 +- sound/soc/stm/stm32_sai.h | 54 +- sound/soc/stm/stm32_sai_sub.c | 14 +- sound/soc/stm/stm32_spdifrx.c | 37 +- sound/soc/sunxi/sun4i-codec.c | 20 +- sound/soc/sunxi/sun4i-i2s.c | 71 +- sound/soc/sunxi/sun4i-spdif.c | 49 +- sound/soc/sunxi/sun50i-codec-analog.c | 50 + sound/soc/tegra/Makefile | 2 +- sound/soc/tegra/tegra_alc5632.c | 37 +- sound/soc/tegra/tegra_max98090.c | 17 +- sound/soc/tegra/tegra_rt5640.c | 17 +- sound/soc/tegra/tegra_rt5677.c | 37 +- sound/soc/tegra/tegra_sgtl5000.c | 37 +- sound/soc/tegra/tegra_wm8753.c | 17 +- sound/soc/tegra/tegra_wm8903.c | 17 +- sound/soc/tegra/tegra_wm9712.c | 14 +- sound/soc/tegra/trimslice.c | 19 +- sound/soc/ti/ams-delta.c | 10 +- sound/soc/ti/davinci-evm.c | 105 +- sound/soc/ti/davinci-mcasp.c | 81 +- sound/soc/ti/n810.c | 11 +- sound/soc/ti/omap-abe-twl6040.c | 36 +- sound/soc/ti/omap-hdmi.c | 20 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/ti/omap-twl4030.c | 36 +- sound/soc/ti/omap3pandora.c | 20 +- sound/soc/ti/osk5912.c | 11 +- sound/soc/ti/rx51.c | 23 +- sound/soc/txx9/txx9aclc-generic.c | 10 +- sound/soc/ux500/mop500.c | 36 +- sound/usb/bcd2000/Makefile | 2 +- 318 files changed, 26455 insertions(+), 4825 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 Documentation/devicetree/bindings/sound/rt1011.txt create mode 100755 Documentation/devicetree/bindings/sound/rt1308.txt delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 include/dt-bindings/sound/madera.h create mode 100644 include/dt-bindings/sound/meson-g12a-tohdmitx.h create mode 100644 include/sound/madera-pdata.h delete mode 100644 include/uapi/sound/sof/eq.h delete mode 100644 include/uapi/sound/sof/manifest.h delete mode 100644 include/uapi/sound/sof/tone.h delete mode 100644 include/uapi/sound/sof/trace.h create mode 100644 sound/soc/codecs/cs47l35.c create mode 100644 sound/soc/codecs/cs47l85.c create mode 100644 sound/soc/codecs/cs47l90.c create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/codecs/madera.c create mode 100644 sound/soc/codecs/madera.h create mode 100644 sound/soc/codecs/rt1011.c create mode 100644 sound/soc/codecs/rt1011.h create mode 100755 sound/soc/codecs/rt1308.c create mode 100755 sound/soc/codecs/rt1308.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c create mode 100644 sound/soc/intel/common/soc-intel-quirks.h create mode 100644 sound/soc/meson/g12a-tohdmitx.c

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: fix Line6 Helix audio format rates
by Nicola Lunghi 07 Jul '19

07 Jul '19

Line6 Helix and HX stomp devices don't support retrieving the number of clock sample rate. Add a quirk to set it to 48Khz by default. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 46 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 43 insertions(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index c02b51a82775..31780dfb8eb9 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -285,6 +285,33 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, return nr_rates; } +/* Line6 Helix series don't support the UAC2_CS_RANGE usb function + * call. Return a static table of known clock rates. + */ +static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, + struct audioformat *fp) +{ + switch (chip->usb_id) { + case USB_ID(0x0E41, 0x4241): /* Line6 Helix */ + case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ + case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ + case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + /* supported rates: 48Khz */ + kfree(fp->rate_table); + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + if (!fp->rate_table) + return -ENOMEM; + fp->nr_rates = 1; + fp->rate_min = 48000; + fp->rate_max = 48000; + fp->rates = SNDRV_PCM_RATE_48000; + fp->rate_table[0] = 48000; + return 0; + } + + return -ENODEV; +} + /* * parse the format descriptor and stores the possible sample rates * on the audioformat table (audio class v2 and v3). @@ -294,7 +321,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, { struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; - int nr_triplets, data_size, ret = 0; + int nr_triplets, data_size, ret, ret_l6 = 0; int clock = snd_usb_clock_find_source(chip, fp->protocol, fp->clock, false); @@ -313,9 +340,22 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, tmp, sizeof(tmp)); if (ret < 0) { - dev_err(&dev->dev, - "%s(): unable to retrieve number of sample rates (clock %d)\n", + /* line6 helix devices don't support UAC2_CS_CONTROL_SAM_FREQ call */ + ret_l6 = line6_parse_audio_format_rates_quirk(chip, fp); + if (ret_l6 == -ENODEV) { + /* no line6 device found continue showing the error */ + dev_err(&dev->dev, + "%s(): unable to retrieve number of sample rates (clock %d)\n", + __func__, clock); + goto err; + } + if (ret_l6 == 0) { + dev_info(&dev->dev, + "%s(): unable to retrieve number of sample rates: set it to a predefined value (clock %d).\n", __func__, clock); + return 0; + } + ret = ret_l6; goto err; } -- 2.19.1

2 1

[alsa-devel] [PATCH 0/6] ALSA: firewire: fix wrong reference count for stream functionality at error path of rawmidi interface
by Takashi Sakamoto 07 Jul '19

07 Jul '19

Hi, All of drivers in ALSA firewire stack maintain packet streaming by reference count. However at error path of MIDI interface, some drivers have a bug that the count remains at error path of rawmidi interface. This causes the drivers don't stop packet streaming even if it's not required. This patchset fixes drivers including the bug. Takashi Sakamoto (6): ALSA: bebob: fix wrong reference count for stream functionality at error path of rawmidi interface ALSA: fireworks: fix wrong reference count for stream functionality at error path of rawmidi interface ALSA: oxfw: fix wrong reference count for stream functionality at error path of rawmidi interface ALSA: dice: fix wrong reference count for stream functionality at error path of rawmidi interface ALSA: firewire-digi00x: fix wrong reference count for stream functionality at error path of rawmidi interface firewire-motu: fix wrong reference count for stream functionality at error path of rawmidi interface sound/firewire/bebob/bebob_midi.c | 2 ++ sound/firewire/dice/dice-midi.c | 2 ++ sound/firewire/digi00x/digi00x-midi.c | 2 ++ sound/firewire/fireworks/fireworks_midi.c | 2 ++ sound/firewire/motu/motu-midi.c | 2 ++ sound/firewire/oxfw/oxfw-midi.c | 2 ++ 6 files changed, 12 insertions(+) -- 2.20.1

2 7

[alsa-devel] [PATCH v2 resend 0/3] ASoC: SOF: fix suspend ordering with runtime idle
by Kai Vehmanen 06 Jul '19

06 Jul '19

Hi all, this series addresses issues with ordering of HDA codec and controller suspends in the runtime PM flows. The implemented logic for SOF is similar to what has been used by the Intel AZX HDA driver. To implement this, first a fix is needed to hdac_hdmic ASoC codec driver. SOF framework also needs to be extended to allow SOF devices to implement a runtime_idle callback. Third, concrete implementation is in a separate patch for APL/CNL Intel hardware, for which strict ordering of codec-controller power down sequence needs to be maintained. As this extends the SOF device interface, Pierre asked me to send to the list for wider review. This series has been prereviewd at SOF github as: https://github.com/thesofproject/linux/pull/1003 v2 resend: - no changes since v2 -- resending as 2 weeks has passed and multiple reviewed-by acks already received v2: - Rebased on broonie/for-next, there was a conflict with patch "ASoC: SOF: Intel: hda: release link DMA for paused streams during suspend" - Added Reviewed-by tags from v1 round Kai Vehmanen (3): ASoC: hdac_hdmi: report codec link up/down status to bus ASoC: SOF: add runtime idle callback ASoC: SOF: Intel: implement runtime idle for CNL/APL sound/soc/codecs/hdac_hdmi.c | 2 ++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/hda-dsp.c | 13 +++++++++++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/ops.h | 8 ++++++++ sound/soc/sof/pm.c | 8 ++++++++ sound/soc/sof/sof-acpi-dev.c | 2 +- sound/soc/sof/sof-pci-dev.c | 2 +- sound/soc/sof/sof-priv.h | 2 ++ 10 files changed, 38 insertions(+), 2 deletions(-) -- 2.17.1

2 6

[alsa-devel] [PATCH v2] ASoC: sunxi: sun50i-codec-analog: Add earpiece
by Luca Weiss 05 Jul '19

05 Jul '19

This adds the necessary registers and audio routes to play audio using the Earpiece, that's supported on the A64. Signed-off-by: Luca Weiss <luca(a)z3ntu.xyz> --- Changes v1 -> v2: * Make the earpiece enable register a DAPM widget * Adjust the audio routes to include the new Earpiece Amp widget * Left/Right Analog Mixer => Left/Right Mixer sound/soc/sunxi/sun50i-codec-analog.c | 50 +++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index d105c90c3706..6d1de565350e 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -49,6 +49,15 @@ #define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 #define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 +#define SUN50I_ADDA_EARPIECE_CTRL0 0x03 +#define SUN50I_ADDA_EARPIECE_CTRL0_EAR_RAMP_TIME 4 +#define SUN50I_ADDA_EARPIECE_CTRL0_ESPSR 0 + +#define SUN50I_ADDA_EARPIECE_CTRL1 0x04 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN 7 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE 6 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL 0 + #define SUN50I_ADDA_LINEOUT_CTRL0 0x05 #define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 #define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 @@ -172,6 +181,10 @@ static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), ); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_earpiece_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); /* volume / mute controls */ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { @@ -225,6 +238,15 @@ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { SUN50I_ADDA_LINEOUT_CTRL0_LEN, SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + SOC_SINGLE_TLV("Earpiece Playback Volume", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL, 0x1f, 0, + sun50i_codec_earpiece_vol_scale), + + SOC_SINGLE("Earpiece Playback Switch", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE, 1, 0), + }; static const char * const sun50i_codec_hp_src_enum_text[] = { @@ -257,6 +279,20 @@ static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { sun50i_codec_lineout_src_enum), }; +static const char * const sun50i_codec_earpiece_src_enum_text[] = { + "DACR", "DACL", "Right Mixer", "Left Mixer", +}; + +static SOC_ENUM_SINGLE_DECL(sun50i_codec_earpiece_src_enum, + SUN50I_ADDA_EARPIECE_CTRL0, + SUN50I_ADDA_EARPIECE_CTRL0_ESPSR, + sun50i_codec_earpiece_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_earpiece_src[] = { + SOC_DAPM_ENUM("Earpiece Source Playback Route", + sun50i_codec_earpiece_src_enum), +}; + static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { /* DAC */ SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, @@ -285,6 +321,12 @@ static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_MUX("Earpiece Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_earpiece_src), + SND_SOC_DAPM_OUT_DRV("Earpiece Amp", SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("EARPIECE"), + /* Microphone inputs */ SND_SOC_DAPM_INPUT("MIC1"), @@ -388,6 +430,14 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, + + /* Earpiece Routes */ + { "Earpiece Source Playback Route", "DACL", "Left DAC" }, + { "Earpiece Source Playback Route", "DACR", "Right DAC" }, + { "Earpiece Source Playback Route", "Left Mixer", "Left Mixer" }, + { "Earpiece Source Playback Route", "Right Mixer", "Right Mixer" }, + { "Earpiece Amp", NULL, "Earpiece Source Playback Route" }, + { "EARPIECE", NULL, "Earpiece Amp" }, }; static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { -- 2.22.0

3 2

[alsa-devel] [PATCH] ASoC: rt5665: remove redundant assignment to variable idx
by Colin King 05 Jul '19

05 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> The variable idx is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 87263317085a..c050d84a6916 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1478,7 +1478,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); - int pd, idx = -EINVAL; + int pd, idx; pd = rl6231_get_pre_div(rt5665->regmap, RT5665_ADDA_CLK_1, RT5665_I2S_PD1_SFT); -- 2.20.1

2 1

[alsa-devel] [PATCH -next] ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write()
by Wei Yongjun 05 Jul '19

05 Jul '19

'string' is malloced in sof_dfsentry_write() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Fixes: 091c12e1f50c ("ASoC: SOF: debug: add new debugfs entries for IPC flood test") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sof/debug.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 54bb53bfc81b..2388477a965e 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -162,7 +162,7 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, else ret = kstrtoul(string, 0, &ipc_count); if (ret < 0) - return ret; + goto out; /* limit max duration/ipc count for flood test */ if (flood_duration_test) { @@ -191,7 +191,7 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, "error: debugfs write failed to resume %d\n", ret); pm_runtime_put_noidle(sdev->dev); - return ret; + goto out; } /* flood test */

3 2

[alsa-devel] [PATCH 2/3 v2] pcm_file: improve error checking in write_wav_header function
by Adam Miartus 05 Jul '19

05 Jul '19

previously errno would be returned even for cases where it may have not been populated, for example one of the write functions failing, or writing only partial buffer, now progress through write operations separately and report errno when appropriate Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> --- src/pcm/pcm_file.c | 38 ++++++++++++++++++++++++++++++-------- 1 file changed, 30 insertions(+), 8 deletions(-) diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 03c1640..da7f087 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -327,6 +327,8 @@ static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) static int write_wav_header(snd_pcm_t *pcm) { snd_pcm_file_t *file = pcm->private_data; + ssize_t res; + static const char header[] = { 'R', 'I', 'F', 'F', 0x24, 0, 0, 0, @@ -341,15 +343,35 @@ static int write_wav_header(snd_pcm_t *pcm) setup_wav_header(pcm, &file->wav_header); - if (write(file->fd, header, sizeof(header)) != sizeof(header) || - write(file->fd, &file->wav_header, sizeof(file->wav_header)) != - sizeof(file->wav_header) || - write(file->fd, header2, sizeof(header2)) != sizeof(header2)) { - int err = errno; - SYSERR("%s write header failed, file data may be corrupt", file->fname); - return -err; - } + res = write(file->fd, header, sizeof(header)); + if (res != sizeof(header)) + goto write_error; + + res = write(file->fd, &file->wav_header, sizeof(file->wav_header)); + if (res != sizeof(file->wav_header)) + goto write_error; + + res = write(file->fd, header2, sizeof(header2)); + if (res != sizeof(header2)) + goto write_error; + return 0; + +write_error: + /* + * print real errno if available and return EIO, reason for this is + * to block possible EPIPE in case file->fd is a pipe. EPIPE from + * file->fd conflicts with EPIPE from playback stream which should + * be used to signal XRUN on playback device + */ + if (res < 0) + SYSERR("%s write header failed, file data may be corrupt", file->fname); + else + SNDERR("%s write header incomplete, file data may be corrupt", file->fname); + + memset(&file->wav_header, 0, sizeof(struct wav_fmt)); + + return -EIO; } /* fix up the length fields in WAV header */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: cs4281: remove redundant assignment to variable val and remove a goto
by Colin King 05 Jul '19

05 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> The variable val is being assigned with a value that is never read and it is being updated later with a new value. The assignment is redundant and can be removed. Also remove a goto statement and a label and replace with a break statement. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/cs4281.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index a2cce3ecda6f..04c712647853 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -694,7 +694,7 @@ static int snd_cs4281_trigger(struct snd_pcm_substream *substream, int cmd) static unsigned int snd_cs4281_rate(unsigned int rate, unsigned int *real_rate) { - unsigned int val = ~0; + unsigned int val; if (real_rate) *real_rate = rate; @@ -707,9 +707,8 @@ static unsigned int snd_cs4281_rate(unsigned int rate, unsigned int *real_rate) case 44100: return 1; case 48000: return 0; default: - goto __variable; + break; } - __variable: val = 1536000 / rate; if (real_rate) *real_rate = 1536000 / val; -- 2.20.1

3 3

[alsa-devel] bug in playing 6-channel sound
by Julian Bradfield 05 Jul '19

05 Jul '19

Some five years ago, the following post https://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/072265.… reported a failure to play six-channel sound, that could be worked around by setting the prealloc value for the device to various values *other than* the default 64. I have just run into this playing a movie from my laptop (Thinkpad Yoga X1, Intel HDA PCH sound card outputting through HDMI) running Xubuntu 18.04, up to date as at the time of posting. I see exactly the same problem as in that post, and the same workaround works for me - I tried prealloc values 63,65,32,128, and any of them is ok. I'm a bit baffled that such a problem has survived for several years. If any developer would like me to provide debugging information, please get in touch - I have some time available at present, but not enough to learn ALSA and debug it myself! I will remain subscribed to the list for a few weeks to read any followups.

2 1

[alsa-devel] [GIT PULL]: soundwire updates for v5.3-rc1
by Vinod Koul 05 Jul '19

05 Jul '19

Hello Greg, Please pull to receive the updates for soundwire for v5.3. The following changes since commit a188339ca5a396acc588e5851ed7e19f66b0ebd9: Linux 5.2-rc1 (2019-05-19 15:47:09 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.3-rc1 for you to fetch changes up to 2aeac95d1a4cc85aae57ab842d5c3340df0f817f: soundwire: add module_sdw_driver helper macro (2019-07-01 11:52:43 +0530) ---------------------------------------------------------------- soundwire updates for v5.3-rc1 Updates for 5.3 include: - module_sdw_driver macro for drivers - Documentation updates for code-blocks - Improvement from Pierre on intel and cadence driver - Clarification of DisCo properties and updates ---------------------------------------------------------------- Jan Kotas (1): soundwire: cdns: Fix compilation error on arm64 Mauro Carvalho Chehab (1): docs: soundwire: locking: fix tags for a code-block Pierre-Louis Bossart (15): soundwire: intel: filter SoundWire controller device search soundwire: mipi_disco: fix master/link error soundwire: add port-related definitions soundwire: remove master data port properties soundwire: mipi-disco: remove master_count property for masters soundwire: rename 'freq' fields soundwire: mipi-disco: fix clock stop modes soundwire: clarify comment soundwire: rename/clarify MIPI DisCo properties soundwire: cadence_master: use rate_limited dynamic debug soundwire: cadence_master: log Slave status mask on errors soundwire: cadence_master: check the number of bidir PDIs soundwire: Intel: add log for number of PCM and PDM PDIs soundwire: fix typo in comments soundwire: intel_init: add checks on link numbers Srinivas Kandagatla (1): soundwire: add module_sdw_driver helper macro Documentation/driver-api/soundwire/locking.rst | 4 +- drivers/soundwire/bus.c | 6 +- drivers/soundwire/cadence_master.c | 30 +++++---- drivers/soundwire/intel.c | 17 +++-- drivers/soundwire/intel.h | 2 +- drivers/soundwire/intel_init.c | 25 +++++++- drivers/soundwire/mipi_disco.c | 35 +++++----- drivers/soundwire/stream.c | 8 +-- include/linux/soundwire/sdw.h | 88 ++++++++++++++++++++------ include/linux/soundwire/sdw_type.h | 11 ++++ 10 files changed, 162 insertions(+), 64 deletions(-) -- ~Vinod

2 1

[alsa-devel] [PATCH] sound: pci: asihpi: Remove unneeded variable change
by Hariprasad Kelam 05 Jul '19

05 Jul '19

this patch fixes below issue reported by coccicheck sound/pci/asihpi/asihpi.c:1558:5-11: Unneeded variable: "change". Return "1" on line 1564 Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/pci/asihpi/asihpi.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e7234f3..2a21a3d 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1519,7 +1519,6 @@ static int snd_asihpi_volume_get(struct snd_kcontrol *kcontrol, static int snd_asihpi_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int change; u32 h_control = kcontrol->private_value; short an_gain_mB[HPI_MAX_CHANNELS]; @@ -1530,9 +1529,8 @@ static int snd_asihpi_volume_put(struct snd_kcontrol *kcontrol, /* change = asihpi->mixer_volume[addr][0] != left || asihpi->mixer_volume[addr][1] != right; */ - change = 1; hpi_handle_error(hpi_volume_set_gain(h_control, an_gain_mB)); - return change; + return 1; } static const DECLARE_TLV_DB_SCALE(db_scale_100, -10000, VOL_STEP_mB, 0); @@ -1555,13 +1553,12 @@ static int snd_asihpi_volume_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { u32 h_control = kcontrol->private_value; - int change = 1; /* HPI currently only supports all or none muting of multichannel volume ALSA Switch element has opposite sense to HPI mute: on==unmuted, off=muted */ int mute = ucontrol->value.integer.value[0] ? 0 : HPI_BITMASK_ALL_CHANNELS; hpi_handle_error(hpi_volume_set_mute(h_control, mute)); - return change; + return 1; } static int snd_asihpi_volume_add(struct snd_card_asihpi *asihpi, -- 2.7.4

2 1

[alsa-devel] audio lost from speaker after reboot from windows on the device ALC295
by He, Bo 05 Jul '19

05 Jul '19

Hi, patch_realtek.c maintainer: I see one issue that reboot from windows and boot to ubuntu, the audio lost from speaker, I suspect there are some bugs in patch_realtek.c drivers, the device is ALC295 and the device id is 0x10ec0295. I have done the below experiments: 1. reboot from windows to windows, the audio is persist . 2. reboot from windows to ubuntu, the audio lost from speaker, but can hear if I hotplug one earphone. 3. if the issue reproduce after reboot from windows, reboot the ubuntu can't restore the audio, I suspect it's warm reset. 4. if I write the port 0xcf9 with 0xe to do cold reset, the audio can restore. 5. if I do suspend/resume, the audio can restore, I suspect do cold boot and suspend will trigger the platform reset to reset the ALC295. 6. if I do double function reset (write the verb 0x7ff in alc_init), the audio is still can't restore. snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_CODEC_RESET, 0); /* Function reset */ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_CODEC_RESET, 0); /* double Function reset */ 7. the issue is first found on kernel 4.19.50, I still see the issue with the latest kernel 5.2-rc2, is it possible windows change some default registers, but ALC295 don't initialize the register?

3 4

[alsa-devel] [PATCH 0/7] ASoC: Intel: Skylake: Driver fundaments overhaul
by Cezary Rojewski 04 Jul '19

04 Jul '19

Skylake driver is divided into two modules: - snd_soc_skl - snd_soc_skl_ipc and nothing would be wrong if not for the fact that both cannot exist without one another. IPC module is not some kind of extension, as it is the case for snd_hda_ext_core which is separated from snd_hda_core - legacy hda interface. It's as much core Skylake module as snd_soc_skl is. Statement backup by existence of circular dependency between this two. To eliminate said problem, struct skl_sst has been created. From that momment, Skylake has been plagued by header errors (incomplete sturcts, unknown references etc.) whenever something new is to be added or code is cleaned up. Fix this flawed design by merging snd_soc_skl and snd_soc_skl_ipc. Also, do not forget about struct skl_sst redundancy. Followup changes address harmful assumptions and false logic which driver currently implements e.g.: attempt to take role of master for DSP scheduling when in fact entire control takes place in DSP. Amadeusz Sławiński (2): ASoC: Intel: Skylake: Combine snd_soc_skl_ipc and snd_soc_skl ASoC: Intel: Skylake: Do not disable FW notifications Cezary Rojewski (5): ASoC: Intel: Skylake: Merge skl_sst and skl into skl_dev struct ASoC: Intel: Skylake: Remove MCPS available check ASoC: Intel: Skylake: Remove memory available check ASoC: Intel: Skylake: Make MCPS and CPS params obsolete ASoC: Intel: Skylake: Cleanup skl_module_cfg declaration sound/soc/intel/common/sst-ipc.h | 1 + sound/soc/intel/skylake/Makefile | 12 +- sound/soc/intel/skylake/bxt-sst.c | 50 +-- sound/soc/intel/skylake/cnl-sst-dsp.h | 7 +- sound/soc/intel/skylake/cnl-sst.c | 37 +- sound/soc/intel/skylake/skl-debug.c | 14 +- sound/soc/intel/skylake/skl-messages.c | 245 ++++++------- sound/soc/intel/skylake/skl-nhlt.c | 18 +- sound/soc/intel/skylake/skl-pcm.c | 74 ++-- sound/soc/intel/skylake/skl-ssp-clk.c | 4 +- sound/soc/intel/skylake/skl-sst-dsp.c | 10 +- sound/soc/intel/skylake/skl-sst-dsp.h | 29 +- sound/soc/intel/skylake/skl-sst-ipc.c | 8 +- sound/soc/intel/skylake/skl-sst-ipc.h | 52 +-- sound/soc/intel/skylake/skl-sst-utils.c | 37 +- sound/soc/intel/skylake/skl-sst.c | 51 +-- sound/soc/intel/skylake/skl-topology.c | 441 ++++++++---------------- sound/soc/intel/skylake/skl-topology.h | 43 +-- sound/soc/intel/skylake/skl.c | 54 +-- sound/soc/intel/skylake/skl.h | 102 ++++-- 20 files changed, 546 insertions(+), 743 deletions(-) -- 2.17.1

1 7

[alsa-devel] [PATCH v2] ASoC: wcd9335: remove multiple defines.
by Srinivas Kandagatla 04 Jul '19

04 Jul '19

Found during review that there are multiple defines of same constants. This patch removes them! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: rebased on top of kernel/git/broonie/sound.git for-next sound/soc/codecs/wcd9335.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 85737fe54474..1bbbe421b999 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -86,11 +86,6 @@ #define WCD9335_DEC_PWR_LVL_HP 0x04 #define WCD9335_DEC_PWR_LVL_DF 0x00 -#define TX_HPF_CUT_OFF_FREQ_MASK 0x60 -#define CF_MIN_3DB_4HZ 0x0 -#define CF_MIN_3DB_75HZ 0x1 -#define CF_MIN_3DB_150HZ 0x2 - #define WCD9335_SLIM_RX_CH(p) \ {.port = p + WCD9335_RX_START, .shift = p,} -- 2.21.0

1 0

[alsa-devel] [PATCH] ASoC: wcd9335: remove multiple defines.
by Srinivas Kandagatla 04 Jul '19

04 Jul '19

Found during review that there are multiple defines of same constants. This patch removes them! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd9335.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 6f205638a8b8..da6c50ccecd9 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -87,11 +87,6 @@ #define WCD9335_DEC_PWR_LVL_HP 0x04 #define WCD9335_DEC_PWR_LVL_DF 0x00 -#define TX_HPF_CUT_OFF_FREQ_MASK 0x60 -#define CF_MIN_3DB_4HZ 0x0 -#define CF_MIN_3DB_75HZ 0x1 -#define CF_MIN_3DB_150HZ 0x2 - #define WCD9335_MBHC_MAX_BUTTONS (8) #define WCD9335_SLIM_RX_CH(p) \ -- 2.21.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix parse of UAC2 Extension Units
by Takashi Iwai 04 Jul '19

04 Jul '19

Extension Unit (XU) is used to have a compatible layout with Processing Unit (PU) on UAC1, and the usb-audio driver code assumed it for parsing the descriptors. Meanwhile, on UAC2, XU became slightly incompatible with PU; namely, XU has a one-byte bmControls bitmap while PU has two bytes bmControls bitmap. This incompatibility results in the read of a wrong address for the last iExtension field, which ended up with an incorrect string for the mixer element name, as recently reported for Focusrite Scarlett 18i20 device. This patch corrects the present misalignment by introducing a couple of new macros and calling them depending on the descriptor type. Reported-by: Stefan Sauer <ensonic(a)hora-obscura.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/linux/usb/audio.h | 37 +++++++++++++++++++++++++++++++++++++ sound/usb/mixer.c | 16 ++++++++++------ 2 files changed, 47 insertions(+), 6 deletions(-) diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h index ddc5396800aa..76b7c3f6cd0d 100644 --- a/include/uapi/linux/usb/audio.h +++ b/include/uapi/linux/usb/audio.h @@ -450,6 +450,43 @@ static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_desc } } +/* + * Extension Unit (XU) has almost compatible layout with Processing Unit, but + * on UAC2, it has a different bmControls size (bControlSize); it's 1 byte for + * XU while 2 bytes for PU. The last iExtension field is a one-byte index as + * well as iProcessing field of PU. + */ +static inline __u8 uac_extension_unit_bControlSize(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + switch (protocol) { + case UAC_VERSION_1: + return desc->baSourceID[desc->bNrInPins + 4]; + case UAC_VERSION_2: + return 1; /* in UAC2, this value is constant */ + case UAC_VERSION_3: + return 4; /* in UAC3, this value is constant */ + default: + return 1; + } +} + +static inline __u8 uac_extension_unit_iExtension(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + __u8 control_size = uac_extension_unit_bControlSize(desc, protocol); + + switch (protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + return *(uac_processing_unit_bmControls(desc, protocol) + + control_size); + case UAC_VERSION_3: + return 0; /* UAC3 does not have this field */ + } +} + /* 4.5.2 Class-Specific AS Interface Descriptor */ struct uac1_as_header_descriptor { __u8 bLength; /* in bytes: 7 */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e003b5e7b01a..ac121b10c51c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2318,7 +2318,7 @@ static struct procunit_info extunits[] = { */ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw_desc, struct procunit_info *list, - char *name) + bool extension_unit) { struct uac_processing_unit_descriptor *desc = raw_desc; int num_ins; @@ -2335,6 +2335,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, static struct procunit_info default_info = { 0, NULL, default_value_info }; + const char *name = extension_unit ? + "Extension Unit" : "Processing Unit"; if (desc->bLength < 13) { usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); @@ -2448,7 +2450,10 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, } else if (info->name) { strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); } else { - nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol); + if (extension_unit) + nameid = uac_extension_unit_iExtension(desc, state->mixer->protocol); + else + nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol); len = 0; if (nameid) len = snd_usb_copy_string_desc(state->chip, @@ -2481,10 +2486,10 @@ static int parse_audio_processing_unit(struct mixer_build *state, int unitid, case UAC_VERSION_2: default: return build_audio_procunit(state, unitid, raw_desc, - procunits, "Processing Unit"); + procunits, false); case UAC_VERSION_3: return build_audio_procunit(state, unitid, raw_desc, - uac3_procunits, "Processing Unit"); + uac3_procunits, false); } } @@ -2495,8 +2500,7 @@ static int parse_audio_extension_unit(struct mixer_build *state, int unitid, * Note that we parse extension units with processing unit descriptors. * That's ok as the layout is the same. */ - return build_audio_procunit(state, unitid, raw_desc, - extunits, "Extension Unit"); + return build_audio_procunit(state, unitid, raw_desc, extunits, true); } /* -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: hda/ca0132 - remove redundant assignment to variable 'changed'
by Colin King 04 Jul '19

04 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> The variable 'changed' is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c0565437ffed..0d51823d7270 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -5980,7 +5980,7 @@ static int ca0132_alt_volume_put(struct snd_kcontrol *kcontrol, int ch = get_amp_channels(kcontrol); long *valp = ucontrol->value.integer.value; hda_nid_t vnid = 0; - int changed = 1; + int changed; switch (nid) { case 0x02: -- 2.20.1

2 1

[alsa-devel] [PATCH v2 33/35] sound/pci: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 04 Jul '19

04 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memcpy, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memcpy. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- Changes in v2: - Fix a typo in commit message (memset -> memcpy) sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index b612a536a5a1..35bd3e7c8ce1 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2189,11 +2189,10 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); + commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { -- 2.11.0

2 1

[alsa-devel] Dell Headphone Mic can't record after S3
by Kailang 04 Jul '19

04 Jul '19

Hi Takashi, Headphone Mic can't record after S3 on Dell headset mode platform. The S3 mode was deep. S2idle didn't have this issue. BR, Kailang

2 1

[alsa-devel] [PATCH 0/2] ASoC: meson: axg-tdm-formatter: add g12a reset
by Jerome Brunet 04 Jul '19

04 Jul '19

This patchset add the possibility to provide a reset to the tdm formatter. Such reset is available on the g12a SoC family and helps solve a random channel output shift when using more than one output lane. Jerome Brunet (2): ASoC: meson: axg-tdm-formatter: add reset to the bindings documentation ASoC: meson: axg-tdm-formatter: add reset .../sound/amlogic,axg-tdm-formatters.txt | 6 +++- sound/soc/meson/axg-tdm-formatter.c | 29 +++++++++++++++++++ 2 files changed, 34 insertions(+), 1 deletion(-) -- 2.21.0

2 4

[alsa-devel] [PATCH] ASoC: core: Return -ENOTSUPP from set_channel_map() if no operation provided
by Srinivas Kandagatla 04 Jul '19

04 Jul '19

It makes it easier for common code to work with snd_soc_dai_set_channel_map() by distinguishing between operation not being supported and an error. This is done inline with others snd_soc_dai.* apis. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f08bef7328d7..62aae5a5c14b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2688,7 +2688,7 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot, rx_num, rx_slot); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); -- 2.21.0

2 1

[alsa-devel] [v3] ASoC: max98357a: avoid speaker pop when playback startup
by mac.chiang＠intel.com 04 Jul '19

04 Jul '19

From: Mac Chiang <mac.chiang(a)intel.com> Loud speaker pop happens during playback even when in slience playback. Specify Max98357a amp delay times to make sure clocks are always earlier than sdmode on. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v3: - rebase to branch for-next v2: - added device_property_read_u32() func --- --- .../devicetree/bindings/sound/max98357a.txt | 4 ++ sound/soc/codecs/max98357a.c | 55 ++++++++++++++++++---- 2 files changed, 50 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index 28645a2..4bce14c 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -9,6 +9,10 @@ Optional properties: - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. If this option is not specified then driver does not manage the pin state (e.g. chip is always on). +- sdmode-delay : specify delay time for SD_MODE pin. + If this option is specified, which means it's required i2s clocks + ready before SD_MODE is unmuted in order to avoid the speaker pop noise. + It's observed that 5ms is sufficient. Example: diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 6f724c9..6f0e28f 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -19,24 +19,42 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +struct max98357a_priv { + struct delayed_work enable_sdmode_work; + struct gpio_desc *sdmode; + unsigned int sdmode_delay; +}; + +static void max98357a_enable_sdmode_work(struct work_struct *work) +{ + struct max98357a_priv *max98357a = + container_of(work, struct max98357a_priv, + enable_sdmode_work.work); + + gpiod_set_value(max98357a->sdmode, 1); +} + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); - if (!sdmode) + if (!max98357a->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + queue_delayed_work(system_power_efficient_wq, + &max98357a->enable_sdmode_work, + msecs_to_jiffies(max98357a->sdmode_delay)); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + cancel_delayed_work_sync(&max98357a->enable_sdmode_work); + gpiod_set_value(max98357a->sdmode, 0); break; } @@ -90,14 +108,33 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { - struct gpio_desc *sdmode; + struct max98357a_priv *max98357a; + int ret; - sdmode = devm_gpiod_get_optional(&pdev->dev, + max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL); + + if (!max98357a) + return -ENOMEM; + + max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); - dev_set_drvdata(&pdev->dev, sdmode); + if (IS_ERR(max98357a->sdmode)) + return PTR_ERR(max98357a->sdmode); + + ret = device_property_read_u32(&pdev->dev, "sdmode-delay", + &max98357a->sdmode_delay); + + if (ret) { + max98357a->sdmode_delay = 0; + dev_dbg(&pdev->dev, + "no optional property 'sdmode-delay' found, default: no delay\n"); + } + + dev_set_drvdata(&pdev->dev, max98357a); + + INIT_DELAYED_WORK(&max98357a->enable_sdmode_work, + max98357a_enable_sdmode_work); return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6afe-dai: Add missing Slimbus0 audio route
by Srinivas Kandagatla 04 Jul '19

04 Jul '19

For some reason SLIMBus RX0 playback is not added to audio routes. This patch adds the missing route. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index dc645ba4d8d0..c1a7624eaf17 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -447,6 +447,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"HDMI Playback", NULL, "HDMI_RX"}, {"Display Port Playback", NULL, "DISPLAY_PORT_RX"}, + {"Slimbus Playback", NULL, "SLIMBUS_0_RX"}, {"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"}, {"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"}, {"Slimbus3 Playback", NULL, "SLIMBUS_3_RX"}, -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: fix use-after-free in graph_for_each_link
by Wen Yang 04 Jul '19

04 Jul '19

After calling of_node_put() on the codec_ep and codec_port variables, they are still being used, which may result in use-after-free. We fix this issue by calling of_node_put() after the last usage. Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/generic/audio-graph-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index e438011..30a4e83 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -421,9 +421,6 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_get_parent(codec_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); graph_parse_convert(dev, codec_ep, &adata); @@ -443,6 +440,9 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, else ret = func_noml(priv, cpu_ep, codec_ep, li); + of_node_put(codec_ep); + of_node_put(codec_port); + if (ret < 0) return ret; -- 2.9.5

2 1

[alsa-devel] [PATCH v2 34/35] sound/soc/codecs: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 04 Jul '19

04 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memcpy, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memcpy. Acked-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- Changes in v2: - Fix a typo in commit message (memset -> memcpy) - Split into two patches sound/soc/codecs/wm0010.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 727d6703c905..807826f30f58 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -515,7 +515,7 @@ static int wm0010_stage2_load(struct snd_soc_component *component) dev_dbg(component->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); + img = kmemdup(&fw->data[0], fw->size, GFP_KERNEL | GFP_DMA); if (!img) { ret = -ENOMEM; goto abort2; @@ -527,8 +527,6 @@ static int wm0010_stage2_load(struct snd_soc_component *component) goto abort1; } - memcpy(img, &fw->data[0], fw->size); - spi_message_init(&m); memset(&t, 0, sizeof(t)); t.rx_buf = out; -- 2.11.0

2 1

[alsa-devel] GIT: Regarding the issue we are facing in the commit 37728639ae05de702825d96bd1d42e24ae772248
by vanitha.channaiah＠in.bosch.com 04 Jul '19

04 Jul '19

This mail is regarding the fix at below: commit 37728639ae05de702825d96bd1d42e24ae772248 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Sat Feb 7 15:01:31 2004 +00 The commit has following changes : ******************************* code ***************************************** static int snd_pcm_rate_hw_refine_cchange(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_hw_params_t *sparams) { + buffer_size = snd_pcm_hw_param_get_interval(params, SND_PCM_HW_PARAM_BUFFER_SIZE); + /* + * this condition probably needs more work: + * in case when the buffer_size is known and we are looking + * for best period_size, we should prefer situation when + * (buffer_size / period_size) * period_size == buffer_size + */ + if (snd_interval_single(buffer_size) && buffer_size->integer) { + snd_interval_t *period_size; + period_size = (snd_interval_t *)snd_pcm_hw_param_get_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE); + if (!snd_interval_checkempty(period_size) && + period_size->openmin && period_size->openmax && + period_size->min + 1 == period_size->max) { + if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { + snd_interval_set_value(period_size, period_size->min); + } else if ((buffer_size->max / period_size->max) * period_size->max == buffer_size->max) { + snd_interval_set_value(period_size, period_size->max); + } + } + } } ******************************* code **************************************** Update rate depending on period_size and period_time. Issue found in Hardware : a. RCAR salvator-xs which supports 2 channel. b. IMX which supports 8 channel. 1. Usecase which is PASSED without the mentioned commit and FAILED with the commit. For the usecase with below command : rate = 11025 channel = 6 $aplay -Dentertainment_main -r11025 -c6 -fS16_LE /dev/urandom a. With the commit: In the calculation of RATE at Rule 7: RATE=11025 dependent parameters are: PERIOD_SIZE=88 PERIOD_TIME=8000 parameters min max open_min open_max interval PERIOD_SIZE 88 88 0 0 0 PERIO_TIME 8000 8000 0 0 0 RATE 11025 11000 0 0 1 RATE values are calculated in snd_interval_refine() Return value: INVALID as rate_min > rate_max This is because, dependent parameter “period_size” is rounded to 88 in “rate plugin” in snd_pcm_rate_hw_refine_cchange() Since, the below condition gets satisfied(buffer_size = 352 aligned to period_size = 88), period_size gets rounded to 88. if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { This commit changes causing the issue to get rate_min > rate_max. The flow of code execution is as follows: - snd_pcm_hw_refine_slave() - Enters snd_pcm_rate_hw_refine_cchange(), rounding of period_size to 88. - snd_pcm_rate_hw_refine_cchange() exit. - snd_pcm_hw_refine_soft() is called, here exists params->rmask to evaluate for SAMPLE_BITS, FRAME_BITS, PERIOD_BYTES, BUFFER_BYTES - execution of RULES calculation. At RULE 7 rate calculation INVALID error is observed. b. Without the commit: In the case if period_size doesn’t get rounded off to 88, the usecase would get PASSED. Rate calculation goes fine with period_size open interval (88, 89) parameters min max open_min open_max interval PERIOD_SIZE 88 89 0 0 0 PERIOD_TIME 8000 8000 0 0 0 RATE 11025 11025 0 0 1 Here, there is no issue of rate_min > rate_max. **************************************************************************************************************************************************************** Now, I just have one scenario where with/without commit, rate calculation goes fine. 2. Usecase which is PASSED with and without above commit. For the usecase with below command : rate =11025 channel=2 $aplay -Dentertainment_main -r11025 -c2 -fS16_LE /dev/urandom a. Including the above commit code: Below is the calculation that goes: The flow of code execution is as follows: - snd_pcm_hw_refine_slave() - Enters snd_pcm_rate_hw_refine_cchange() , rounding of period_size to 88. - snd_pcm_rate_hw_refine_cchange() exit. - snd_pcm_hw_refine_soft() is called, here params->rmask is 0. - RULES calculation doesnt occur. b. Without the commit: In the case if period_size doesn’t get rounded off to 88. Rate calculation goes fine with period_size open interval (88, 89) Is there any dependency for the commit ? If yes, can you please suggest the corner case which fails without the commit ? Can we revert the changes ?

3 3

[alsa-devel] [PATCH] sound/soc/codecs/tas5720.c: add ACPI support
by Nikolaus Voss 04 Jul '19

04 Jul '19

Add support for ACPI enumeration for tas5720 and tas5722. Use device_match API to unify access to driver data for DT and ACPI. Aggregate variant stuff into its own struct and directly reference it in variant data for i2c/of/acpi_device_id. Signed-off-by: Nikolaus Voss <nikolaus.voss(a)loewensteinmedical.de> --- sound/soc/codecs/tas5720.c | 215 +++++++++++++++++-------------------- 1 file changed, 99 insertions(+), 116 deletions(-) diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index 37fab8f22800..ea973764c745 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -7,6 +7,7 @@ * Author: Andreas Dannenberg <dannenberg(a)ti.com> */ +#include <linux/acpi.h> #include <linux/module.h> #include <linux/errno.h> #include <linux/device.h> @@ -28,9 +29,10 @@ /* Define how often to check (and clear) the fault status register (in ms) */ #define TAS5720_FAULT_CHECK_INTERVAL 200 -enum tas572x_type { - TAS5720, - TAS5722, +struct tas5720_variant { + const int device_id; + const struct regmap_config reg_config; + const struct snd_soc_component_driver comp_drv; }; static const char * const tas5720_supply_names[] = { @@ -44,7 +46,7 @@ struct tas5720_data { struct snd_soc_component *component; struct regmap *regmap; struct i2c_client *tas5720_client; - enum tas572x_type devtype; + const struct tas5720_variant *variant; struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; struct delayed_work fault_check_work; unsigned int last_fault; @@ -179,17 +181,13 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, goto error_snd_soc_component_update_bits; /* Configure TDM slot width. This is only applicable to TAS5722. */ - switch (tas5720->devtype) { - case TAS5722: + if (tas5720->variant->device_id == TAS5722_DEVICE_ID) { ret = snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, TAS5722_TDM_SLOT_16B, slot_width == 16 ? TAS5722_TDM_SLOT_16B : 0); if (ret < 0) goto error_snd_soc_component_update_bits; - break; - default: - break; } return 0; @@ -277,7 +275,7 @@ static void tas5720_fault_check_work(struct work_struct *work) static int tas5720_codec_probe(struct snd_soc_component *component) { struct tas5720_data *tas5720 = snd_soc_component_get_drvdata(component); - unsigned int device_id, expected_device_id; + unsigned int device_id; int ret; tas5720->component = component; @@ -301,21 +299,9 @@ static int tas5720_codec_probe(struct snd_soc_component *component) goto probe_fail; } - switch (tas5720->devtype) { - case TAS5720: - expected_device_id = TAS5720_DEVICE_ID; - break; - case TAS5722: - expected_device_id = TAS5722_DEVICE_ID; - break; - default: - dev_err(component->dev, "unexpected private driver data\n"); - return -EINVAL; - } - - if (device_id != expected_device_id) + if (device_id != tas5720->variant->device_id) dev_warn(component->dev, "wrong device ID. expected: %u read: %u\n", - expected_device_id, device_id); + tas5720->variant->device_id, device_id); /* Set device to mute */ ret = snd_soc_component_update_bits(component, TAS5720_DIGITAL_CTRL2_REG, @@ -462,24 +448,6 @@ static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config tas5720_regmap_config = { - .reg_bits = 8, - .val_bits = 8, - - .max_register = TAS5720_MAX_REG, - .cache_type = REGCACHE_RBTREE, - .volatile_reg = tas5720_is_volatile_reg, -}; - -static const struct regmap_config tas5722_regmap_config = { - .reg_bits = 8, - .val_bits = 8, - - .max_register = TAS5722_MAX_REG, - .cache_type = REGCACHE_RBTREE, - .volatile_reg = tas5720_is_volatile_reg, -}; - /* * DAC analog gain. There are four discrete values to select from, ranging * from 19.2 dB to 26.3dB. @@ -558,40 +526,6 @@ static const struct snd_soc_dapm_route tas5720_audio_map[] = { { "OUT", NULL, "DAC" }, }; -static const struct snd_soc_component_driver soc_component_dev_tas5720 = { - .probe = tas5720_codec_probe, - .remove = tas5720_codec_remove, - .suspend = tas5720_suspend, - .resume = tas5720_resume, - .controls = tas5720_snd_controls, - .num_controls = ARRAY_SIZE(tas5720_snd_controls), - .dapm_widgets = tas5720_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), - .dapm_routes = tas5720_audio_map, - .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -static const struct snd_soc_component_driver soc_component_dev_tas5722 = { - .probe = tas5720_codec_probe, - .remove = tas5720_codec_remove, - .suspend = tas5720_suspend, - .resume = tas5720_resume, - .controls = tas5722_snd_controls, - .num_controls = ARRAY_SIZE(tas5722_snd_controls), - .dapm_widgets = tas5720_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), - .dapm_routes = tas5720_audio_map, - .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - /* PCM rates supported by the TAS5720 driver */ #define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) @@ -637,29 +571,25 @@ static int tas5720_probe(struct i2c_client *client, { struct device *dev = &client->dev; struct tas5720_data *data; - const struct regmap_config *regmap_config; + const struct tas5720_variant *type; int ret; int i; + type = device_get_match_data(&client->dev); + if (!type && id) + type = (const struct tas5720_variant *)id->driver_data; + + if (!type) + return -EINVAL; + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); if (!data) return -ENOMEM; data->tas5720_client = client; - data->devtype = id->driver_data; + data->variant = type; - switch (id->driver_data) { - case TAS5720: - regmap_config = &tas5720_regmap_config; - break; - case TAS5722: - regmap_config = &tas5722_regmap_config; - break; - default: - dev_err(dev, "unexpected private driver data\n"); - return -EINVAL; - } - data->regmap = devm_regmap_init_i2c(client, regmap_config); + data->regmap = devm_regmap_init_i2c(client, &type->reg_config); if (IS_ERR(data->regmap)) { ret = PTR_ERR(data->regmap); dev_err(dev, "failed to allocate register map: %d\n", ret); @@ -678,51 +608,104 @@ static int tas5720_probe(struct i2c_client *client, dev_set_drvdata(dev, data); - switch (id->driver_data) { - case TAS5720: - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5720, - tas5720_dai, - ARRAY_SIZE(tas5720_dai)); - break; - case TAS5722: - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5722, - tas5720_dai, - ARRAY_SIZE(tas5720_dai)); - break; - default: - dev_err(dev, "unexpected private driver data\n"); - return -EINVAL; - } - if (ret < 0) { - dev_err(dev, "failed to register component: %d\n", ret); - return ret; - } + ret = devm_snd_soc_register_component(&client->dev, + &type->comp_drv, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); - return 0; + if (ret < 0) + dev_err(dev, "failed to register component: %d\n", ret); + + return ret; } +static const struct tas5720_variant variant[] = { + { + .device_id = TAS5720_DEVICE_ID, + .reg_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS5720_MAX_REG, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5720_is_volatile_reg, + }, + .comp_drv = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5720_snd_controls, + .num_controls = ARRAY_SIZE(tas5720_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1 + }, + }, + { + .device_id = TAS5722_DEVICE_ID, + .reg_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS5722_MAX_REG, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5720_is_volatile_reg, + }, + .comp_drv = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5722_snd_controls, + .num_controls = ARRAY_SIZE(tas5722_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + }, + }, +}; + static const struct i2c_device_id tas5720_id[] = { - { "tas5720", TAS5720 }, - { "tas5722", TAS5722 }, + { "tas5720", (kernel_ulong_t)&variant[0] }, + { "tas5722", (kernel_ulong_t)&variant[1] }, { } }; MODULE_DEVICE_TABLE(i2c, tas5720_id); #if IS_ENABLED(CONFIG_OF) static const struct of_device_id tas5720_of_match[] = { - { .compatible = "ti,tas5720", }, - { .compatible = "ti,tas5722", }, + { .compatible = "ti,tas5720", .data = &variant[0], }, + { .compatible = "ti,tas5722", .data = &variant[1], }, { }, }; MODULE_DEVICE_TABLE(of, tas5720_of_match); #endif +#if IS_ENABLED(CONFIG_ACPI) +static const struct acpi_device_id tas5720_acpi_match[] = { + { "10TI5720", (kernel_ulong_t)&variant[0] }, + { "10TI5722", (kernel_ulong_t)&variant[1] }, + { } +}; +MODULE_DEVICE_TABLE(acpi, tas5720_acpi_match); +#endif + static struct i2c_driver tas5720_i2c_driver = { .driver = { .name = "tas5720", .of_match_table = of_match_ptr(tas5720_of_match), + .acpi_match_table = ACPI_PTR(tas5720_acpi_match), }, .probe = tas5720_probe, .id_table = tas5720_id, -- 2.17.1

4 13

[alsa-devel] [PATCH V2 0/2] recover the channel swap after xrun
by shengjiu.wang＠nxp.com 03 Jul '19

03 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> recover the channel swap after xrun Shengjiu Wang (2): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun sound/soc/fsl/fsl_esai.c | 267 ++++++++++++++++++++++++++++----------- 1 file changed, 194 insertions(+), 73 deletions(-) Changes in v2 - add one patch for wrap operations to functions. - fix some coding style issue -- 2.21.0

3 5

[alsa-devel] when aplaymidi exits the sustained notes are played forever.
by GitHub issues - opened 03 Jul '19

03 Jul '19

alsa-project/alsa-utils issue #22 was opened from mercuriete: Transfered from here: https://github.com/OpenTTD/OpenTTD/issues/7637 ### Expected Result All notes are stoped before exit aplaymidi ### Actual Result Notes are sustained forever and I have to press the panic button on Qsynth ### Steps to Reproduce 1. Open Qsynth with fluid-soundfonts 2. do this: ```aplaymidi --port 128:0 file.mid``` 3. do control+c to kill aplaymidi 4. Qsynth plays a note forever. ### Possible fix Capture SIGINT and send midi off messages before exit properly. Issue URL : https://github.com/alsa-project/alsa-utils/issues/22 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH 1/2] ASoC: codecs: ad193x: Group register initialization at probe
by Codrin Ciubotariu 03 Jul '19

03 Jul '19

Create a structure with the register initialization values at probe and use it to initialize all the registers at once. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- The order of the initialization is changed, but it doesn't seem to matter. There is one checkpatch warning, let me know if you want to remove it: WARNING: line over 80 characters #32: FILE: sound/soc/codecs/ad193x.c:425: + { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ sound/soc/codecs/ad193x.c | 52 +++++++++++++++++++++++++-------------- 1 file changed, 33 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 3ebc0524f4b2..f944228f014e 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -413,6 +413,38 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { .ops = &ad193x_dai_ops, }; +struct ad193x_reg_default { + unsigned int reg; + unsigned int val; +}; + +/* codec register values to set after reset */ +static void ad193x_reg_default_init(struct ad193x_priv *ad193x) +{ + const struct ad193x_reg_default reg_init[] = { + { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ + { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ + { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ + { 4, 0x1A }, /* DAC_CTRL2: 48kHz de-emphasis, unmute dac */ + { 5, 0x00 }, /* DAC_CHNL_MUTE: unmute DAC channels */ + }; + const struct ad193x_reg_default reg_adc_init[] = { + { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ + { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ + }; + int i; + + for (i = 0; i < ARRAY_SIZE(reg_init); i++) + regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val); + + if (ad193x_has_adc(ad193x)) { + for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) { + regmap_write(ad193x->regmap, reg_adc_init[i].reg, + reg_adc_init[i].val); + } + } +} + static int ad193x_component_probe(struct snd_soc_component *component) { struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(component); @@ -420,25 +452,7 @@ static int ad193x_component_probe(struct snd_soc_component *component) int num, ret; /* default setting for ad193x */ - - /* unmute dac channels */ - regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0); - /* de-emphasis: 48kHz, powedown dac */ - regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); - /* dac in tdm mode */ - regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - - /* adc only */ - if (ad193x_has_adc(ad193x)) { - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); - } - - /* pll input: mclki/xi */ - regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ - regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + ad193x_reg_default_init(ad193x); /* adc only */ if (ad193x_has_adc(ad193x)) { -- 2.20.1

4 5

[alsa-devel] [PATCH v2 35/35] sound/soc/intel: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 03 Jul '19

03 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memcpy, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memcpy. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- Changes in v2: - Fix a typo in commit message (memset -> memcpy) - Split into two patches sound/soc/intel/atom/sst/sst_loader.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index ce11c36848c4..cc95af35c060 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -288,14 +288,13 @@ static int sst_cache_and_parse_fw(struct intel_sst_drv *sst, { int retval = 0; - sst->fw_in_mem = kzalloc(fw->size, GFP_KERNEL); + sst->fw_in_mem = kmemdup(fw->data, fw->size, GFP_KERNEL); if (!sst->fw_in_mem) { retval = -ENOMEM; goto end_release; } dev_dbg(sst->dev, "copied fw to %p", sst->fw_in_mem); dev_dbg(sst->dev, "phys: %lx", (unsigned long)virt_to_phys(sst->fw_in_mem)); - memcpy(sst->fw_in_mem, fw->data, fw->size); retval = sst_parse_fw_memcpy(sst, fw->size, &sst->memcpy_list); if (retval) { dev_err(sst->dev, "Failed to parse fw\n"); -- 2.11.0

1 0

[alsa-devel] [PATCH 30/30] sound/soc: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 03 Jul '19

03 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memset, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memset. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- sound/soc/codecs/wm0010.c | 4 +--- sound/soc/intel/atom/sst/sst_loader.c | 3 +-- 2 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 727d6703c905..807826f30f58 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -515,7 +515,7 @@ static int wm0010_stage2_load(struct snd_soc_component *component) dev_dbg(component->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); + img = kmemdup(&fw->data[0], fw->size, GFP_KERNEL | GFP_DMA); if (!img) { ret = -ENOMEM; goto abort2; @@ -527,8 +527,6 @@ static int wm0010_stage2_load(struct snd_soc_component *component) goto abort1; } - memcpy(img, &fw->data[0], fw->size); - spi_message_init(&m); memset(&t, 0, sizeof(t)); t.rx_buf = out; diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index ce11c36848c4..cc95af35c060 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -288,14 +288,13 @@ static int sst_cache_and_parse_fw(struct intel_sst_drv *sst, { int retval = 0; - sst->fw_in_mem = kzalloc(fw->size, GFP_KERNEL); + sst->fw_in_mem = kmemdup(fw->data, fw->size, GFP_KERNEL); if (!sst->fw_in_mem) { retval = -ENOMEM; goto end_release; } dev_dbg(sst->dev, "copied fw to %p", sst->fw_in_mem); dev_dbg(sst->dev, "phys: %lx", (unsigned long)virt_to_phys(sst->fw_in_mem)); - memcpy(sst->fw_in_mem, fw->data, fw->size); retval = sst_parse_fw_memcpy(sst, fw->size, &sst->memcpy_list); if (retval) { dev_err(sst->dev, "Failed to parse fw\n"); -- 2.11.0

2 2

[alsa-devel] [PATCH 29/30] sound/pci: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 03 Jul '19

03 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memset, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memset. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index b612a536a5a1..35bd3e7c8ce1 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2189,11 +2189,10 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); + commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { -- 2.11.0

1 0

[alsa-devel] [PATCH 0/3] pcm_file report EIO when fail to write to debug file
by Adam Miartus 03 Jul '19

03 Jul '19

Previously, I introduced a patch that may return EPIPE error when pcm_file plugin fails to write to output file for some reason. The purpose of reporting the error was to evade assert in code to allow API user error handling and recovery, or more clarity in debugging. Failing to write to a file usually means bad file descriptor, running out of free memory, insufficient file permissions or in case file is a pipe there may be other reasons preventing the write operation. All these are not easily recoverable by restaring playback as EPIPE error which is used to signal XRUN might suggest. Therefore, use EIO to signal a different kind of error. write_wav_header function is refactored a bit for more detailed error handling in case of failed write operation. Adam Miartus (3): pcm_file: use EIO instead of EPIPE when failing to write output file pcm_file: improve error checking in write_wav_header function pcm_file: in case of failed write clear file buffer variables src/pcm/pcm_file.c | 54 +++++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 41 insertions(+), 13 deletions(-) -- 2.7.4

2 7

[alsa-devel] [PATCH] ASoC: sunxi: sun50i-codec-analog: Add earpiece
by Luca Weiss 03 Jul '19

03 Jul '19

This adds the necessary registers and audio routes to play audio using the Earpiece, that's supported on the A64. Signed-off-by: Luca Weiss <luca(a)z3ntu.xyz> --- So, first of all: This is my first audio patch and I hope I didn't make too many mistakes :) , especially with the routes at the bottom of the patch. What I'm really unsure about, is how the enable & mute registers should be handled. Should I put both registers into a SOC_DOUBLE("Earpiece Playback Switch",...)? sound/soc/sunxi/sun50i-codec-analog.c | 51 +++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index d105c90c3706..6c19fea992c5 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -49,6 +49,15 @@ #define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 #define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 +#define SUN50I_ADDA_EARPIECE_CTRL0 0x03 +#define SUN50I_ADDA_EARPIECE_CTRL0_EAR_RAMP_TIME 4 +#define SUN50I_ADDA_EARPIECE_CTRL0_ESPSR 0 + +#define SUN50I_ADDA_EARPIECE_CTRL1 0x04 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN 7 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE 6 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL 0 + #define SUN50I_ADDA_LINEOUT_CTRL0 0x05 #define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 #define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 @@ -172,6 +181,10 @@ static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), ); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_earpiece_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); /* volume / mute controls */ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { @@ -225,6 +238,19 @@ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { SUN50I_ADDA_LINEOUT_CTRL0_LEN, SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + SOC_SINGLE_TLV("Earpiece Playback Volume", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL, 0x1f, 0, + sun50i_codec_earpiece_vol_scale), + + SOC_SINGLE("Earpiece Playback Switch (enable)", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN, 1, 0), + + SOC_SINGLE("Earpiece Playback Switch", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE, 1, 0), + }; static const char * const sun50i_codec_hp_src_enum_text[] = { @@ -257,6 +283,20 @@ static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { sun50i_codec_lineout_src_enum), }; +static const char * const sun50i_codec_earpiece_src_enum_text[] = { + "DACR", "DACL", "Right Analog Mixer", "Left Analog Mixer", +}; + +static SOC_ENUM_SINGLE_DECL(sun50i_codec_earpiece_src_enum, + SUN50I_ADDA_EARPIECE_CTRL0, + SUN50I_ADDA_EARPIECE_CTRL0_ESPSR, + sun50i_codec_earpiece_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_earpiece_src[] = { + SOC_DAPM_ENUM("Earpiece Source Playback Route", + sun50i_codec_earpiece_src_enum), +}; + static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { /* DAC */ SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, @@ -285,6 +325,10 @@ static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_MUX("Earpiece Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_earpiece_src), + SND_SOC_DAPM_OUTPUT("EARPIECE"), + /* Microphone inputs */ SND_SOC_DAPM_INPUT("MIC1"), @@ -388,6 +432,13 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, + + /* Earpiece Routes */ + { "Earpiece Source Playback Route", "DACL", "Left DAC" }, + { "Earpiece Source Playback Route", "DACR", "Right DAC" }, + { "Earpiece Source Playback Route", "Left Analog Mixer", "Left Mixer" }, + { "Earpiece Source Playback Route", "Right Analog Mixer", "Right Mixer" }, + { "EARPIECE", NULL, "Earpiece Source Playback Route" }, }; static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { -- 2.22.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix Line6 Helix audio format rates
by Nicola Lunghi 03 Jul '19

03 Jul '19

Line6 Helix and HX stomp don't support retrieving the number of clock sample rate. Add a quirk to return the default value of 48Khz. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 28 +++++++++++++++++++++++++--- 1 file changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index c02b51a82775..05442f6ada62 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -313,10 +313,32 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, tmp, sizeof(tmp)); if (ret < 0) { - dev_err(&dev->dev, - "%s(): unable to retrieve number of sample rates (clock %d)\n", + switch (chip->usb_id) { + /* LINE 6 HX pedals don't support getting the clock sample rate. + * Set the framerate to 48khz by default + */ + case USB_ID(0x0E41, 0x4244): /* HELIX */ + case USB_ID(0x0E41, 0x4246): /* HX STOMP */ + dev_warn(&dev->dev, + "%s(): line6 helix: unable to retrieve number of sample rates. Set it to default value (clock %d).\n", __func__, clock); - goto err; + fp->nr_rates = 1; + fp->rate_min = 48000; + fp->rate_max = 48000; + fp->rates = SNDRV_PCM_RATE_48000; + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + if (!fp->rate_table) { + ret = -ENOMEM; + goto err_free; + } + fp->rate_table[0] = 48000; + return 0; + default: + dev_err(&dev->dev, + "%s(): unable to retrieve number of sample rates (clock %d)\n", + __func__, clock); + goto err; + } } nr_triplets = (tmp[1] << 8) | tmp[0]; -- 2.19.1

4 7

[alsa-devel] [PATCH 00/46] ASoC: consider CPU-Platform possibility
by Kuninori Morimoto 03 Jul '19

03 Jul '19

Hi Mark My posted patch (= "ASoC: xxx: don't select unnecessary Platform") removed Platform component settings from Card driver if it selects CPU as Platform. ALSA SoC will ignore duplicate component. I thought these Card driver don't need Platform, but it was wrong. Some CPU component is using generic DMAEngine component. In such case, CPU and Platform have same name but different component, and some Card needs both. My posted above patch breaks (not all but) few Cards. We can revert one-by-one if we get report, but better to revert all from stable kernel point of view. These patches revert all patches. I'm so sorry to bother you and users. I hope these can fixup issue. Kuninori Morimoto (46): ASoC: vc4: vc4_htmi: consider CPU-Platform possibility ASoC: atmel: atmel-classd: consider CPU-Platform possibility ASoC: atmel: atmel-pdmic: consider CPU-Platform possibility ASoC: atmel: atmel_wm8904: consider CPU-Platform possibility unnecessary Platform" ASoC: amtel: mikroe-proto: consider CPU-Platform possibility ASoC: atmel: sam9g20_wm8731: consider CPU-Platform possibility ASoC: atmel: sam9x5_wm8731: consider CPU-Platform possibility ASoC: atmel: tse850-pcm5142: consider CPU-Platform possibility ASoC: fsl: eukrea-tlv320: consider CPU-Platform possibility ASoC: fsl: fsl-asoc-card: consider CPU-Platform possibility ASoC: fsl: imx-es8328: consider CPU-Platform possibility ASoC: fsl: imx-sgtl5000: consider CPU-Platform possibility ASoC: fsl: imx-spdif: consider CPU-Platform possibility ASoC: fsl: imx-audmix: consider CPU-Platform possibility ASoC: kirkwood: armada-370-db: consider CPU-Platform possibility ASoC: mxs: mxs-sgtl5000: consider CPU-Platform possibility ASoC: qcom: apq8016_sbc: consider CPU-Platform possibility ASoC: qcom: storm: consider CPU-Platform possibility ASoC: rockchip: rk3288_hdmi_analog: consider CPU-Platform possibility ASoC: rockchip: rockchip_max98090: consider CPU-Platform possibility ASoC: rockchip: rockchip_rt5645: consider CPU-Platform possibility ASoC: samsung: arndale_rt5631: consider CPU-Platform possibility ASoC: samsung: smdk_wm8994: consider CPU-Platform possibility ASoC: samsung: snow: consider CPU-Platform possibility ASoC: samsung: tm2_wm5110: consider CPU-Platform possibility ASoC: sirf: sirf-audio: consider CPU-Platform possibility ASoC: sunxi: sun4i-codec: consider CPU-Platform possibility ASoC: tegra: tegra_alc5632: consider CPU-Platform possibility ASoC: tegra: tegra_max98090: consider CPU-Platform possibility ASoC: tegra: tegra_rt5640: consider CPU-Platform possibility ASoC: tegra: tegra_rt5677: consider CPU-Platform possibility ASoC: tegra: tegra_sgtl5000: consider CPU-Platform possibility ASoC: tegra: tegra_wm8753: consider CPU-Platform possibility ASoC: tegra: tegra_wm8903: consider CPU-Platform possibility ASoC: tegra: tegra_wm9712: consider CPU-Platform possibility ASoC: tegra: trimslice: consider CPU-Platform possibility ASoC: ti: davinci-evm: consider CPU-Platform possibility ASoC: ti: omap-abe-twl6040: consider CPU-Platform possibility ASoC: ti: omap-hdmi: consider CPU-Platform possibility ASoC: ti: omap-twl4030: consider CPU-Platform possibility ASoC: ti: rx51: consider CPU-Platform possibility ASoC: ux500: mop500: consider CPU-Platform possibility ASoC: simple-card-utils: consider CPU-Platform possibility ASoC: qcom: consider CPU-Platform possibility ASoC: rockchip: rk3399_gru_sound: consider CPU-Platform possibility ASoC: soc-utils: respawn dummy Platform drivers/gpu/drm/vc4/vc4_hdmi.c | 4 ++++ sound/soc/atmel/atmel-classd.c | 5 ++++- sound/soc/atmel/atmel-pdmic.c | 5 ++++- sound/soc/atmel/atmel_wm8904.c | 4 +++- sound/soc/atmel/mikroe-proto.c | 7 +++++-- sound/soc/atmel/sam9g20_wm8731.c | 5 ++++- sound/soc/atmel/sam9x5_wm8731.c | 5 ++++- sound/soc/atmel/tse850-pcm5142.c | 4 +++- sound/soc/fsl/eukrea-tlv320.c | 5 ++++- sound/soc/fsl/fsl-asoc-card.c | 6 +++++- sound/soc/fsl/imx-audmix.c | 14 ++++++++++---- sound/soc/fsl/imx-es8328.c | 5 ++++- sound/soc/fsl/imx-sgtl5000.c | 5 ++++- sound/soc/fsl/imx-spdif.c | 5 ++++- sound/soc/generic/simple-card-utils.c | 15 +++------------ sound/soc/kirkwood/armada-370-db.c | 12 +++++++++--- sound/soc/mxs/mxs-sgtl5000.c | 8 ++++++-- sound/soc/qcom/apq8016_sbc.c | 10 +++++++--- sound/soc/qcom/common.c | 4 +--- sound/soc/qcom/storm.c | 4 +++- sound/soc/rockchip/rk3288_hdmi_analog.c | 5 ++++- sound/soc/rockchip/rk3399_gru_sound.c | 19 +++++++++++++------ sound/soc/rockchip/rockchip_max98090.c | 5 ++++- sound/soc/rockchip/rockchip_rt5645.c | 5 ++++- sound/soc/samsung/arndale_rt5631.c | 6 +++++- sound/soc/samsung/smdk_wm8994.c | 9 +++++++-- sound/soc/samsung/snow.c | 10 +++++++--- sound/soc/samsung/tm2_wm5110.c | 12 +++++++++--- sound/soc/sirf/sirf-audio.c | 5 ++++- sound/soc/soc-utils.c | 25 +++++++++++++++++++++++++ sound/soc/sunxi/sun4i-codec.c | 5 ++++- sound/soc/tegra/tegra_alc5632.c | 7 ++++++- sound/soc/tegra/tegra_max98090.c | 5 ++++- sound/soc/tegra/tegra_rt5640.c | 5 ++++- sound/soc/tegra/tegra_rt5677.c | 6 +++++- sound/soc/tegra/tegra_sgtl5000.c | 7 ++++++- sound/soc/tegra/tegra_wm8753.c | 5 ++++- sound/soc/tegra/tegra_wm8903.c | 5 ++++- sound/soc/tegra/tegra_wm9712.c | 5 ++++- sound/soc/tegra/trimslice.c | 6 +++++- sound/soc/ti/davinci-evm.c | 5 ++++- sound/soc/ti/omap-abe-twl6040.c | 22 ++++++++++++++-------- sound/soc/ti/omap-hdmi.c | 5 ++++- sound/soc/ti/omap-twl4030.c | 12 ++++++++++-- sound/soc/ti/rx51.c | 5 ++++- sound/soc/ux500/mop500.c | 8 ++++++-- 46 files changed, 261 insertions(+), 85 deletions(-) -- 2.7.4

3 95

[alsa-devel] [PATCH 1/2] ASoC: SOF: ipc: update sof_ipc_stream_params
by Keyon Jie 03 Jul '19

03 Jul '19

From: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> The host period bytes needs to be passed to firmware. Currently this field is used as notification for host about period copy completion. Therefore we need to split these two informations. Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- include/sound/sof/stream.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h index 643f175cb479..44acfa62fa69 100644 --- a/include/sound/sof/stream.h +++ b/include/sound/sof/stream.h @@ -83,10 +83,10 @@ struct sof_ipc_stream_params { uint16_t sample_valid_bytes; uint16_t sample_container_bytes; - /* for notifying host period has completed - 0 means no period IRQ */ uint32_t host_period_bytes; + uint16_t no_period_irq; /* 1 means period IRQ mode OFF */ - uint32_t reserved[2]; + uint16_t reserved[3]; uint16_t chmap[SOF_IPC_MAX_CHANNELS]; /**< channel map - SOF_CHMAP_ */ } __packed; -- 2.20.1

2 3

[alsa-devel] [PATCH v2 00/23] ASoC: use devm_xxx
by Kuninori Morimoto 03 Jul '19

03 Jul '19

Hi Mark These are v2 patch of converting for devm_xxx(). v2 is caring about related empty functions. Kuninori Morimoto (23): ASoC: atmel: atmel-pcm-dma.c: use devm_snd_dmaengine_pcm_register() ASoC: sunxi: sun4i-i2s.c: use devm_snd_dmaengine_pcm_register() ASoC: tegra: tegra_pcm.c: use devm_snd_dmaengine_pcm_register() ASoC: ux500: ux500_pcm.c: use devm_snd_dmaengine_pcm_register() ASoC: au1x: i2sc.c: use devm_snd_soc_register_component() ASoC: au1x: ac97c.c: use devm_snd_soc_register_component() ASoC: au1x: psc-ac97.c: use devm_snd_soc_register_component() ASoC: au1x: psc-i2s.c: use devm_snd_soc_register_component() ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() ASoC: cirrus: ep93xx-ac97.c: use devm_snd_soc_register_component() ASoC: rt5682: use devm_snd_soc_register_component() ASoC: ak4118: use devm_snd_soc_register_component() ASoC: cros_ec_codec: use devm_snd_soc_register_component() ASoC: fsl: imx-ssi.c: use devm_snd_soc_register_component() ASoC: fsl: mpc5200_psc_i2s.c: use devm_snd_soc_register_component() ASoC: fsl: mpc5200_psc_ac97.c: use devm_snd_soc_register_component() ASoC: pxa: pxa2xx-ac97.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_i2s.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_spdif.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_ac97.c: use devm_snd_soc_register_component() ASoC: tegra: tegra30_i2s.c: use devm_snd_soc_register_component() ASoC: ti: davinci-i2s.c: use devm_snd_soc_register_component() ASoC: ux500: ux500_msp_dai.c: use devm_snd_soc_register_component() sound/soc/atmel/atmel-pcm-dma.c | 9 ++------- sound/soc/atmel/atmel-pcm-pdc.c | 5 ----- sound/soc/atmel/atmel-pcm.h | 8 -------- sound/soc/atmel/atmel_ssc_dai.c | 11 ----------- sound/soc/au1x/ac97c.c | 4 +--- sound/soc/au1x/i2sc.c | 4 +--- sound/soc/au1x/psc-ac97.c | 6 ++---- sound/soc/au1x/psc-i2s.c | 6 ++---- sound/soc/cirrus/ep93xx-ac97.c | 10 +++------- sound/soc/cirrus/ep93xx-i2s.c | 7 ++----- sound/soc/codecs/ak4118.c | 11 ++--------- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/rt5682.c | 13 +++---------- sound/soc/fsl/imx-ssi.c | 8 ++------ sound/soc/fsl/mpc5200_psc_ac97.c | 3 +-- sound/soc/fsl/mpc5200_psc_i2s.c | 3 +-- sound/soc/pxa/pxa2xx-ac97.c | 3 +-- sound/soc/sunxi/sun4i-i2s.c | 4 +--- sound/soc/tegra/tegra20_ac97.c | 11 +++-------- sound/soc/tegra/tegra20_i2s.c | 11 +++-------- sound/soc/tegra/tegra20_spdif.c | 12 ++++-------- sound/soc/tegra/tegra30_i2s.c | 11 +++-------- sound/soc/tegra/tegra_pcm.c | 11 +++-------- sound/soc/tegra/tegra_pcm.h | 1 - sound/soc/ti/davinci-i2s.c | 8 ++------ sound/soc/ux500/ux500_msp_dai.c | 12 ++---------- sound/soc/ux500/ux500_pcm.c | 9 +-------- sound/soc/ux500/ux500_pcm.h | 1 - 28 files changed, 46 insertions(+), 158 deletions(-) -- 2.7.4

3 34

[alsa-devel] [PATCH] ALSA: line6: Fix write on zero-sized buffer
by Takashi Iwai 02 Jul '19

02 Jul '19

LINE6 drivers allocate the buffers based on the value returned from usb_maxpacket() calls. The manipulated device may return zero for this, and this results in the kmalloc() with zero size (and it may succeed) while the other part of the driver code writes the packet data with the fixed size -- which eventually overwrites. This patch adds a simple sanity check for the invalid buffer size for avoiding that problem. Reported-by: syzbot+219f00fb49874dcaea17(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/pcm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 72c6f8e82a7e..78c2d6cab3b5 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -560,6 +560,11 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->max_packet_size_out = usb_maxpacket(line6->usbdev, usb_sndisocpipe(line6->usbdev, ep_write), 1); + if (!line6pcm->max_packet_size_in || !line6pcm->max_packet_size_out) { + dev_err(line6pcm->line6->ifcdev, + "cannot get proper max packet size\n"); + return -EINVAL; + } spin_lock_init(&line6pcm->out.lock); spin_lock_init(&line6pcm->in.lock); -- 2.16.4

1 0

[alsa-devel] [PATCH v6] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 02 Jul '19

02 Jul '19

The supported formats are S16_LE and S24_LE now. However, S24_LE is not supported when TDM is 0 and it is not in the right justified mode. We should remove 24-bit format in that situation to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- Changed the order of the conditional. Remove the snd_pcm_hw_constraint_msbits function. Use removing 24 bits format instead of fixing at 16 bits format. sound/soc/codecs/max98090.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..9fbb4c31bcf1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,24 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* + * When TDM = 0, remove 24-bit format support if it is not in right + * justified mode. + */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J && + !max98090->tdm_slots) + substream->runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_S24_LE; + + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2334,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.410.gd8fdbe21b5-goog

2 1

[alsa-devel] [PATCH -next] ASoC: madera: Remove duplicated include from cs47l35.c
by YueHaibing 02 Jul '19

02 Jul '19

Remove duplicated include. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs47l35.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 511d0d6fa962..e3585c1dab3d 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -19,7 +19,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/tlv.h> -#include <sound/tlv.h> #include <linux/irqchip/irq-madera.h> #include <linux/mfd/madera/core.h>

3 2

[alsa-devel] [PATCH -next] ASoC: rt1308: Fix platform_no_drv_owner.cocci warnings
by YueHaibing 02 Jul '19

02 Jul '19

Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt1308.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 619a630e982e..d673506c7c39 100755 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -884,7 +884,6 @@ static void rt1308_i2c_shutdown(struct i2c_client *client) static struct i2c_driver rt1308_i2c_driver = { .driver = { .name = "rt1308", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(rt1308_of_match), .acpi_match_table = ACPI_PTR(rt1308_acpi_match), },

2 1

[alsa-devel] Applied "ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 02 Jul '19

02 Jul '19

The patch ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cb9002a186329b35f32f74021551aecbfe9a696c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 28 Jun 2019 13:09:00 +0900 Subject: [PATCH] ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() We have devm_xxx version of snd_soc_register_component, let's use it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/cirrus/ep93xx-i2s.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 6ca899ba9484..0b4355e95f84 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -473,19 +473,17 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, info); - err = snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component, + err = devm_snd_soc_register_component(&pdev->dev, &ep93xx_i2s_component, &ep93xx_i2s_dai, 1); if (err) goto fail_put_lrclk; err = devm_ep93xx_pcm_platform_register(&pdev->dev); if (err) - goto fail_unregister; + goto fail_put_lrclk; return 0; -fail_unregister: - snd_soc_unregister_component(&pdev->dev); fail_put_lrclk: clk_put(info->lrclk); fail_put_sclk: @@ -500,7 +498,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev) { struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); clk_put(info->lrclk); clk_put(info->sclk); clk_put(info->mclk); -- 2.20.1

1 0

[alsa-devel] [RFC PATCH 0/5] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 02 Jul '19

02 Jul '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support and module_sdw_driver macro. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Srinivas Kandagatla (5): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices soundwire: add module_sdw_driver helper macro dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.txt | 27 + .../devicetree/bindings/soundwire/bus.txt | 48 + drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 54 +- include/linux/soundwire/sdw_type.h | 11 + sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1160 +++++++++++++++++ 9 files changed, 1312 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.txt create mode 100644 Documentation/devicetree/bindings/soundwire/bus.txt create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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[alsa-devel] [PATCH v3] ALSA: hda: Fix widget_mutex incomplete protection
by Evan Green 01 Jul '19

01 Jul '19

The widget_mutex was introduced to serialize callers to hda_widget_sysfs_{re}init. However, its protection of the sysfs widget array is incomplete. For example, it is acquired around the call to hda_widget_sysfs_reinit(), which actually creates the new array, but isn't still acquired when codec->num_nodes and codec->start_nid is updated. So the lock ensures one thread sets up the new array at a time, but doesn't ensure which thread's value will end up in codec->num_nodes. If a larger num_nodes wins but a smaller array was set up, the next call to refresh_widgets() will touch free memory as it iterates over codec->num_nodes that aren't there. The widget_lock really protects both the tree as well as codec->num_nodes, start_nid, and end_nid, so make sure it's held across that update. It should also be held during snd_hdac_get_sub_nodes(), so that a very old read from that function doesn't end up clobbering a later update. Fixes: ed180abba7f1 ("ALSA: hda: Fix race between creating and refreshing sysfs entries") Signed-off-by: Evan Green <evgreen(a)chromium.org> --- Changes in v3: - Moved locking back into callers (Takashi) - Combined update_widgets exit flow (Takashi) Changes in v2: - Introduced widget_mutex relocation sound/hda/hdac_device.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6907dbefd08c..3842f9d34b7c 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -400,27 +400,33 @@ static void setup_fg_nodes(struct hdac_device *codec) int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs) { hda_nid_t start_nid; - int nums, err; + int nums, err = 0; + /* + * Serialize against multiple threads trying to update the sysfs + * widgets array. + */ + mutex_lock(&codec->widget_lock); nums = snd_hdac_get_sub_nodes(codec, codec->afg, &start_nid); if (!start_nid || nums <= 0 || nums >= 0xff) { dev_err(&codec->dev, "cannot read sub nodes for FG 0x%02x\n", codec->afg); - return -EINVAL; + err = -EINVAL; + goto unlock; } if (sysfs) { - mutex_lock(&codec->widget_lock); err = hda_widget_sysfs_reinit(codec, start_nid, nums); - mutex_unlock(&codec->widget_lock); if (err < 0) - return err; + goto unlock; } codec->num_nodes = nums; codec->start_nid = start_nid; codec->end_nid = start_nid + nums; - return 0; +unlock: + mutex_unlock(&codec->widget_lock); + return err; } EXPORT_SYMBOL_GPL(snd_hdac_refresh_widgets); -- 2.20.1

2 1

[alsa-devel] [PATCH v2 0/2] ALSA: hda: Widget memory fixes
by Evan Green 01 Jul '19

01 Jul '19

This series fixes concurrency issues with the sysfs widget array. The first function patches up the locking that was introduced recently to protect more of the data structure. The second patch fixes a race between a reinit and the initial population of the array which could result in a length and array getting out of sync. Changes in v2: - Introduced widget_mutex relocation Evan Green (2): ALSA: hda: Fix widget_mutex incomplete protection ALSA: hda: Use correct start/count for sysfs init sound/hda/hdac_device.c | 21 ++++++++++++++------- sound/hda/hdac_sysfs.c | 18 ++++++++++-------- sound/hda/local.h | 3 ++- 3 files changed, 26 insertions(+), 16 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH v2] ALSA: firewire-lib/fireworks: fix miss detection of received MIDI messages
by Takashi Sakamoto 01 Jul '19

01 Jul '19

In IEC 61883-6, 8 MIDI data streams are multiplexed into single MIDI conformant data channel. The index of stream is calculated by modulo 8 of the value of data block counter. In fireworks, the value of data block counter in CIP header has a quirk with firmware version v5.0.0, v5.7.3 and v5.8.0. This brings ALSA IEC 61883-1/6 packet streaming engine to miss detection of MIDI messages. This commit fixes the miss detection to modify the value of data block counter for the modulo calculation. For maintainers, this bug exists since a commit 18f5ed365d3f ("ALSA: fireworks/firewire-lib: add support for recent firmware quirk") in Linux kernel v4.2. There're many changes since the commit. This fix can be backported to Linux kernel v4.4 or later. I tagged a base commit to the backport for your convenience. Besides, my work for Linux kernel v5.3 brings heavy code refactoring and some structure members are renamed in 'sound/firewire/amdtp-stream.h'. The content of this patch brings conflict when merging -rc tree with this patch and the latest tree. I request maintainers to solve the conflict to replace 'tx_first_dbc' with 'ctx_data.tx.first_dbc'. Fixes: df075feefbd3 ("ALSA: firewire-lib: complete AM824 data block processing layer") Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-am824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index bebddc60fde8..99654e7eb2d4 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -388,7 +388,7 @@ static void read_midi_messages(struct amdtp_stream *s, u8 *b; for (f = 0; f < frames; f++) { - port = (s->data_block_counter + f) % 8; + port = (8 - s->tx_first_dbc + s->data_block_counter + f) % 8; b = (u8 *)&buffer[p->midi_position]; len = b[0] - 0x80; -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib/fireworks: fix miss detection of received MIDI messages
by Takashi Sakamoto 01 Jul '19

01 Jul '19

In IEC 61883-6, 8 MIDI data streams are multiplexed into single MIDI conformant data channel. The index of stream is calculated by modulo 8 of the value of data block counter. In fireworks, the value of data block counter in CIP header has a quirk with firmware version v5.0.0, v5.7.3 and v5.8.0. This brings ALSA IEC 61883-1/6 packet streaming engine to miss detection of MIDI messages. This commit fixes the miss detection to modify the value of data block counter for the modulo calculation. For maintainers, this bug exists since a commit 18f5ed365d3f ("ALSA: fireworks/firewire-lib: add support for recent firmware quirk") in Linux kernel v4.2. There're many changes since the commit. This fix can be backported to Linux kernel v4.4 or later. I tagged a base commit to the backport for your convenience. Besides, my work for Linux kernel v5.3 brings heavy code refactoring and some structure members are renamed in 'sound/firewire/amdtp-stream.h'. The content of this patch brings conflict when merging -rc tree with this patch to the latest tree. I request maintainers to solve the conflict by replacing 'tx_first_dbc' with 'ctx_data.tx.first_dbc'. Fixes: df075feefbd3 ("ALSA: firewire-lib: complete AM824 data block processing layer") Cc: <stable(a)vger.kernel.org> # v4.4+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-am824.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index 4210e5c6262e..4d677fcb4fc2 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -321,6 +321,7 @@ static void read_midi_messages(struct amdtp_stream *s, u8 *b; for (f = 0; f < frames; f++) { + port = (8 - s->tx_first_dbc + s->data_block_counter + f) % 8; port = (s->data_block_counter + f) % 8; b = (u8 *)&buffer[p->midi_position]; -- 2.20.1

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[alsa-devel] ALSA repos
by Daniel Baluta 01 Jul '19

01 Jul '19

Hi Jaroslav, What are trying to understand where are the official ALSA project repos. We found: - Github: https://github.com/alsa-project and - alsa-project.org: https://git.alsa-project.org/ Which one should we use for our daily builds? For example we have been using alsa-project.org repo for tinycompress. But in the last few days git clone didn't work! Now it seems to be back online. Also, we are using Yocto poky recipes: http://git.yoctoproject.org/cgit.cgi/poky/tree/meta/recipes-multimedia/alsa… which use pre-build archives from: https://www.alsa-project.org/files/pub/ So, any clarifications on the release deliverables for ALSA project would be highly appreciated. thanks, Daniel.

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix a headphone detection issue when using SOF
by Rander Wang 01 Jul '19

01 Jul '19

To save power, the hda hdmi driver in ASoC invokes snd_hdac_ext_bus_link_put to disable CORB/RIRB buffers DMA if there is no user of bus and invokes snd_hdac_ext_bus_link_get to set up CORB/RIRB buffers when it is used. Unsolicited responses is disabled in snd_hdac_bus_stop_cmd_io called by snd_hdac_ext_bus_link_put , but it is not enabled in snd_hdac_bus_init_cmd_io called by snd_hdac_ext_bus_link_get. So for put-get sequence, Unsolicited responses is disabled and headphone can't be detected by hda codecs. Now unsolicited responses is only enabled in snd_hdac_bus_reset_link which resets controller. The function is only called for setup of controller. This patch enables Unsolicited responses after RIRB is initialized in snd_hdac_bus_init_cmd_io which works together with snd_hdac_bus_reset_link to set up controller. Tested legacy hda driver and SOF driver on intel whiskeylake. Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> --- sound/hda/hdac_controller.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index d6a91429c058..c24fc8d266a9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -78,6 +78,8 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) snd_hdac_chip_writew(bus, RINTCNT, 1); /* enable rirb dma and response irq */ snd_hdac_chip_writeb(bus, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN); + /* Accept unsolicited responses */ + snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, AZX_GCTL_UNSOL); spin_unlock_irq(&bus->reg_lock); } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_cmd_io); @@ -416,9 +418,6 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) return -EBUSY; } - /* Accept unsolicited responses */ - snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, AZX_GCTL_UNSOL); - /* detect codecs */ if (!bus->codec_mask) { bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); -- 2.14.1

2 1

[alsa-devel] why error return in sound/soc/codecs/pcm3168a.c::pcm3168a_set_dai_fmt ?
by Jurgen Lambrecht 01 Jul '19

01 Jul '19

Hi, When following https://bootlin.com/blog/eight-channels-audio-on-i-mx7-with-pcm3168/ I get this error (from soc_core.c): ... kernel: [ 3.059634][ T15] pcm3168a 3-0039: ASoC: Failed to set DAI format: -22 I could propose this patch (I get that printf when applying the patch) (all other errors in that file have a printf): diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 08d3fe192e65..75ccbb919902 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -353,6 +353,7 @@ static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, case SND_SOC_DAIFMT_NB_NF: break; default: + dev_err(component->dev, "wrong INV_MASK\n"); return -EINVAL; } But why does this code returns an error for code that has no effect? OK, it could be because SND_SOC_DAIFMT_INV_MASK must be SND_SOC_DAIFMT_NB_NF. But then Alexandre's blog contains an error, because the dac TDM sets frame-inversion in its dts (and it should not). -- Kind Regards, *Jürgen Lambrecht* R&D Associate Tel: +32 51 303045 www.televic-rail.com <https://www.televic-rail.com> Televic Rail NV - Leo Bekaertlaan 1 - 8870 Izegem - Belgium Company number 0825 539 581 - RPR Kortrijk

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Re: [alsa-devel] [BUG] Line6 Helix / X Stomp: parse_audio_format_rates_v2v3(): unable to retrieve number of sample rates (clock 16)
by nick83ola 01 Jul '19

01 Jul '19

On Thu, 27 Jun 2019 at 17:22, Jens Verwiebe <info(a)jensverwiebe.de> wrote: > > Hi Nick > > This looks like the same hurdle i once had with Helix. > > Imho you should first try my patch to format.c and extend the > vendor/model clause to your device: > > diff --git a/sound/usb/format.c b/sound/usb/format.c > index 2c44386..1c0107d 100644 > --- a/sound/usb/format.c > +++ b/sound/usb/format.c > @@ -298,6 +298,20 @@ static int parse_audio_format_rates_v2(struct > snd_usb_audio *chip, > goto err; > } > > + /* > + * Line6 HELIX does not respond to sample rate > + * query requests. The only valid rate is 48000. > + */ > + if (chip->usb_id == USB_ID(0x0e41, 0x4244)) { > + fp->nr_rates = 1; > + fp->rate_min = 48000; > + fp->rate_max = 48000; > + fp->rates = SNDRV_PCM_RATE_48000; > + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); > + fp->rate_table[0] = 48000; > + return 0; > + } > + > /* get the number of sample rates first by only fetching 2 bytes */ > ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, > USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, > I can confirm that your patch get the device recognized in kernel 5.2 (I modfied it a bit). I'll work on this a bit and try to submit a patch to alsa to at least having the device available. Regarding the second part ( get_usb_high_speed_rate(rate + 5);) I think I have to debug it better because I don't think it can be accepted upstream. (maybe someone from the mailing list can help?) > I personally encountered still slight pops from time to time which i > "healed" with an ugly second patch: > > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index c90607e..026095a 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -878,6 +878,8 @@ int snd_usb_endpoint_set_params(struct > snd_usb_endpoint *ep, > > if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) > ep->freqn = get_usb_full_speed_rate(rate); > + else if (USB_ID(0x0e41, 0x4244)) > + ep->freqn = get_usb_high_speed_rate(rate + 5); // ugly helix hack > else > ep->freqn = get_usb_high_speed_rate(rate); > > > I still never could get the Line6 device to really follow the > syncadaption in 16.16 format. > > I tried all kinda voodoo like endpoint redirection ( like axe fx needs ) > to no avail. Perhaps you have better ideas. > > I would be very interested in getting Line6 devices usb-compliant mode > working 100 % reliable. > > The dream would be to have all samplerates. Btw:: i have sniffed data > from Helix connected to a windows machine, > > ioreg data from helix connected to an OSX(macOS) machine and even a bit > experimented with retdec on the windows drivers ;). > > This for now, 'am just on the jump to the Hamburg Linux Group :) > > Lets stay in contact about this stuff. > > Cheers ... Jens > > > Am 27.06.19 um 09:45 schrieb nick83ola: > > Hi all, > > I just purchased an HX stomp and when I connect to linux I get the > > following error message: > > > > [ 322.404503] usb 1-6: new high-speed USB device number 13 using xhci_hcd > > [ 322.531804] usb 1-6: New USB device found, idVendor=0e41, > > idProduct=4246, bcdDevice= 2.00 > > [ 322.531806] usb 1-6: New USB device strings: Mfr=1, Product=3, > > SerialNumber=2 > > [ 322.531806] usb 1-6: Product: HX Stomp > > [ 322.531807] usb 1-6: Manufacturer: LINE 6 > > [ 322.531808] usb 1-6: SerialNumber: 3021888 > > [ 322.535789] usb 1-6: parse_audio_format_rates_v2v3(): unable to > > retrieve number of sample rates (clock 16) > > [ 322.535907] usb 1-6: parse_audio_format_rates_v2v3(): unable to > > retrieve number of sample rates (clock 16) > > > > I have some kernel development experience but not a lot with usb so > > can someone point me where to start looking? > > > > there was a previous discussion on the mailing list related to this by > > Jens Verwiebe (and other request for help). > > Here are the relevant patches/discussion > > > > https://www.spinics.net/lists/alsa-devel/msg71459.html > > https://patchwork.kernel.org/patch/10113121/ > > > > If someone can help me I have an HX stomp here and time to do some debugging. > > > > Also if someone from Line6 is listening would be helpful to have some hints. > > > > Cheers > > Nicola Lunghi > > > -- > > Jens Verwiebe

1 0

[alsa-devel] [PATCH] ALSA: xen-front: fix unintention integer overflow on left shifts
by Colin King 01 Jul '19

01 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> Shifting the integer value 1 is evaluated using 32-bit arithmetic and then used in an expression that expects a 64-bit value, so there is potentially an integer overflow. Fix this by using the BIT_ULL macro to perform the shift. Addresses-Coverity: ("Unintentional integer overflow") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/xen/xen_snd_front_alsa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index b14ab512c2ce..e01631959ed8 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -196,7 +196,7 @@ static u64 to_sndif_formats_mask(u64 alsa_formats) mask = 0; for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) if (pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa) & alsa_formats) - mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif; + mask |= BIT_ULL(ALSA_SNDIF_FORMATS[i].sndif); return mask; } @@ -208,7 +208,7 @@ static u64 to_alsa_formats_mask(u64 sndif_formats) mask = 0; for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) - if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats) + if (BIT_ULL(ALSA_SNDIF_FORMATS[i].sndif) & sndif_formats) mask |= pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa); return mask; -- 2.20.1

3 2

[alsa-devel] [PATCH RFC V3] ALSA: usb-audio: Scarlett Gen 2 mixer interface
by Geoffrey D. Bennett 30 Jun '19

30 Jun '19

Add mixer quirk for the Focusrite Scarlett 6i6, 18i8, and 18i20 Gen 2 audio interfaces. Although the interfaces are USB compliant, additional input/output level controls and hardware routing/mixing functionality are available using proprietary USB requests. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- Hi all, This patch adds the following controls for the Scarlett 6i6, 18i8, and 18i20 Gen 2: - Master volume knob indicator (18i20 only) - Mute and dim switches (18i20 only) - Volume controls for the analogue HW outputs - HW/SW volume switches for the 10 analogue HW outputs (18i20 only) - Line Level/Instrument Level and Pad controls (6i6 and 18i8 only) - Output mux (where the sound for the HW outputs comes from; defaults to PCM outputs) - Capture mux (where the sound for PCM recording comes from; defaults to HW inputs) - Matrix mux (where the sound going into the mixer comes from; 18 inputs default off) - Mixer matrix (18 inputs * 10 outputs = 180 controls) - Level meters Changes since v1: - Add support for the Scarlett 18i8 Gen 2 - Save configuration parameters to NVRAM - Implemented feedback from Takashi's email 24/Apr/2019 - Moved private field from struct snd_usb_audio to struct usb_mixer_interface - Added timer and buffer fields to struct usb_mixer_interface - Other small code fixes/cleanups/improvements Changes since v2: - Add support for the Scarlett 6i6 Gen 2 - Add support for controlling the mute/dim switches (18i20 only) - Implemented feedback from Takashi's email 29/Apr/2019: One pointer field private_data and one function pointer private_free. Replaced timer with delayed work (thanks for that suggestion; it makes the code a *lot* simpler!). - The new functionality is disabled by default as there were reports of some Gen 2 devices not responding to the initialisation sequence and hanging. Added module parameter "scarlett_gen2_mixer_enable". If this parameter is not set, logs a message "Focusrite Scarlett Gen 2 Mixer Driver disabled; use options snd_usb_audio scarlett_gen2_mixer_enable=1 to enable and report any issues to g(a)b4.vu". Outstanding issues/questions: - Takashi, you wrote last time "Also, the delayed work needs to be canceled or flushed at PM suspend callback as well as disconnecting." I'm not sure why it should be cancelled; is it not okay to do on resume? But I tested suspending while the delayed work was waiting and it seemed like the delayed work was cancelled anyway. Probably better to do a flush on suspend though; should this be done through a hook like: diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2444737a14b2..1572a89267c6 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3547,6 +3547,8 @@ static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) { snd_usb_mixer_inactivate(mixer); + if (mixer->private_suspend) + mixer->private_suspend(mixer); return 0; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index fa6e216a06f0..d94c688c65f7 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -28,6 +28,7 @@ struct usb_mixer_interface { void *private_data; void (*private_free)(struct usb_mixer_interface *private_data); + void (*private_suspend)(struct usb_mixer_interface *); }; #define MAX_CHANNELS 16 /* max logical channels */ ? Thanks, Geoffrey. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index d330f74c90e6..8cf7081e17cb 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,6 +11,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 53dccbfe392b..2444737a14b2 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3518,6 +3518,8 @@ void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) usb_kill_urb(mixer->urb); if (mixer->rc_urb) usb_kill_urb(mixer->rc_urb); + if (mixer->private_free) + mixer->private_free(mixer); mixer->disconnected = true; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3d12af8bf191..fa6e216a06f0 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -25,6 +25,9 @@ struct usb_mixer_interface { u8 rc_buffer[6]; bool disconnected; + + void *private_data; + void (*private_free)(struct usb_mixer_interface *); }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a751a18ca4c2..1ae00a1e6c47 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -45,6 +45,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -2271,6 +2272,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */ + case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..05a8e47a85f4 --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,2078 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 6i6/18i8/18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett <g at b4.vu> + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +/* Mixer Interface for the Focusrite Scarlett 6i6/18i8/18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * Scarlett 18i8 support added in April 2019. + * + * Scarlett 6i6 support added in June 2019 (thanks to Martin Wittmann + * for providing usbmon output and testing). + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * - line/inst level and pad controls + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ Software gain per channel. + * | Master Gain |<-- 18i20 only: Switch per channel + * +------+------+ to select HW or SW gain control. + * | + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +static int scarlett_gen2_mixer_enable = 0; +module_param(scarlett_gen2_mixer_enable, int, 0444); +MODULE_PARM_DESC(scarlett_gen2_mixer_enable, + "Focusrite Scarlett Gen 2 Mixer Driver Enable"); + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT_MIXER_MIN_DB -80 +#define SCARLETT_MIXER_BIAS (-SCARLETT_MIXER_MIN_DB * 2) +#define SCARLETT_MIXER_MAX_DB 6 +#define SCARLETT_MIXER_MAX_VALUE \ + ((SCARLETT_MIXER_MAX_DB - SCARLETT_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_ANALOGUE_MAX 10 + +/* Maximum number of level and pad switches */ +#define SCARLETT2_LEVEL_SWITCH_MAX 2 +#define SCARLETT2_PAD_SWITCH_MAX 4 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +/* Hardware buttons on the 18i20 */ +#define SCARLETT2_BUTTON_MAX 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + u8 line_out_hw_vol; /* line out hw volume is sw controlled */ + u8 button_count; /* number of buttons */ + u8 level_input_count; /* inputs with level selectable */ + u8 pad_input_count; /* inputs with pad selectable */ + const char * const line_out_descrs[SCARLETT2_ANALOGUE_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_mixer_data { + struct usb_mixer_interface *mixer; + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + struct delayed_work work; + const struct scarlett2_device_info *info; + int num_mux_srcs; + u16 scarlett2_seq; + u8 vol_updated; + u8 master_vol; + u8 vol[SCARLETT2_ANALOGUE_MAX]; + u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + u8 level_switch[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; + u8 buttons[SCARLETT2_BUTTON_MAX]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_BUTTON_MAX]; + u8 mux[SCARLETT2_MUX_MAX]; + u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +static void snd_scarlett_gen2_private_free(struct usb_mixer_interface *); + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s6i6_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first two analogue inputs have an optional pad. */ + .pad_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 8 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 4, 4, 4, 4, 4 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 6, 6, 6, 6, 6 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i8_gen2_info = { + /* The first two analogue inputs can be switched between line + * and instrument levels. + */ + .level_input_count = 2, + + /* The first four analogue inputs have an optional pad. */ + .pad_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 4 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 6, 6, 6, 6 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 0, 0, 0, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +static const struct scarlett2_device_info s18i20_gen2_info = { + /* The analogue line outputs on the 18i20 can be switched + * between software and hardware volume control + */ + .line_out_hw_vol = 1, + + /* Mute and dim buttons */ + .button_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_CONFIG_SAVE 6 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* volume status is read together (matches scarlett2_config_items[]) */ +struct scarlett2_usb_volume_status { + /* mute & dim buttons */ + u8 buttons[SCARLETT2_BUTTON_MAX]; + + u8 pad1; + + /* software volume setting */ + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + + /* actual volume of output inc. dim (-18dB) */ + s16 hw_vol[SCARLETT2_ANALOGUE_MAX]; + + u8 pad2[SCARLETT2_ANALOGUE_MAX]; + + /* sw (0) or hw (1) controlled */ + u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; + + u8 pad3[6]; + + /* front panel volume knob */ + s16 master_vol; +} __packed; + +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_BUTTONS = 0, + SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, + SCARLETT2_CONFIG_SW_HW_SWITCH = 2, + SCARLETT2_CONFIG_LEVEL_SWITCH = 3, + SCARLETT2_CONFIG_PAD_SWITCH = 4, + SCARLETT2_CONFIG_COUNT = 5 +}; + +/* Location, size, and activation command number for the configuration + * parameters + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +static const struct scarlett2_config + scarlett2_config_items[SCARLETT2_CONFIG_COUNT] = { + /* Mute/Dim Buttons */ + { + .offset = 0x31, + .size = 1, + .activate = 2 + }, + + /* Line Out Volume */ + { + .offset = 0x34, + .size = 2, + .activate = 1 + }, + + /* SW/HW Volume Switch */ + { + .offset = 0x66, + .size = 1, + .activate = 3 + }, + + /* Level Switch */ + { + .offset = 0x7c, + .size = 1, + .activate = 7 + }, + + /* Pad Switch */ + { + .offset = 0x84, + .size = 1, + .activate = 8 + } +}; + +/* proprietary request/response format */ +struct scarlett2_usb_packet { + u32 cmd; + u16 size; + u16 seq; + u32 error; + u32 pad; + u8 data[]; +}; + +#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) + +static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, + struct scarlett2_usb_packet *req, + u32 cmd, u16 req_size) +{ + /* sequence must go up by 1 for each request */ + u16 seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; +} + +/* Send a proprietary format request to the Scarlett interface */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, u32 cmd, + void *req_data, u16 req_size, void *resp_data, u16 resp_size) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; + u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; + + struct scarlett2_usb_packet *req = 0, *resp = 0; + + int err = 0; + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + err = -ENOMEM; + goto unlock; + } + + scarlett2_fill_request_header(private, req, cmd, req_size); + + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + if (err != req_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB request result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + /* send a second message to get the response */ + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + err = -ENOMEM; + goto unlock; + } + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB response result cmd %x was %d\n", + cmd, err); + err = -EINVAL; + goto unlock; + } + + if (resp->cmd != req->cmd || + resp->seq != req->seq || + resp_size != le16_to_cpu(resp->size) || + resp->error || + resp->pad) { + usb_audio_err( + mixer->chip, + "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd), + le16_to_cpu(req->seq), le16_to_cpu(resp->seq), + resp_size, le16_to_cpu(resp->size), + le16_to_cpu(resp->error), + le16_to_cpu(resp->pad)); + err = -EINVAL; + goto unlock; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + +unlock: + kfree(req); + kfree(resp); + mutex_unlock(&private->usb_mutex); + return err; +} + +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct work_struct *work) +{ + struct scarlett2_mixer_data *private = + container_of(work, struct scarlett2_mixer_data, work.work); + struct usb_mixer_interface *mixer = private->mixer; + u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); + + scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(u32), + NULL, 0); +} + +/* Send a USB message to set a configuration parameter (volume level, + * sw/hw volume switch, line/inst level switch, or pad switch) + */ +static int scarlett2_usb_set_config( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int value) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + struct { + u32 offset; + u32 bytes; + s32 value; + } __packed req; + u32 req2; + int err; + struct scarlett2_mixer_data *private = mixer->private_data; + + /* Send the configuration parameter data */ + req.offset = cpu_to_le32(config_item.offset + index * config_item.size); + req.bytes = cpu_to_le32(config_item.size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + config_item.size, + NULL, 0); + if (err < 0) + return err; + + /* Activate the change */ + req2 = cpu_to_le32(config_item.activate); + err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + /* Schedule the change to be written to NVRAM */ + schedule_delayed_work(&private->work, msecs_to_jiffies(2000)); + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + int offset, void *buf, int size) +{ + struct { + u32 offset; + u32 size; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + return scarlett2_usb(mixer, SCARLETT2_USB_GET_DATA, + &req, sizeof(req), buf, size); +} + +/* Send a USB message to get configuration parameters; result placed in *buf */ +static int scarlett2_usb_get_config( + struct usb_mixer_interface *mixer, + int config_item_num, int count, void *buf) +{ + const struct scarlett2_config config_item = + scarlett2_config_items[config_item_num]; + int size = config_item.size * count; + + return scarlett2_usb_get(mixer, config_item.offset, buf, size); +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + struct { + u16 mix_num; + u16 data[SCARLETT2_INPUT_MIX_MAX]; + } __packed req; + + int i, j; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + return scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(u16), + NULL, 0); +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct { + u16 pad; + u16 num; + u32 data[SCARLETT2_MUX_MAX]; + } __packed req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + u16 *levels) +{ + struct { + u16 pad; + u16 num_meters; + u32 magic; + } __packed req; + u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* Update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < private->info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol[index] = val; + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, val - SCARLETT_VOLUME_BIAS); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_gain, -SCARLETT_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + index, private->master_vol - SCARLETT_VOLUME_BIAS); + if (err < 0) + goto unlock; + + /* Notify of RO/RW change */ + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Line Level/Instrument Level Switch Controls ***/ + +static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Line", "Inst" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->level_switch[elem->control]; + return 0; +} + +static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->level_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->level_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_level_enum_ctl_info, + .get = scarlett2_level_enum_ctl_get, + .put = scarlett2_level_enum_ctl_put, +}; + +/*** Pad Switch Controls ***/ + +static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = + private->pad_switch[elem->control]; + return 0; +} + +static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->pad_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pad_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_pad_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_pad_ctl_get, + .put = scarlett2_pad_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + cancel_delayed_work_sync(&private->work); + mutex_lock(&private->data_mutex); + + oval = private->buttons[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->buttons[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, + index, val); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get, + .put = scarlett2_button_ctl_put +}; + +/*** Create the analogue output controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), "Master HW Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_master_volume_ctl, + 0, 1, s, &private->master_vol_ctl); + if (err < 0) + return err; + } + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(s, sizeof(s), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(s, sizeof(s), + "Line %02d Playback Volume", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_line_out_volume_ctl, + i, 1, s, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + if (info->line_out_hw_vol) { + snprintf(s, sizeof(s), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_sw_hw_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + /* Add HW button controls */ + for (i = 0; i < private->info->button_count; i++) { + err = scarlett2_add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Create the analogue input controls ***/ + +static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add input level (line/inst) controls */ + for (i = 0; i < info->level_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Level Capture Enum", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_level_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + /* Add input pad controls */ + for (i = 0; i < info->pad_input_count; i++) { + snprintf(s, sizeof(s), "Line In %d Pad Capture Switch", i + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_pad_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) + goto unlock; + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(mixer, mix_num); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_mixer, + SCARLETT_MIXER_MIN_DB * 100, + SCARLETT_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(s, sizeof(s), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_mixer_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_mixer_data *private = mixer->private_data; + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) + goto unlock; + + private->mux[index] = val; + err = scarlett2_usb_set_mux(mixer); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(s, sizeof(s) - 5, descr, channel + 1); + strcat(s, " Enum"); + + err = scarlett2_add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, s, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, + "Level Meter", NULL); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_mixer_data *private = + kzalloc(sizeof(struct scarlett2_mixer_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + INIT_DELAYED_WORK(&private->work, scarlett2_config_save); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + private->mixer = mixer; + mixer->private_data = private; + mixer->private_free = snd_scarlett_gen2_private_free; + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, + NULL, 0, NULL, 0); +} + +/* Read line-in config and line-out volume settings on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; + u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + if (info->level_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count, + level_switches); + if (err < 0) + return err; + for (i = 0; i < info->level_input_count; i++) + private->level_switch[i] = level_switches[i]; + } + + if (info->pad_input_count) { + err = scarlett2_usb_get_config( + mixer, + SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, + pad_switches); + if (err < 0) + return err; + for (i = 0; i < info->pad_input_count; i++) + private->pad_switch[i] = pad_switches[i]; + } + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = + info->line_out_hw_vol + && volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < info->button_count; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + int i; + + private->vol_updated = 1; + + for (i = 0; i < private->info->button_count; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + usb_audio_err(mixer->chip, + "scarlett mixer interrupt length %d\n", len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + void *transfer_buffer; + + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb( + mixer->urb, + mixer->chip->dev, + usb_rcvintpipe(mixer->chip->dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT), + transfer_buffer, + SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, + mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8203): + info = &s6i6_gen2_info; + break; + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + if (!scarlett_gen2_mixer_enable) { + usb_audio_err(mixer->chip, + "Focusrite Scarlett Gen 2 Mixer Driver disabled; use " + "options snd_usb_audio scarlett_gen2_mixer_enable=1 " + "to enable and report any issues to g(a)b4.vu"); + return 0; + } + + /* Initialise private data, routing, sequence number */ + err = scarlett2_init_private(mixer, info); + if (err < 0) + return err; + + /* Read volume levels and controls from the interface */ + err = scarlett2_read_configs(mixer); + if (err < 0) + return err; + + /* Create the analogue output controls */ + err = scarlett2_add_line_out_ctls(mixer); + if (err < 0) + return err; + + /* Create the analogue input controls */ + err = scarlett2_add_line_in_ctls(mixer); + if (err < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + err = scarlett2_add_mux_enums(mixer); + if (err < 0) + return err; + + /* Create the matrix mixer controls */ + err = scarlett2_add_mixer_ctls(mixer); + if (err < 0) + return err; + + /* Create the level meter controls */ + err = scarlett2_add_meter_ctl(mixer); + if (err < 0) + return err; + + /* Set up the interrupt polling if there are hardware buttons */ + if (info->button_count) { + err = scarlett2_mixer_status_create(mixer); + if (err < 0) + return err; + } + + return 0; +} + +static void snd_scarlett_gen2_private_free(struct usb_mixer_interface *mixer) +{ + struct scarlett2_mixer_data *private = mixer->private_data; + + cancel_delayed_work_sync(&private->work); + kfree(private); + mixer->private_data = NULL; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */

1 0

[alsa-devel] focusrite scarlett 18i20 : Mixer controls with corrupted names for
by Stefan Sauer 29 Jun '19

29 Jun '19

Hi, when I run amixer -D hw:4 controls | sort -n -t = -k2 I get the output below numid=1,iface=MIXER,name='KKKKKKKKKKKKKÃÃÃÃÃÃÃÃÃÃÃ Switch' numid=2,iface=CARD,name='Internal Validity' numid=3,iface=CARD,name='S/PDIF Validity' numid=4,iface=CARD,name='ADAT Validity' numid=5,iface=MIXER,name='Scarlett 18i20 USB-Sync Clock Source' numid=6,iface=MIXER,name=' Switch' numid=7,iface=CARD,name='Keep Interface' numid=8,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=10,iface=MIXER,name='Master 1 (Monitor) Playback Switch' numid=11,iface=MIXER,name='Master 1 (Monitor) Playback Volume' ... lots of extra lines ... Please note the lines for numid=1 and numid=6. The first one contains some garbage and the 2nd one look like it should start with another word. I now added some debug printing here: https://github.com/torvalds/linux/blob/master/sound/usb/mixer_scarlett.c#L5… and get output in dmesg: [ 2971.642137] usb 1-2: Product: Scarlett 18i20 USB [ 2971.642141] usb 1-2: Manufacturer: Focusrite [ 2971.709773] new ctrl: name='Master Playback Switch', index=10, offset=1, num=0: numid=8 [ 2971.709781] new ctrl: name='Master Playback Volume', index=10, offset=2, num=0: numid=9 [ 2971.709788] new ctrl: name='Master 1 (Monitor) Playback Switch', index=10, offset=1, num=1: numid=10 So the first 8 controls are added somewhere else. Looks like this is from mixer.c and after echo -n 'file sound/usb/mixer.c +p' >/sys/kernel/debug/dynamic_debug/control I get [ 4405.855432] usb 1-2: [51] PU [KKKKKKKKKKKKKÃÃÃÃÃÃÃÃÃÃÃ Switch] ch = 1, val = 0/1 [ 4405.856423] usb 1-2: [52] PU [ Switch] ch = 1, val = 0/1 I now added more debug prints into https://github.com/torvalds/linux/blob/master/sound/usb/mixer.c#L2431 and its the code that calls snd_usb_copy_string_desc() [ 5750.124123] usb 1-2: nameid=90, len=35 [ 5750.124157] usb 1-2: [51] PU [KKKKKKKKKKKKKÃÃÃÃÃÃÃÃÃÃÃ Switch] ch = 1, val = 0/1 [ 5750.125241] usb 1-2: nameid=82, len=1 [ 5750.125260] usb 1-2: [52] PU [ Switch] ch = 1, val = 0/1 In both cases the returned len seems wrong and the function does not seems to copy any useful string here. snd_usb_copy_string_desc() is just a wrapper around usb_string(). Is my hardware returning bogus data in usb descriptors? Can this be address through some quirks table? Any other ideas? Thanks! Stefan

2 8

[alsa-devel] [PATCH - use build platform compiler while cross compilation 1/1] use build compilers in cross compilation setup
by Ding Xiang Fei 29 Jun '19

29 Jun '19

This patch is to make the build script always use the build platform compilers, especially during the cross compilation process, since the Makefile will pick up the cross compilers instead of the compilers on the build machine. Signed-off-by: Ding Xiang Fei <dingxiangfei2009(a)gmail.com> diff --git a/compile b/compile new file mode 120000 index 0000000..349ac66 --- /dev/null +++ b/compile @@ -0,0 +1 @@ +/nix/store/hwcmvzv53jzws5vnaga16l5dmqvslwkz-automake-1.16.1/share/automake-1.16/compile \ No newline at end of file diff --git a/configure.ac b/configure.ac index 368854e..ee0e242 100644 --- a/configure.ac +++ b/configure.ac @@ -1,6 +1,8 @@ AC_PREREQ(2.59) AC_INIT(alsa-firmware, 1.0.29) +AC_CONFIG_MACRO_DIR([m4]) AC_PROG_CC +AC_PROG_CC_FOR_BUILD AC_PROG_INSTALL AC_PROG_LN_S AC_HEADER_STDC diff --git a/echoaudio/Makefile.am b/echoaudio/Makefile.am index 815742f..5ef6086 100644 --- a/echoaudio/Makefile.am +++ b/echoaudio/Makefile.am @@ -74,33 +74,42 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer DSP/LoaderDSP.c loader_dsp.fw - ./fw_writer DSP/Darla20DSP.c darla20_dsp.fw - ./fw_writer DSP/Gina20DSP.c gina20_dsp.fw - ./fw_writer DSP/Layla20DSP.c layla20_dsp.fw - ./fw_writer ASIC/LaylaASIC.c layla20_asic.fw - ./fw_writer DSP/Darla24DSP.c darla24_dsp.fw - ./fw_writer DSP/Gina24DSP.c gina24_301_dsp.fw - ./fw_writer ASIC/Gina24ASIC.c gina24_301_asic.fw - ./fw_writer DSP/Gina24_361DSP.c gina24_361_dsp.fw - ./fw_writer ASIC/Gina24ASIC_361.c gina24_361_asic.fw - ./fw_writer DSP/Layla24DSP.c layla24_dsp.fw - ./fw_writer ASIC/Layla24_1ASIC.c layla24_1_asic.fw - ./fw_writer ASIC/Layla24_2A_ASIC.c layla24_2A_asic.fw - ./fw_writer ASIC/Layla24_2S_ASIC.c layla24_2S_asic.fw - ./fw_writer DSP/MonaDSP.c mona_301_dsp.fw - ./fw_writer ASIC/Mona1ASIC48.c mona_301_1_asic_48.fw - ./fw_writer ASIC/Mona1ASIC96.c mona_301_1_asic_96.fw - ./fw_writer DSP/Mona361DSP.c mona_361_dsp.fw - ./fw_writer ASIC/Mona1ASIC48_361.c mona_361_1_asic_48.fw - ./fw_writer ASIC/Mona1ASIC96_361.c mona_361_1_asic_96.fw - ./fw_writer ASIC/Mona2ASIC.c mona_2_asic.fw - ./fw_writer DSP/MiaDSP.c mia_dsp.fw - ./fw_writer DSP/Echo3gDSP.c echo3g_dsp.fw - ./fw_writer ASIC/3G_ASIC.c 3g_asic.fw - ./fw_writer DSP/IndigoDSP.c indigo_dsp.fw - ./fw_writer DSP/IndigoIODSP.c indigo_io_dsp.fw - ./fw_writer DSP/IndigoIOxDSP.c indigo_iox_dsp.fw - ./fw_writer DSP/IndigoDJDSP.c indigo_dj_dsp.fw - ./fw_writer DSP/IndigoDJxDSP.c indigo_djx_dsp.fw +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) DSP/LoaderDSP.c loader_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Darla20DSP.c darla20_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Gina20DSP.c gina20_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Layla20DSP.c layla20_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/LaylaASIC.c layla20_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Darla24DSP.c darla24_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Gina24DSP.c gina24_301_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Gina24ASIC.c gina24_301_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Gina24_361DSP.c gina24_361_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Gina24ASIC_361.c gina24_361_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Layla24DSP.c layla24_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Layla24_1ASIC.c layla24_1_asic.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Layla24_2A_ASIC.c layla24_2A_asic.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Layla24_2S_ASIC.c layla24_2S_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/MonaDSP.c mona_301_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC48.c mona_301_1_asic_48.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC96.c mona_301_1_asic_96.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Mona361DSP.c mona_361_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC48_361.c mona_361_1_asic_48.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC96_361.c mona_361_1_asic_96.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona2ASIC.c mona_2_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/MiaDSP.c mia_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Echo3gDSP.c echo3g_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/3G_ASIC.c 3g_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoDSP.c indigo_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoIODSP.c indigo_io_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoIOxDSP.c indigo_iox_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoDJDSP.c indigo_dj_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoDJxDSP.c indigo_djx_dsp.fw diff --git a/emu/Makefile.am b/emu/Makefile.am index 3a633c0..e5a3017 100644 --- a/emu/Makefile.am +++ b/emu/Makefile.am @@ -22,5 +22,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/hdsploader/Makefile.am b/hdsploader/Makefile.am index 48802a6..b13245b 100644 --- a/hdsploader/Makefile.am +++ b/hdsploader/Makefile.am @@ -32,5 +32,14 @@ EXTRA_DIST = $(dsp_hex_files:%.bin=%.dat) \ tobin.c CLEANFILES = $(dsp_hex_files) -$(dsp_hex_files): tobin - ./tobin +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(tobin_OBJECTS) : CC=$(CC_FOR_BUILD) +$(tobin_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(tobin_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +tobin$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(dsp_hex_files): tobin$(BUILD_EXEEXT) + ./$< diff --git a/m4/ax_prog_cc_for_build.m4 b/m4/ax_prog_cc_for_build.m4 new file mode 100644 index 0000000..12cb005 --- /dev/null +++ b/m4/ax_prog_cc_for_build.m4 @@ -0,0 +1,125 @@ +# =========================================================================== +# https://www.gnu.org/software/autoconf-archive/ax_prog_cc_for_build.html +# =========================================================================== +# +# SYNOPSIS +# +# AX_PROG_CC_FOR_BUILD +# +# DESCRIPTION +# +# This macro searches for a C compiler that generates native executables, +# that is a C compiler that surely is not a cross-compiler. This can be +# useful if you have to generate source code at compile-time like for +# example GCC does. +# +# The macro sets the CC_FOR_BUILD and CPP_FOR_BUILD macros to anything +# needed to compile or link (CC_FOR_BUILD) and preprocess (CPP_FOR_BUILD). +# The value of these variables can be overridden by the user by specifying +# a compiler with an environment variable (like you do for standard CC). +# +# It also sets BUILD_EXEEXT and BUILD_OBJEXT to the executable and object +# file extensions for the build platform, and GCC_FOR_BUILD to `yes' if +# the compiler we found is GCC. All these variables but GCC_FOR_BUILD are +# substituted in the Makefile. +# +# LICENSE +# +# Copyright (c) 2008 Paolo Bonzini <bonzini(a)gnu.org> +# +# Copying and distribution of this file, with or without modification, are +# permitted in any medium without royalty provided the copyright notice +# and this notice are preserved. This file is offered as-is, without any +# warranty. + +#serial 9 + +AU_ALIAS([AC_PROG_CC_FOR_BUILD], [AX_PROG_CC_FOR_BUILD]) +AC_DEFUN([AX_PROG_CC_FOR_BUILD], [dnl +AC_REQUIRE([AC_PROG_CC])dnl +AC_REQUIRE([AC_PROG_CPP])dnl +AC_REQUIRE([AC_EXEEXT])dnl +AC_REQUIRE([AC_CANONICAL_HOST])dnl + +dnl Use the standard macros, but make them use other variable names +dnl +pushdef([ac_cv_prog_CPP], ac_cv_build_prog_CPP)dnl +pushdef([ac_cv_prog_gcc], ac_cv_build_prog_gcc)dnl +pushdef([ac_cv_prog_cc_works], ac_cv_build_prog_cc_works)dnl +pushdef([ac_cv_prog_cc_cross], ac_cv_build_prog_cc_cross)dnl +pushdef([ac_cv_prog_cc_g], ac_cv_build_prog_cc_g)dnl +pushdef([ac_cv_exeext], ac_cv_build_exeext)dnl +pushdef([ac_cv_objext], ac_cv_build_objext)dnl +pushdef([ac_exeext], ac_build_exeext)dnl +pushdef([ac_objext], ac_build_objext)dnl +pushdef([CC], CC_FOR_BUILD)dnl +pushdef([CPP], CPP_FOR_BUILD)dnl +pushdef([CFLAGS], CFLAGS_FOR_BUILD)dnl +pushdef([CPPFLAGS], CPPFLAGS_FOR_BUILD)dnl +pushdef([LDFLAGS], LDFLAGS_FOR_BUILD)dnl +pushdef([host], build)dnl +pushdef([host_alias], build_alias)dnl +pushdef([host_cpu], build_cpu)dnl +pushdef([host_vendor], build_vendor)dnl +pushdef([host_os], build_os)dnl +pushdef([ac_cv_host], ac_cv_build)dnl +pushdef([ac_cv_host_alias], ac_cv_build_alias)dnl +pushdef([ac_cv_host_cpu], ac_cv_build_cpu)dnl +pushdef([ac_cv_host_vendor], ac_cv_build_vendor)dnl +pushdef([ac_cv_host_os], ac_cv_build_os)dnl +pushdef([ac_cpp], ac_build_cpp)dnl +pushdef([ac_compile], ac_build_compile)dnl +pushdef([ac_link], ac_build_link)dnl + +save_cross_compiling=$cross_compiling +save_ac_tool_prefix=$ac_tool_prefix +cross_compiling=no +ac_tool_prefix= + +AC_PROG_CC +AC_PROG_CPP +AC_EXEEXT + +ac_tool_prefix=$save_ac_tool_prefix +cross_compiling=$save_cross_compiling + +dnl Restore the old definitions +dnl +popdef([ac_link])dnl +popdef([ac_compile])dnl +popdef([ac_cpp])dnl +popdef([ac_cv_host_os])dnl +popdef([ac_cv_host_vendor])dnl +popdef([ac_cv_host_cpu])dnl +popdef([ac_cv_host_alias])dnl +popdef([ac_cv_host])dnl +popdef([host_os])dnl +popdef([host_vendor])dnl +popdef([host_cpu])dnl +popdef([host_alias])dnl +popdef([host])dnl +popdef([LDFLAGS])dnl +popdef([CPPFLAGS])dnl +popdef([CFLAGS])dnl +popdef([CPP])dnl +popdef([CC])dnl +popdef([ac_objext])dnl +popdef([ac_exeext])dnl +popdef([ac_cv_objext])dnl +popdef([ac_cv_exeext])dnl +popdef([ac_cv_prog_cc_g])dnl +popdef([ac_cv_prog_cc_cross])dnl +popdef([ac_cv_prog_cc_works])dnl +popdef([ac_cv_prog_gcc])dnl +popdef([ac_cv_prog_CPP])dnl + +dnl Finally, set Makefile variables +dnl +BUILD_EXEEXT=$ac_build_exeext +BUILD_OBJEXT=$ac_build_objext +AC_SUBST(BUILD_EXEEXT)dnl +AC_SUBST(BUILD_OBJEXT)dnl +AC_SUBST([CFLAGS_FOR_BUILD])dnl +AC_SUBST([CPPFLAGS_FOR_BUILD])dnl +AC_SUBST([LDFLAGS_FOR_BUILD])dnl +]) diff --git a/maestro3/Makefile.am b/maestro3/Makefile.am index 7544f12..0826570 100644 --- a/maestro3/Makefile.am +++ b/maestro3/Makefile.am @@ -17,5 +17,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/sb16/Makefile.am b/sb16/Makefile.am index 17eead9..58d0032 100644 --- a/sb16/Makefile.am +++ b/sb16/Makefile.am @@ -18,5 +18,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/vxloader/Makefile.am b/vxloader/Makefile.am index ca21494..adcc204 100644 --- a/vxloader/Makefile.am +++ b/vxloader/Makefile.am @@ -43,5 +43,14 @@ hotplugfwdir = hotplugfw_DATA = endif -%.xlx: %.rbt toxlx - ./toxlx < $< > $@ +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(toxlx_OBJECTS) : CC=$(CC_FOR_BUILD) +$(toxlx_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(toxlx_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +toxlx$(BUILD_EXEEXT): $(toxlx_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +%.xlx: %.rbt toxlx$(BUILD_EXEEXT) + ./toxlx$(BUILD_EXEEXT) < $< > $@ diff --git a/wavefront/Makefile.am b/wavefront/Makefile.am index e8c9fe1..8f67b59 100644 --- a/wavefront/Makefile.am +++ b/wavefront/Makefile.am @@ -17,5 +17,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/ymfpci/Makefile.am b/ymfpci/Makefile.am index 8795fbc..645b471 100644 --- a/ymfpci/Makefile.am +++ b/ymfpci/Makefile.am @@ -17,5 +17,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) -- 2.22.0

1 0

[alsa-devel] [PATCH] ASoC: meson: axg-card: remove useless check on codec
by Jerome Brunet 28 Jun '19

28 Jun '19

While checking cpus before dereferencing the pointer is required, it is not necessary for codecs. 'codec' can't possibly be NULL in the loop Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 70bb0cbad233..14a8321744da 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -118,8 +118,7 @@ static void axg_card_clean_references(struct axg_card *priv) if (link->cpus) of_node_put(link->cpus->of_node); for_each_link_codecs(link, j, codec) - if (codec) - of_node_put(codec->of_node); + of_node_put(codec->of_node); } } -- 2.21.0

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: soc-core: update dai_link init
by Jerome Brunet 28 Jun '19

28 Jun '19

My initial goal with this patchset was to allow a dai_link to have no no platform component, instead of having dummy by default. However, when rebasing, I discovered that Kuninori Morimoto had recently done that in a different way :) I am still submitting my change since it should allow multiple platform components on a dai_link, which is one of the FIXME note in soc-core. I have also added a check on the codecs component availability to align on what was done for platforms and cpus Change since v1 [0]: * Fix registartion typo * Rename dlc variable to codec/platform [0]: https://lkml.kernel.org/r/20190626133617.25959-1-jbrunet@baylibre.com Jerome Brunet (2): ASoC: soc-core: defer card registration if codec component is missing ASoC: soc-core: support dai_link with platforms_num != 1 include/sound/soc.h | 6 ++++ sound/soc/soc-core.c | 67 +++++++++++++++++++++----------------------- 2 files changed, 38 insertions(+), 35 deletions(-) -- 2.21.0

2 4

[alsa-devel] [PATCH][next] ASoC: topology: fix memory leaks on sm, se and sbe
by Colin King 28 Jun '19

28 Jun '19

From: Colin Ian King <colin.king(a)canonical.com> Currently when a kstrdup fails the error exit paths don't free the allocations for sm, se and sbe. This can be fixed by assigning kc[i].private_value to these before doing the ksrtdup so that the error exit path will be able to free these objects. Addresses-Coverity: ("Resource leak") Fixes: 9f90af3a9952 ("ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index fc1f1d6f9e92..dc463f1a9e24 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1326,10 +1326,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", mc->hdr.name, i); + kc[i].private_value = (long)sm; kc[i].name = kstrdup(mc->hdr.name, GFP_KERNEL); if (kc[i].name == NULL) goto err_sm; - kc[i].private_value = (long)sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = mc->hdr.access; @@ -1412,10 +1412,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); + kc[i].private_value = (long)se; kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); if (kc[i].name == NULL) goto err_se; - kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; @@ -1524,10 +1524,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); + kc[i].private_value = (long)sbe; kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); if (kc[i].name == NULL) goto err_sbe; - kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access; -- 2.20.1

2 1

[alsa-devel] [PATCH v4] ASoC: rt1308: Add RT1308 amplifier driver
by derek.fang＠realtek.com 28 Jun '19

28 Jun '19

From: Derek Fang <derek.fang(a)realtek.com> This is the initial amplifier driver for rt1308. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt1308.txt | 17 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1308.c | 899 +++++++++++++++++++++ sound/soc/codecs/rt1308.h | 291 +++++++ 5 files changed, 1215 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/rt1308.txt create mode 100755 sound/soc/codecs/rt1308.c create mode 100755 sound/soc/codecs/rt1308.h diff --git a/Documentation/devicetree/bindings/sound/rt1308.txt b/Documentation/devicetree/bindings/sound/rt1308.txt new file mode 100755 index 0000000..2d46084 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1308.txt @@ -0,0 +1,17 @@ +RT1308 audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1308". + +- reg : The I2C address of the device. + + +Example: + +rt1308: rt1308@10 { + compatible = "realtek,rt1308"; + reg = <0x10>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b2d445b..9f89a53 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -146,6 +146,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT298 if I2C select SND_SOC_RT1011 if I2C select SND_SOC_RT1305 if I2C + select SND_SOC_RT1308 if I2C select SND_SOC_RT5514 if I2C select SND_SOC_RT5616 if I2C select SND_SOC_RT5631 if I2C @@ -904,6 +905,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5682=y default y if SND_SOC_RT1011=y default y if SND_SOC_RT1305=y + default y if SND_SOC_RT1308=y default m if SND_SOC_RT5514=m default m if SND_SOC_RT5616=m default m if SND_SOC_RT5640=m @@ -919,6 +921,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5682=m default m if SND_SOC_RT1011=m default m if SND_SOC_RT1305=m + default m if SND_SOC_RT1308=m config SND_SOC_RL6347A tristate @@ -947,6 +950,9 @@ config SND_SOC_RT1011 config SND_SOC_RT1305 tristate +config SND_SOC_RT1308 + tristate + config SND_SOC_RT5514 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ed74f5b..5b4bb8c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -148,6 +148,7 @@ snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o snd-soc-rt1305-objs := rt1305.o +snd-soc-rt1308-objs := rt1308.o snd-soc-rt274-objs := rt274.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o @@ -428,6 +429,7 @@ obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o +obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c new file mode 100755 index 0000000..619a630 --- /dev/null +++ b/sound/soc/codecs/rt1308.c @@ -0,0 +1,899 @@ +/* + * rt1308.c -- RT1308 ALSA SoC amplifier component driver + * + * Copyright 2019 Realtek Semiconductor Corp. + * Author: Derek Fang <derek.fang(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_gpio.h> +#include <linux/acpi.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rl6231.h" +#include "rt1308.h" + +static const struct reg_sequence init_list[] = { + + { RT1308_I2C_I2S_SDW_SET, 0x01014005 }, + { RT1308_CLASS_D_SET_2, 0x227f5501 }, + { RT1308_PADS_1, 0x50150505 }, + { RT1308_VREF, 0x18100000 }, + { RT1308_IV_SENSE, 0x87010000 }, + { RT1308_DUMMY_REG, 0x00000200 }, + { RT1308_SIL_DET, 0x61c30000 }, + { RT1308_DC_CAL_2, 0x00ffff00 }, + { RT1308_CLK_DET, 0x01000000 }, + { RT1308_POWER_STATUS, 0x00800000 }, + { RT1308_DAC_SET, 0xafaf0700 }, + +}; +#define RT1308_INIT_REG_LEN ARRAY_SIZE(init_list) + +struct rt1308_priv { + struct snd_soc_component *component; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck; + int bclk; + int master; + + int pll_src; + int pll_in; + int pll_out; +}; + +static const struct reg_default rt1308_reg[] = { + + { 0x01, 0x1f3f5f00 }, + { 0x02, 0x07000000 }, + { 0x03, 0x80003e00 }, + { 0x04, 0x80800600 }, + { 0x05, 0x0aaa1a0a }, + { 0x06, 0x52000000 }, + { 0x07, 0x00000000 }, + { 0x08, 0x00600000 }, + { 0x09, 0xe1030000 }, + { 0x0a, 0x00000000 }, + { 0x0b, 0x30000000 }, + { 0x0c, 0x7fff7000 }, + { 0x10, 0xffff0700 }, + { 0x11, 0x0a000000 }, + { 0x12, 0x60040000 }, + { 0x13, 0x00000000 }, + { 0x14, 0x0f300000 }, + { 0x15, 0x00000022 }, + { 0x16, 0x02000000 }, + { 0x17, 0x01004045 }, + { 0x18, 0x00000000 }, + { 0x19, 0x00000000 }, + { 0x1a, 0x80000000 }, + { 0x1b, 0x10325476 }, + { 0x1c, 0x1d1d0000 }, + { 0x20, 0xd2101300 }, + { 0x21, 0xf3ffff00 }, + { 0x22, 0x00000000 }, + { 0x23, 0x00000000 }, + { 0x24, 0x00000000 }, + { 0x25, 0x00000000 }, + { 0x26, 0x00000000 }, + { 0x27, 0x00000000 }, + { 0x28, 0x00000000 }, + { 0x29, 0x00000000 }, + { 0x2a, 0x00000000 }, + { 0x2b, 0x00000000 }, + { 0x2c, 0x00000000 }, + { 0x2d, 0x00000000 }, + { 0x2e, 0x00000000 }, + { 0x2f, 0x00000000 }, + { 0x30, 0x01000000 }, + { 0x31, 0x20025501 }, + { 0x32, 0x00000000 }, + { 0x33, 0x105a0000 }, + { 0x34, 0x10100000 }, + { 0x35, 0x2aaa52aa }, + { 0x36, 0x00c00000 }, + { 0x37, 0x20046100 }, + { 0x50, 0x10022f00 }, + { 0x51, 0x003c0000 }, + { 0x54, 0x04000000 }, + { 0x55, 0x01000000 }, + { 0x56, 0x02000000 }, + { 0x57, 0x02000000 }, + { 0x58, 0x02000000 }, + { 0x59, 0x02000000 }, + { 0x5b, 0x02000000 }, + { 0x5c, 0x00000000 }, + { 0x5d, 0x00000000 }, + { 0x5e, 0x00000000 }, + { 0x5f, 0x00000000 }, + { 0x60, 0x02000000 }, + { 0x61, 0x00000000 }, + { 0x62, 0x00000000 }, + { 0x63, 0x00000000 }, + { 0x64, 0x00000000 }, + { 0x65, 0x02000000 }, + { 0x66, 0x00000000 }, + { 0x67, 0x00000000 }, + { 0x68, 0x00000000 }, + { 0x69, 0x00000000 }, + { 0x6a, 0x02000000 }, + { 0x6c, 0x00000000 }, + { 0x6d, 0x00000000 }, + { 0x6e, 0x00000000 }, + { 0x70, 0x10EC1308 }, + { 0x71, 0x00000000 }, + { 0x72, 0x00000000 }, + { 0x73, 0x00000000 }, + { 0x74, 0x00000000 }, + { 0x75, 0x00000000 }, + { 0x76, 0x00000000 }, + { 0x77, 0x00000000 }, + { 0x78, 0x00000000 }, + { 0x79, 0x00000000 }, + { 0x7a, 0x00000000 }, + { 0x7b, 0x00000000 }, + { 0x7c, 0x00000000 }, + { 0x7d, 0x00000000 }, + { 0x7e, 0x00000000 }, + { 0x7f, 0x00020f00 }, + { 0x80, 0x00000000 }, + { 0x81, 0x00000000 }, + { 0x82, 0x00000000 }, + { 0x83, 0x00000000 }, + { 0x84, 0x00000000 }, + { 0x85, 0x00000000 }, + { 0x86, 0x00000000 }, + { 0x87, 0x00000000 }, + { 0x88, 0x00000000 }, + { 0x89, 0x00000000 }, + { 0x8a, 0x00000000 }, + { 0x8b, 0x00000000 }, + { 0x8c, 0x00000000 }, + { 0x8d, 0x00000000 }, + { 0x8e, 0x00000000 }, + { 0x90, 0x50250905 }, + { 0x91, 0x15050000 }, + { 0xa0, 0x00000000 }, + { 0xa1, 0x00000000 }, + { 0xa2, 0x00000000 }, + { 0xa3, 0x00000000 }, + { 0xa4, 0x00000000 }, + { 0xb0, 0x00000000 }, + { 0xb1, 0x00000000 }, + { 0xb2, 0x00000000 }, + { 0xb3, 0x00000000 }, + { 0xb4, 0x00000000 }, + { 0xb5, 0x00000000 }, + { 0xb6, 0x00000000 }, + { 0xb7, 0x00000000 }, + { 0xb8, 0x00000000 }, + { 0xb9, 0x00000000 }, + { 0xba, 0x00000000 }, + { 0xbb, 0x00000000 }, + { 0xc0, 0x01000000 }, + { 0xc1, 0x00000000 }, + { 0xf0, 0x00000000 }, +}; + +static int rt1308_reg_init(struct snd_soc_component *component) +{ + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + + return regmap_multi_reg_write(rt1308->regmap, init_list, + RT1308_INIT_REG_LEN); +} + +static bool rt1308_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1308_RESET: + case RT1308_RESET_N: + case RT1308_CLK_2: + case RT1308_SIL_DET: + case RT1308_CLK_DET: + case RT1308_DC_DET: + case RT1308_DAC_SET: + case RT1308_DAC_BUF: + case RT1308_SDW_REG_RDATA: + case RT1308_DC_CAL_1: + case RT1308_PVDD_OFFSET_CTL: + case RT1308_CAL_OFFSET_DAC_PBTL: + case RT1308_CAL_OFFSET_DAC_L: + case RT1308_CAL_OFFSET_DAC_R: + case RT1308_CAL_OFFSET_PWM_L: + case RT1308_CAL_OFFSET_PWM_R: + case RT1308_CAL_PWM_VOS_ADC_L: + case RT1308_CAL_PWM_VOS_ADC_R: + case RT1308_MBIAS: + case RT1308_POWER_STATUS: + case RT1308_POWER_INT: + case RT1308_SINE_TONE_GEN_2: + case RT1308_BQ_SET: + case RT1308_BQ_PARA_UPDATE: + case RT1308_VEN_DEV_ID: + case RT1308_VERSION_ID: + case RT1308_EFUSE_1: + case RT1308_EFUSE_READ_PVDD_L: + case RT1308_EFUSE_READ_PVDD_R: + case RT1308_EFUSE_READ_PVDD_PTBL: + case RT1308_EFUSE_READ_DEV: + case RT1308_EFUSE_READ_R0: + case RT1308_EFUSE_READ_ADC_L: + case RT1308_EFUSE_READ_ADC_R: + case RT1308_EFUSE_READ_ADC_PBTL: + case RT1308_EFUSE_RESERVE: + case RT1308_EFUSE_DATA_0_MSB: + case RT1308_EFUSE_DATA_0_LSB: + case RT1308_EFUSE_DATA_1_MSB: + case RT1308_EFUSE_DATA_1_LSB: + case RT1308_EFUSE_DATA_2_MSB: + case RT1308_EFUSE_DATA_2_LSB: + case RT1308_EFUSE_DATA_3_MSB: + case RT1308_EFUSE_DATA_3_LSB: + case RT1308_EFUSE_STATUS_1: + case RT1308_EFUSE_STATUS_2: + case RT1308_DUMMY_REG: + return true; + default: + return false; + } +} + +static bool rt1308_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1308_RESET: + case RT1308_RESET_N: + case RT1308_CLK_GATING ... RT1308_DC_DET_THRES: + case RT1308_DAC_SET ... RT1308_AD_FILTER_SET: + case RT1308_DC_CAL_1 ... RT1308_POWER_INT: + case RT1308_SINE_TONE_GEN_1: + case RT1308_SINE_TONE_GEN_2: + case RT1308_BQ_SET: + case RT1308_BQ_PARA_UPDATE: + case RT1308_BQ_PRE_VOL_L ... RT1308_BQ_POST_VOL_R: + case RT1308_BQ1_L_H0 ... RT1308_BQ2_R_A2: + case RT1308_VEN_DEV_ID: + case RT1308_VERSION_ID: + case RT1308_SPK_BOUND: + case RT1308_BQ1_EQ_L_1 ... RT1308_BQ2_EQ_R_3: + case RT1308_EFUSE_1 ... RT1308_EFUSE_RESERVE: + case RT1308_PADS_1: + case RT1308_PADS_2: + case RT1308_TEST_MODE: + case RT1308_TEST_1: + case RT1308_TEST_2: + case RT1308_TEST_3: + case RT1308_TEST_4: + case RT1308_EFUSE_DATA_0_MSB ... RT1308_EFUSE_STATUS_2: + case RT1308_TCON_1: + case RT1308_TCON_2: + case RT1308_DUMMY_REG: + case RT1308_MAX_REG: + return true; + default: + return false; + } +} + +static int rt1308_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(30); + snd_soc_component_update_bits(component, RT1308_POWER_STATUS, + RT1308_POW_PDB_REG_BIT, RT1308_POW_PDB_REG_BIT); + msleep(40); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, RT1308_POWER_STATUS, + RT1308_POW_PDB_REG_BIT, 0); + usleep_range(150000, 200000); + break; + + default: + break; + } + + return 0; +} + +static const char * const rt1308_rx_data_ch_select[] = { + "LR", + "LL", + "RL", + "RR", +}; + +static SOC_ENUM_SINGLE_DECL(rt1308_rx_data_ch_enum, RT1308_DATA_PATH, 24, + rt1308_rx_data_ch_select); + +static const struct snd_kcontrol_new rt1308_snd_controls[] = { + + /* I2S Data Channel Selection */ + SOC_ENUM("RX Channel Select", rt1308_rx_data_ch_enum), +}; + +static const struct snd_kcontrol_new rt1308_sto_dac_l = + SOC_DAPM_SINGLE("Switch", RT1308_DAC_SET, + RT1308_DVOL_MUTE_L_EN_SFT, 1, 1); + +static const struct snd_kcontrol_new rt1308_sto_dac_r = + SOC_DAPM_SINGLE("Switch", RT1308_DAC_SET, + RT1308_DVOL_MUTE_R_EN_SFT, 1, 1); + +static const struct snd_soc_dapm_widget rt1308_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Supply Widgets */ + SND_SOC_DAPM_SUPPLY("MBIAS20U", RT1308_POWER, + RT1308_POW_MBIAS20U_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ALDO", RT1308_POWER, + RT1308_POW_ALDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DBG", RT1308_POWER, + RT1308_POW_DBG_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DACL", RT1308_POWER, + RT1308_POW_DACL_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLK25M", RT1308_POWER, + RT1308_POW_CLK25M_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_R", RT1308_POWER, + RT1308_POW_ADC_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_L", RT1308_POWER, + RT1308_POW_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DLDO", RT1308_POWER, + RT1308_POW_DLDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", RT1308_POWER, + RT1308_POW_VREF_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIXER_R", RT1308_POWER, + RT1308_POW_MIXER_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIXER_L", RT1308_POWER, + RT1308_POW_MIXER_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MBIAS4U", RT1308_POWER, + RT1308_POW_MBIAS4U_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2_LDO", RT1308_POWER, + RT1308_POW_PLL2_LDO_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", RT1308_POWER, + RT1308_POW_PLL2B_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", RT1308_POWER, + RT1308_POW_PLL2F_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F2", RT1308_POWER, + RT1308_POW_PLL2F2_EN_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B2", RT1308_POWER, + RT1308_POW_PLL2B2_EN_BIT, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("DAC Power", RT1308_POWER, + RT1308_POW_DAC1_BIT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("DAC L", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_l), + SND_SOC_DAPM_SWITCH("DAC R", SND_SOC_NOPM, 0, 0, &rt1308_sto_dac_r), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1308_classd_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), +}; + +static const struct snd_soc_dapm_route rt1308_dapm_routes[] = { + + { "DAC", NULL, "AIF1RX" }, + + { "DAC", NULL, "MBIAS20U" }, + { "DAC", NULL, "ALDO" }, + { "DAC", NULL, "DBG" }, + { "DAC", NULL, "DACL" }, + { "DAC", NULL, "CLK25M" }, + { "DAC", NULL, "ADC_R" }, + { "DAC", NULL, "ADC_L" }, + { "DAC", NULL, "DLDO" }, + { "DAC", NULL, "VREF" }, + { "DAC", NULL, "MIXER_R" }, + { "DAC", NULL, "MIXER_L" }, + { "DAC", NULL, "MBIAS4U" }, + { "DAC", NULL, "PLL2_LDO" }, + { "DAC", NULL, "PLL2B" }, + { "DAC", NULL, "PLL2F" }, + { "DAC", NULL, "PLL2F2" }, + { "DAC", NULL, "PLL2B2" }, + + { "DAC L", "Switch", "DAC" }, + { "DAC R", "Switch", "DAC" }, + { "DAC L", NULL, "DAC Power" }, + { "DAC R", NULL, "DAC Power" }, + + { "CLASS D", NULL, "DAC L" }, + { "CLASS D", NULL, "DAC R" }, + { "SPOL", NULL, "CLASS D" }, + { "SPOR", NULL, "CLASS D" }, +}; + +static int rt1308_get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt1308_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt1308->lrck = params_rate(params); + pre_div = rt1308_get_clk_info(rt1308->sysclk, rt1308->lrck); + if (pre_div < 0) { + dev_err(component->dev, + "Unsupported clock setting %d\n", rt1308->lrck); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + bclk_ms = frame_size > 32; + rt1308->bclk = rt1308->lrck * (32 << bclk_ms); + + dev_dbg(component->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + dev_dbg(component->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt1308->lrck, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + val_len |= RT1308_I2S_DL_SEL_16B; + break; + case 20: + val_len |= RT1308_I2S_DL_SEL_20B; + break; + case 24: + val_len |= RT1308_I2S_DL_SEL_24B; + break; + case 8: + val_len |= RT1308_I2S_DL_SEL_8B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT1308_AIF1: + mask_clk = RT1308_DIV_FS_SYS_MASK; + val_clk = pre_div << RT1308_DIV_FS_SYS_SFT; + snd_soc_component_update_bits(component, + RT1308_I2S_SET_2, RT1308_I2S_DL_SEL_MASK, + val_len); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1308_CLK_1, + mask_clk, val_clk); + + return 0; +} + +static int rt1308_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, reg1_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + rt1308->master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT1308_I2S_DF_SEL_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT1308_I2S_DF_SEL_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT1308_I2S_DF_SEL_PCM_B; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg1_val |= RT1308_I2S_BCLK_INV; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT1308_AIF1: + snd_soc_component_update_bits(component, + RT1308_I2S_SET_1, RT1308_I2S_DF_SEL_MASK, + reg_val); + snd_soc_component_update_bits(component, + RT1308_I2S_SET_2, RT1308_I2S_BCLK_MASK, + reg1_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt1308_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0; + + if (freq == rt1308->sysclk && clk_id == rt1308->sysclk_src) + return 0; + + switch (clk_id) { + case RT1308_FS_SYS_S_MCLK: + reg_val |= RT1308_SEL_FS_SYS_SRC_MCLK; + snd_soc_component_update_bits(component, + RT1308_CLK_DET, RT1308_MCLK_DET_EN_MASK, + RT1308_MCLK_DET_EN); + break; + case RT1308_FS_SYS_S_BCLK: + reg_val |= RT1308_SEL_FS_SYS_SRC_BCLK; + break; + case RT1308_FS_SYS_S_PLL: + reg_val |= RT1308_SEL_FS_SYS_SRC_PLL; + break; + case RT1308_FS_SYS_S_RCCLK: + reg_val |= RT1308_SEL_FS_SYS_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT1308_CLK_1, + RT1308_SEL_FS_SYS_MASK, reg_val); + rt1308->sysclk = freq; + rt1308->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt1308_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt1308->pll_src && freq_in == rt1308->pll_in && + freq_out == rt1308->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt1308->pll_in = 0; + rt1308->pll_out = 0; + snd_soc_component_update_bits(component, + RT1308_CLK_1, RT1308_SEL_FS_SYS_MASK, + RT1308_SEL_FS_SYS_SRC_MCLK); + return 0; + } + + switch (source) { + case RT1308_PLL_S_MCLK: + snd_soc_component_update_bits(component, + RT1308_CLK_2, RT1308_SEL_PLL_SRC_MASK, + RT1308_SEL_PLL_SRC_MCLK); + snd_soc_component_update_bits(component, + RT1308_CLK_DET, RT1308_MCLK_DET_EN_MASK, + RT1308_MCLK_DET_EN); + break; + case RT1308_PLL_S_BCLK: + snd_soc_component_update_bits(component, + RT1308_CLK_2, RT1308_SEL_PLL_SRC_MASK, + RT1308_SEL_PLL_SRC_BCLK); + break; + case RT1308_PLL_S_RCCLK: + snd_soc_component_update_bits(component, + RT1308_CLK_2, RT1308_SEL_PLL_SRC_MASK, + RT1308_SEL_PLL_SRC_RCCLK); + freq_in = 25000000; + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT1308_PLL_1, + pll_code.k_code << RT1308_PLL1_K_SFT | + pll_code.m_bp << RT1308_PLL1_M_BYPASS_SFT | + (pll_code.m_bp ? 0 : pll_code.m_code) << RT1308_PLL1_M_SFT | + pll_code.n_code << RT1308_PLL1_N_SFT); + + rt1308->pll_in = freq_in; + rt1308->pll_out = freq_out; + rt1308->pll_src = source; + + return 0; +} + +static int rt1308_probe(struct snd_soc_component *component) +{ + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + + rt1308->component = component; + + return rt1308_reg_init(component); +} + +static void rt1308_remove(struct snd_soc_component *component) +{ + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + + regmap_write(rt1308->regmap, RT1308_RESET, 0); +} + +#ifdef CONFIG_PM +static int rt1308_suspend(struct snd_soc_component *component) +{ + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1308->regmap, true); + regcache_mark_dirty(rt1308->regmap); + + return 0; +} + +static int rt1308_resume(struct snd_soc_component *component) +{ + struct rt1308_priv *rt1308 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1308->regmap, false); + regcache_sync(rt1308->regmap); + + return 0; +} +#else +#define rt1308_suspend NULL +#define rt1308_resume NULL +#endif + +#define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000 +#define RT1308_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops rt1308_aif_dai_ops = { + .hw_params = rt1308_hw_params, + .set_fmt = rt1308_set_dai_fmt, +}; + +static struct snd_soc_dai_driver rt1308_dai[] = { + { + .name = "rt1308-aif", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1308_STEREO_RATES, + .formats = RT1308_FORMATS, + }, + .ops = &rt1308_aif_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt1308 = { + .probe = rt1308_probe, + .remove = rt1308_remove, + .suspend = rt1308_suspend, + .resume = rt1308_resume, + .controls = rt1308_snd_controls, + .num_controls = ARRAY_SIZE(rt1308_snd_controls), + .dapm_widgets = rt1308_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1308_dapm_widgets), + .dapm_routes = rt1308_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1308_dapm_routes), + .set_sysclk = rt1308_set_component_sysclk, + .set_pll = rt1308_set_component_pll, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt1308_regmap = { + .reg_bits = 8, + .val_bits = 32, + .max_register = RT1308_MAX_REG, + .volatile_reg = rt1308_volatile_register, + .readable_reg = rt1308_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt1308_reg, + .num_reg_defaults = ARRAY_SIZE(rt1308_reg), + .use_single_read = true, + .use_single_write = true, +}; + +#ifdef CONFIG_OF +static const struct of_device_id rt1308_of_match[] = { + { .compatible = "realtek,rt1308", }, + { }, +}; +MODULE_DEVICE_TABLE(of, rt1308_of_match); +#endif + +#ifdef CONFIG_ACPI +static struct acpi_device_id rt1308_acpi_match[] = { + { "10EC1308", 0, }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1308_acpi_match); +#endif + +static const struct i2c_device_id rt1308_i2c_id[] = { + { "rt1308", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt1308_i2c_id); + +static void rt1308_efuse(struct rt1308_priv *rt1308) +{ + regmap_write(rt1308->regmap, RT1308_RESET, 0); + + regmap_write(rt1308->regmap, RT1308_POWER, 0xff371600); + regmap_write(rt1308->regmap, RT1308_CLK_1, 0x52100000); + regmap_write(rt1308->regmap, RT1308_I2C_I2S_SDW_SET, 0x01014005); + regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x227f5501); + regmap_write(rt1308->regmap, RT1308_PADS_1, 0x50150505); + regmap_write(rt1308->regmap, RT1308_VREF, 0x18100000); + regmap_write(rt1308->regmap, RT1308_IV_SENSE, 0x87010000); + regmap_write(rt1308->regmap, RT1308_DUMMY_REG, 0x00000200); + regmap_write(rt1308->regmap, RT1308_SIL_DET, 0x61c30000); + regmap_write(rt1308->regmap, RT1308_CLK_DET, 0x03700000); + regmap_write(rt1308->regmap, RT1308_SINE_TONE_GEN_1, 0x50022f00); + regmap_write(rt1308->regmap, RT1308_POWER_STATUS, 0x01800000); + regmap_write(rt1308->regmap, RT1308_DC_CAL_2, 0x00ffff00); + regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x607e5501); + + regmap_write(rt1308->regmap, RT1308_CLK_2, 0x0060e000); + regmap_write(rt1308->regmap, RT1308_EFUSE_1, 0x04fe0f00); + msleep(100); + regmap_write(rt1308->regmap, RT1308_EFUSE_1, 0x44fe0f00); + msleep(20); + regmap_write(rt1308->regmap, RT1308_PVDD_OFFSET_CTL, 0x10000000); + + regmap_write(rt1308->regmap, RT1308_POWER_STATUS, 0x00800000); + regmap_write(rt1308->regmap, RT1308_POWER, 0x0); + regmap_write(rt1308->regmap, RT1308_CLK_1, 0x52000000); + regmap_write(rt1308->regmap, RT1308_CLASS_D_SET_2, 0x227f5501); + regmap_write(rt1308->regmap, RT1308_SINE_TONE_GEN_1, 0x10022f00); +} + +static int rt1308_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt1308_priv *rt1308; + int ret; + unsigned int val; + + rt1308 = devm_kzalloc(&i2c->dev, sizeof(struct rt1308_priv), + GFP_KERNEL); + if (rt1308 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt1308); + + rt1308->regmap = devm_regmap_init_i2c(i2c, &rt1308_regmap); + if (IS_ERR(rt1308->regmap)) { + ret = PTR_ERR(rt1308->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt1308->regmap, RT1308_VEN_DEV_ID, &val); + /* ignore last byte difference */ + if ((val & 0xFFFFFF00) != RT1308_DEVICE_ID_NUM) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt1308\n", val); + return -ENODEV; + } + + rt1308_efuse(rt1308); + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1308, + rt1308_dai, ARRAY_SIZE(rt1308_dai)); +} + +static void rt1308_i2c_shutdown(struct i2c_client *client) +{ + struct rt1308_priv *rt1308 = i2c_get_clientdata(client); + + regmap_write(rt1308->regmap, RT1308_RESET, 0); +} + +static struct i2c_driver rt1308_i2c_driver = { + .driver = { + .name = "rt1308", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt1308_of_match), + .acpi_match_table = ACPI_PTR(rt1308_acpi_match), + }, + .probe = rt1308_i2c_probe, + .shutdown = rt1308_i2c_shutdown, + .id_table = rt1308_i2c_id, +}; +module_i2c_driver(rt1308_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT1308 amplifier driver"); +MODULE_AUTHOR("Derek Fang <derek.fang(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h new file mode 100755 index 0000000..c330aae --- /dev/null +++ b/sound/soc/codecs/rt1308.h @@ -0,0 +1,291 @@ +/* + * RT1308.h -- RT1308 ALSA SoC amplifier component driver + * + * Copyright 2019 Realtek Semiconductor Corp. + * Author: Derek Fang <derek.fang(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _RT1308_H_ +#define _RT1308_H_ + +#define RT1308_DEVICE_ID_NUM 0x10ec1300 + +#define RT1308_RESET 0x00 +#define RT1308_RESET_N 0x01 +#define RT1308_CLK_GATING 0x02 +#define RT1308_PLL_1 0x03 +#define RT1308_PLL_2 0x04 +#define RT1308_PLL_INT 0x05 +#define RT1308_CLK_1 0x06 +#define RT1308_DATA_PATH 0x07 +#define RT1308_CLK_2 0x08 +#define RT1308_SIL_DET 0x09 +#define RT1308_CLK_DET 0x0a +#define RT1308_DC_DET 0x0b +#define RT1308_DC_DET_THRES 0x0c +#define RT1308_DAC_SET 0x10 +#define RT1308_SRC_SET 0x11 +#define RT1308_DAC_BUF 0x12 +#define RT1308_ADC_SET 0x13 +#define RT1308_ADC_SET_INT 0x14 +#define RT1308_I2S_SET_1 0x15 +#define RT1308_I2S_SET_2 0x16 +#define RT1308_I2C_I2S_SDW_SET 0x17 +#define RT1308_SDW_REG_RW 0x18 +#define RT1308_SDW_REG_RDATA 0x19 +#define RT1308_IV_SENSE 0x1a +#define RT1308_I2S_TX_DAC_SET 0x1b +#define RT1308_AD_FILTER_SET 0x1c +#define RT1308_DC_CAL_1 0x20 +#define RT1308_DC_CAL_2 0x21 +#define RT1308_DC_CAL_L_OFFSET 0x22 +#define RT1308_DC_CAL_R_OFFSET 0x23 +#define RT1308_PVDD_OFFSET_CTL 0x24 +#define RT1308_PVDD_OFFSET_L 0x25 +#define RT1308_PVDD_OFFSET_R 0x26 +#define RT1308_PVDD_OFFSET_PBTL 0x27 +#define RT1308_PVDD_OFFSET_PVDD 0x28 +#define RT1308_CAL_OFFSET_DAC_PBTL 0x29 +#define RT1308_CAL_OFFSET_DAC_L 0x2a +#define RT1308_CAL_OFFSET_DAC_R 0x2b +#define RT1308_CAL_OFFSET_PWM_L 0x2c +#define RT1308_CAL_OFFSET_PWM_R 0x2d +#define RT1308_CAL_PWM_VOS_ADC_L 0x2e +#define RT1308_CAL_PWM_VOS_ADC_R 0x2f +#define RT1308_CLASS_D_SET_1 0x30 +#define RT1308_CLASS_D_SET_2 0x31 +#define RT1308_POWER 0x32 +#define RT1308_LDO 0x33 +#define RT1308_VREF 0x34 +#define RT1308_MBIAS 0x35 +#define RT1308_POWER_STATUS 0x36 +#define RT1308_POWER_INT 0x37 +#define RT1308_SINE_TONE_GEN_1 0x50 +#define RT1308_SINE_TONE_GEN_2 0x51 +#define RT1308_BQ_SET 0x54 +#define RT1308_BQ_PARA_UPDATE 0x55 +#define RT1308_BQ_PRE_VOL_L 0x56 +#define RT1308_BQ_PRE_VOL_R 0x57 +#define RT1308_BQ_POST_VOL_L 0x58 +#define RT1308_BQ_POST_VOL_R 0x59 +#define RT1308_BQ1_L_H0 0x5b +#define RT1308_BQ1_L_B1 0x5c +#define RT1308_BQ1_L_B2 0x5d +#define RT1308_BQ1_L_A1 0x5e +#define RT1308_BQ1_L_A2 0x5f +#define RT1308_BQ1_R_H0 0x60 +#define RT1308_BQ1_R_B1 0x61 +#define RT1308_BQ1_R_B2 0x62 +#define RT1308_BQ1_R_A1 0x63 +#define RT1308_BQ1_R_A2 0x64 +#define RT1308_BQ2_L_H0 0x65 +#define RT1308_BQ2_L_B1 0x66 +#define RT1308_BQ2_L_B2 0x67 +#define RT1308_BQ2_L_A1 0x68 +#define RT1308_BQ2_L_A2 0x69 +#define RT1308_BQ2_R_H0 0x6a +#define RT1308_BQ2_R_B1 0x6b +#define RT1308_BQ2_R_B2 0x6c +#define RT1308_BQ2_R_A1 0x6d +#define RT1308_BQ2_R_A2 0x6e +#define RT1308_VEN_DEV_ID 0x70 +#define RT1308_VERSION_ID 0x71 +#define RT1308_SPK_BOUND 0x72 +#define RT1308_BQ1_EQ_L_1 0x73 +#define RT1308_BQ1_EQ_L_2 0x74 +#define RT1308_BQ1_EQ_L_3 0x75 +#define RT1308_BQ1_EQ_R_1 0x76 +#define RT1308_BQ1_EQ_R_2 0x77 +#define RT1308_BQ1_EQ_R_3 0x78 +#define RT1308_BQ2_EQ_L_1 0x79 +#define RT1308_BQ2_EQ_L_2 0x7a +#define RT1308_BQ2_EQ_L_3 0x7b +#define RT1308_BQ2_EQ_R_1 0x7c +#define RT1308_BQ2_EQ_R_2 0x7d +#define RT1308_BQ2_EQ_R_3 0x7e +#define RT1308_EFUSE_1 0x7f +#define RT1308_EFUSE_2 0x80 +#define RT1308_EFUSE_PROG_PVDD_L 0x81 +#define RT1308_EFUSE_PROG_PVDD_R 0x82 +#define RT1308_EFUSE_PROG_R0_L 0x83 +#define RT1308_EFUSE_PROG_R0_R 0x84 +#define RT1308_EFUSE_PROG_DEV 0x85 +#define RT1308_EFUSE_READ_PVDD_L 0x86 +#define RT1308_EFUSE_READ_PVDD_R 0x87 +#define RT1308_EFUSE_READ_PVDD_PTBL 0x88 +#define RT1308_EFUSE_READ_DEV 0x89 +#define RT1308_EFUSE_READ_R0 0x8a +#define RT1308_EFUSE_READ_ADC_L 0x8b +#define RT1308_EFUSE_READ_ADC_R 0x8c +#define RT1308_EFUSE_READ_ADC_PBTL 0x8d +#define RT1308_EFUSE_RESERVE 0x8e +#define RT1308_PADS_1 0x90 +#define RT1308_PADS_2 0x91 +#define RT1308_TEST_MODE 0xa0 +#define RT1308_TEST_1 0xa1 +#define RT1308_TEST_2 0xa2 +#define RT1308_TEST_3 0xa3 +#define RT1308_TEST_4 0xa4 +#define RT1308_EFUSE_DATA_0_MSB 0xb0 +#define RT1308_EFUSE_DATA_0_LSB 0xb1 +#define RT1308_EFUSE_DATA_1_MSB 0xb2 +#define RT1308_EFUSE_DATA_1_LSB 0xb3 +#define RT1308_EFUSE_DATA_2_MSB 0xb4 +#define RT1308_EFUSE_DATA_2_LSB 0xb5 +#define RT1308_EFUSE_DATA_3_MSB 0xb6 +#define RT1308_EFUSE_DATA_3_LSB 0xb7 +#define RT1308_EFUSE_DATA_TEST_MSB 0xb8 +#define RT1308_EFUSE_DATA_TEST_LSB 0xb9 +#define RT1308_EFUSE_STATUS_1 0xba +#define RT1308_EFUSE_STATUS_2 0xbb +#define RT1308_TCON_1 0xc0 +#define RT1308_TCON_2 0xc1 +#define RT1308_DUMMY_REG 0xf0 +#define RT1308_MAX_REG 0xff + +/* PLL1 M/N/K Code-1 (0x03) */ +#define RT1308_PLL1_K_SFT 24 +#define RT1308_PLL1_K_MASK (0x1f << 24) +#define RT1308_PLL1_M_BYPASS_MASK (0x1 << 23) +#define RT1308_PLL1_M_BYPASS_SFT 23 +#define RT1308_PLL1_M_BYPASS (0x1 << 23) +#define RT1308_PLL1_M_MASK (0x3f << 16) +#define RT1308_PLL1_M_SFT 16 +#define RT1308_PLL1_N_MASK (0x7f << 8) +#define RT1308_PLL1_N_SFT 8 + +/* CLOCK-1 (0x06) */ +#define RT1308_DIV_FS_SYS_MASK (0xf << 28) +#define RT1308_DIV_FS_SYS_SFT 28 +#define RT1308_SEL_FS_SYS_MASK (0x7 << 24) +#define RT1308_SEL_FS_SYS_SFT 24 +#define RT1308_SEL_FS_SYS_SRC_MCLK (0x0 << 24) +#define RT1308_SEL_FS_SYS_SRC_BCLK (0x1 << 24) +#define RT1308_SEL_FS_SYS_SRC_PLL (0x2 << 24) +#define RT1308_SEL_FS_SYS_SRC_RCCLK (0x4 << 24) + +/* CLOCK-2 (0x08) */ +#define RT1308_DIV_PRE_PLL_MASK (0xf << 28) +#define RT1308_DIV_PRE_PLL_SFT 28 +#define RT1308_SEL_PLL_SRC_MASK (0x7 << 24) +#define RT1308_SEL_PLL_SRC_SFT 24 +#define RT1308_SEL_PLL_SRC_MCLK (0x0 << 24) +#define RT1308_SEL_PLL_SRC_BCLK (0x1 << 24) +#define RT1308_SEL_PLL_SRC_RCCLK (0x4 << 24) + +/* Clock Detect (0x0a) */ +#define RT1308_MCLK_DET_EN_MASK (0x1 << 25) +#define RT1308_MCLK_DET_EN_SFT 25 +#define RT1308_MCLK_DET_EN (0x1 << 25) +#define RT1308_BCLK_DET_EN_MASK (0x1 << 24) +#define RT1308_BCLK_DET_EN_SFT 24 +#define RT1308_BCLK_DET_EN (0x1 << 24) + +/* DAC Setting (0x10) */ +#define RT1308_DVOL_MUTE_R_EN_SFT 7 +#define RT1308_DVOL_MUTE_L_EN_SFT 6 + +/* I2S Setting-1 (0x15) */ +#define RT1308_I2S_DF_SEL_MASK (0x3 << 12) +#define RT1308_I2S_DF_SEL_SFT 12 +#define RT1308_I2S_DF_SEL_I2S (0x0 << 12) +#define RT1308_I2S_DF_SEL_LEFT (0x1 << 12) +#define RT1308_I2S_DF_SEL_PCM_A (0x2 << 12) +#define RT1308_I2S_DF_SEL_PCM_B (0x3 << 12) +#define RT1308_I2S_DL_RX_SEL_MASK (0x7 << 4) +#define RT1308_I2S_DL_RX_SEL_SFT 4 +#define RT1308_I2S_DL_RX_SEL_16B (0x0 << 4) +#define RT1308_I2S_DL_RX_SEL_20B (0x1 << 4) +#define RT1308_I2S_DL_RX_SEL_24B (0x2 << 4) +#define RT1308_I2S_DL_RX_SEL_32B (0x3 << 4) +#define RT1308_I2S_DL_RX_SEL_8B (0x4 << 4) +#define RT1308_I2S_DL_TX_SEL_MASK (0x7 << 0) +#define RT1308_I2S_DL_TX_SEL_SFT 0 +#define RT1308_I2S_DL_TX_SEL_16B (0x0 << 0) +#define RT1308_I2S_DL_TX_SEL_20B (0x1 << 0) +#define RT1308_I2S_DL_TX_SEL_24B (0x2 << 0) +#define RT1308_I2S_DL_TX_SEL_32B (0x3 << 0) +#define RT1308_I2S_DL_TX_SEL_8B (0x4 << 0) + +/* I2S Setting-2 (0x16) */ +#define RT1308_I2S_DL_SEL_MASK (0x7 << 24) +#define RT1308_I2S_DL_SEL_SFT 24 +#define RT1308_I2S_DL_SEL_16B (0x0 << 24) +#define RT1308_I2S_DL_SEL_20B (0x1 << 24) +#define RT1308_I2S_DL_SEL_24B (0x2 << 24) +#define RT1308_I2S_DL_SEL_32B (0x3 << 24) +#define RT1308_I2S_DL_SEL_8B (0x4 << 24) +#define RT1308_I2S_BCLK_MASK (0x1 << 14) +#define RT1308_I2S_BCLK_SFT 14 +#define RT1308_I2S_BCLK_NORMAL (0x0 << 14) +#define RT1308_I2S_BCLK_INV (0x1 << 14) + +/* Power Control-1 (0x32) */ +#define RT1308_POW_MBIAS20U (0x1 << 31) +#define RT1308_POW_MBIAS20U_BIT 31 +#define RT1308_POW_ALDO (0x1 << 30) +#define RT1308_POW_ALDO_BIT 30 +#define RT1308_POW_DBG (0x1 << 29) +#define RT1308_POW_DBG_BIT 29 +#define RT1308_POW_DACL (0x1 << 28) +#define RT1308_POW_DACL_BIT 28 +#define RT1308_POW_DAC1 (0x1 << 27) +#define RT1308_POW_DAC1_BIT 27 +#define RT1308_POW_CLK25M (0x1 << 26) +#define RT1308_POW_CLK25M_BIT 26 +#define RT1308_POW_ADC_R (0x1 << 25) +#define RT1308_POW_ADC_R_BIT 25 +#define RT1308_POW_ADC_L (0x1 << 24) +#define RT1308_POW_ADC_L_BIT 24 +#define RT1308_POW_DLDO (0x1 << 21) +#define RT1308_POW_DLDO_BIT 21 +#define RT1308_POW_VREF (0x1 << 20) +#define RT1308_POW_VREF_BIT 20 +#define RT1308_POW_MIXER_R (0x1 << 18) +#define RT1308_POW_MIXER_R_BIT 18 +#define RT1308_POW_MIXER_L (0x1 << 17) +#define RT1308_POW_MIXER_L_BIT 17 +#define RT1308_POW_MBIAS4U (0x1 << 16) +#define RT1308_POW_MBIAS4U_BIT 16 +#define RT1308_POW_PLL2_LDO_EN (0x1 << 12) +#define RT1308_POW_PLL2_LDO_EN_BIT 12 +#define RT1308_POW_PLL2B_EN (0x1 << 11) +#define RT1308_POW_PLL2B_EN_BIT 11 +#define RT1308_POW_PLL2F_EN (0x1 << 10) +#define RT1308_POW_PLL2F_EN_BIT 10 +#define RT1308_POW_PLL2F2_EN (0x1 << 9) +#define RT1308_POW_PLL2F2_EN_BIT 9 +#define RT1308_POW_PLL2B2_EN (0x1 << 8) +#define RT1308_POW_PLL2B2_EN_BIT 8 + +/* Power Control-2 (0x36) */ +#define RT1308_POW_PDB_SRC_BIT (0x1 << 27) +#define RT1308_POW_PDB_MN_BIT (0x1 << 25) +#define RT1308_POW_PDB_REG_BIT (0x1 << 24) + + +/* System Clock Source */ +enum { + RT1308_FS_SYS_S_MCLK, + RT1308_FS_SYS_S_BCLK, + RT1308_FS_SYS_S_PLL, + RT1308_FS_SYS_S_RCCLK, /* 25.0 MHz */ +}; + +/* PLL Source */ +enum { + RT1308_PLL_S_MCLK, + RT1308_PLL_S_BCLK, + RT1308_PLL_S_RCCLK, +}; + +enum { + RT1308_AIF1, + RT1308_AIFS +}; + +#endif /* end of _RT1308_H_ */ -- 2.7.4

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[alsa-devel] ca0132 audio in Ubuntu 19.04 only after Windows 10 started, missing ctefx-r3di.bin
by Kai-Heng Feng 28 Jun '19

28 Jun '19

Hi Connor, The bug was filed at Launchpad [1], I think the most notable error is [ 3.768667] snd_hda_intel 0000:00:1f.3: Direct firmware load for ctefx-r3di.bin failed with error -2 The firmware is indeed listed in patch_ca0132.c, but looks like there’s no corresponding file in linux-firmware. Can you please take a look at the bug? [1] https://bugs.launchpad.net/bugs/1833470 Kai-Heng

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